From 9f9af7df7be4509183e9d6917741aeb3403736dc Mon Sep 17 00:00:00 2001 From: "perkj@webrtc.org" Date: Thu, 17 Nov 2011 08:42:33 +0000 Subject: [PATCH] Remove Peerconnection Dev branch. BUG= Issue 139: libjingle gyp warning on Windows TEST= Review URL: http://webrtc-codereview.appspot.com/281001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@960 4adac7df-926f-26a2-2b94-8c16560cd09d --- third_party_mods/libjingle/libjingle.gyp | 128 +- .../talk/app/webrtc_dev/audiotrackimpl.cc | 71 - .../talk/app/webrtc_dev/audiotrackimpl.h | 70 - .../source/talk/app/webrtc_dev/mediastream.h | 189 --- .../app/webrtc_dev/mediastream_unittest.cc | 389 ----- .../talk/app/webrtc_dev/mediastreamhandler.cc | 255 ---- .../talk/app/webrtc_dev/mediastreamhandler.h | 144 -- .../webrtc_dev/mediastreamhandler_unittest.cc | 147 -- .../talk/app/webrtc_dev/mediastreamimpl.cc | 73 - .../talk/app/webrtc_dev/mediastreamimpl.h | 90 -- .../webrtc_dev/mediastreamimpl_unittest.cc | 95 -- .../talk/app/webrtc_dev/mediastreamprovider.h | 52 - .../talk/app/webrtc_dev/mediastreamproxy.cc | 318 ---- .../talk/app/webrtc_dev/mediastreamproxy.h | 111 -- .../app/webrtc_dev/mediastreamtrackproxy.cc | 394 ----- .../app/webrtc_dev/mediastreamtrackproxy.h | 143 -- .../talk/app/webrtc_dev/mediatrackimpl.h | 78 - .../source/talk/app/webrtc_dev/notifierimpl.h | 73 - .../talk/app/webrtc_dev/peerconnection.h | 208 --- .../app/webrtc_dev/peerconnection_unittest.cc | 348 ----- .../webrtc_dev/peerconnection_unittests.cc | 36 - .../peerconnectionfactory_unittest.cc | 111 -- .../webrtc_dev/peerconnectionfactoryimpl.cc | 241 --- .../webrtc_dev/peerconnectionfactoryimpl.h | 96 -- .../talk/app/webrtc_dev/peerconnectionimpl.cc | 290 ---- .../talk/app/webrtc_dev/peerconnectionimpl.h | 109 -- .../webrtc_dev/peerconnectionimpl_unittest.cc | 81 - .../app/webrtc_dev/peerconnectionmessage.cc | 95 -- .../app/webrtc_dev/peerconnectionmessage.h | 102 -- .../peerconnectionmessage_unittest.cc | 169 --- .../app/webrtc_dev/peerconnectionsignaling.cc | 535 ------- .../app/webrtc_dev/peerconnectionsignaling.h | 163 -- .../peerconnectionsignaling_unittest.cc | 455 ------ .../webrtc_dev/sessiondescriptionprovider.h | 58 - .../app/webrtc_dev/streamcollectionimpl.h | 103 -- .../webrtc_dev/test/filevideocapturemodule.cc | 165 --- .../webrtc_dev/test/filevideocapturemodule.h | 203 --- .../talk/app/webrtc_dev/videorendererimpl.cc | 58 - .../talk/app/webrtc_dev/videotrackimpl.cc | 79 - .../talk/app/webrtc_dev/videotrackimpl.h | 73 - .../source/talk/app/webrtc_dev/webrtcjson.cc | 639 -------- .../source/talk/app/webrtc_dev/webrtcjson.h | 62 - .../talk/app/webrtc_dev/webrtcsession.cc | 335 ----- .../talk/app/webrtc_dev/webrtcsession.h | 126 -- .../app/webrtc_dev/webrtcsession_unittest.cc | 135 -- .../app/webrtc_dev/webrtcsessionobserver.h | 47 - .../libjingle/source/talk/base/refcount.h | 90 -- .../source/talk/base/scoped_refptr.h | 132 -- .../peerconnection_client/conductor.cc | 378 ----- .../peerconnection_client/conductor.h | 116 -- .../peerconnection_client/defaults.cc | 58 - .../examples/peerconnection_client/defaults.h | 30 - .../peerconnection_client/linux/main.cc | 85 -- .../peerconnection_client/linux/main_wnd.cc | 464 ------ .../peerconnection_client/linux/main_wnd.h | 118 -- .../examples/peerconnection_client/main.cc | 58 - .../peerconnection_client/main_wnd.cc | 582 -------- .../examples/peerconnection_client/main_wnd.h | 191 --- .../peer_connection_client.cc | 482 ------ .../peer_connection_client.h | 109 -- .../peerconnection_client.gyp | 59 - .../peerconnection_client/server_test.html | 231 --- .../talk/p2p/client/fakeportallocator.h | 92 -- .../source/talk/session/phone/channel.cc | 1317 ----------------- .../source/talk/session/phone/channel.h | 515 ------- .../source/talk/session/phone/mediasession.cc | 591 -------- .../source/talk/session/phone/mediasession.h | 284 ---- .../source/talk/session/phone/streamparams.h | 105 -- 68 files changed, 6 insertions(+), 14023 deletions(-) delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/audiotrackimpl.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/audiotrackimpl.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastream.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastream_unittest.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamhandler.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamhandler.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamhandler_unittest.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamimpl.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamimpl.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamimpl_unittest.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamprovider.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamproxy.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamproxy.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamtrackproxy.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamtrackproxy.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/mediatrackimpl.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/notifierimpl.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnection.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnection_unittest.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnection_unittests.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionfactory_unittest.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionfactoryimpl.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionfactoryimpl.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionimpl.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionimpl.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionimpl_unittest.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionmessage.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionmessage.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionmessage_unittest.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionsignaling.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionsignaling.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionsignaling_unittest.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/sessiondescriptionprovider.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/streamcollectionimpl.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/test/filevideocapturemodule.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/test/filevideocapturemodule.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/videorendererimpl.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/videotrackimpl.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/videotrackimpl.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcjson.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcjson.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsession.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsession.h delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsession_unittest.cc delete mode 100644 third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsessionobserver.h delete mode 100644 third_party_mods/libjingle/source/talk/base/refcount.h delete mode 100644 third_party_mods/libjingle/source/talk/base/scoped_refptr.h delete mode 100644 third_party_mods/libjingle/source/talk/examples/peerconnection_client/conductor.cc delete mode 100644 third_party_mods/libjingle/source/talk/examples/peerconnection_client/conductor.h delete mode 100644 third_party_mods/libjingle/source/talk/examples/peerconnection_client/defaults.cc delete mode 100644 third_party_mods/libjingle/source/talk/examples/peerconnection_client/defaults.h delete mode 100644 third_party_mods/libjingle/source/talk/examples/peerconnection_client/linux/main.cc delete mode 100644 third_party_mods/libjingle/source/talk/examples/peerconnection_client/linux/main_wnd.cc delete mode 100644 third_party_mods/libjingle/source/talk/examples/peerconnection_client/linux/main_wnd.h delete mode 100644 third_party_mods/libjingle/source/talk/examples/peerconnection_client/main.cc delete mode 100644 third_party_mods/libjingle/source/talk/examples/peerconnection_client/main_wnd.cc delete mode 100644 third_party_mods/libjingle/source/talk/examples/peerconnection_client/main_wnd.h delete mode 100644 third_party_mods/libjingle/source/talk/examples/peerconnection_client/peer_connection_client.cc delete mode 100644 third_party_mods/libjingle/source/talk/examples/peerconnection_client/peer_connection_client.h delete mode 100644 third_party_mods/libjingle/source/talk/examples/peerconnection_client/peerconnection_client.gyp delete mode 100644 third_party_mods/libjingle/source/talk/examples/peerconnection_client/server_test.html delete mode 100644 third_party_mods/libjingle/source/talk/p2p/client/fakeportallocator.h delete mode 100644 third_party_mods/libjingle/source/talk/session/phone/channel.cc delete mode 100644 third_party_mods/libjingle/source/talk/session/phone/channel.h delete mode 100644 third_party_mods/libjingle/source/talk/session/phone/mediasession.cc delete mode 100644 third_party_mods/libjingle/source/talk/session/phone/mediasession.h delete mode 100644 third_party_mods/libjingle/source/talk/session/phone/streamparams.h diff --git a/third_party_mods/libjingle/libjingle.gyp b/third_party_mods/libjingle/libjingle.gyp index b12e86c04..4c7fb0884 100644 --- a/third_party_mods/libjingle/libjingle.gyp +++ b/third_party_mods/libjingle/libjingle.gyp @@ -5,7 +5,6 @@ { 'variables': { 'no_libjingle_logging%': 0, - 'peer_connection_dev%': 0, 'libjingle_orig': '../../third_party/libjingle', 'libjingle_mods': '../../third_party_mods/libjingle', 'conditions': [ @@ -109,11 +108,6 @@ }, }, 'conditions': [ - ['peer_connection_dev==1', { - 'include_dirs': [ - '<(libjingle_mods)/source', - ], - }], ['inside_chromium_build==1', { 'defines': [ 'NO_SOUND_SYSTEM', @@ -479,6 +473,8 @@ 'type': 'static_library', 'sources': [ '<(libjingle_orig)/source/talk/p2p/base/candidate.h', + '<(libjingle_orig)/source/talk/session/phone/channel.cc', + '<(libjingle_orig)/source/talk/session/phone/channel.h', '<(libjingle_orig)/source/talk/p2p/base/common.h', '<(libjingle_orig)/source/talk/p2p/base/constants.cc', '<(libjingle_orig)/source/talk/p2p/base/constants.h', @@ -556,6 +552,8 @@ '<(libjingle_orig)/source/talk/session/phone/mediamessages.h', '<(libjingle_orig)/source/talk/session/phone/mediamonitor.cc', '<(libjingle_orig)/source/talk/session/phone/mediamonitor.h', + '<(libjingle_orig)/source/talk/session/phone/mediasession.cc', + '<(libjingle_orig)/source/talk/session/phone/mediasession.h', '<(libjingle_orig)/source/talk/session/phone/mediasessionclient.cc', '<(libjingle_orig)/source/talk/session/phone/mediasessionclient.h', '<(libjingle_orig)/source/talk/session/phone/mediasink.h', @@ -630,24 +628,6 @@ 'libjingle', ], } ], # inside_chromium_build - ['peer_connection_dev==1', { - 'sources': [ - '<(libjingle_mods)/source/talk/base/refcount.h', - '<(libjingle_mods)/source/talk/base/scoped_refptr.h', - '<(libjingle_mods)/source/talk/p2p/client/fakeportallocator.h', - '<(libjingle_mods)/source/talk/session/phone/channel.cc', - '<(libjingle_mods)/source/talk/session/phone/channel.h', - '<(libjingle_mods)/source/talk/session/phone/mediasession.cc', - ], - },{ - 'sources': [ - '<(libjingle_orig)/source/talk/session/phone/channel.cc', - '<(libjingle_orig)/source/talk/session/phone/channel.h', - '<(libjingle_orig)/source/talk/session/phone/mediasession.cc', - '<(libjingle_orig)/source/talk/session/phone/mediasession.h', - '<(libjingle_orig)/source/talk/session/phone/sourceparams.h', - ], - }], # peer_connection_dev ], # conditions }, # seperate project for app @@ -689,104 +669,8 @@ '../../src/system_wrappers/source/system_wrappers.gyp:system_wrappers', 'libjingle_p2p', ], - } ], # inside_chromium_build - ['peer_connection_dev==1', { - 'include_dirs': [ - '<(libjingle_mods)/source', - ], - # sources= empties the list of source file and start new. - # peer_connection_dev is independent of the main branch. - 'sources=': [ - '<(libjingle_mods)/source/talk/app/webrtc_dev/audiotrackimpl.h', - '<(libjingle_mods)/source/talk/app/webrtc_dev/audiotrackimpl.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/mediastream.h', - '<(libjingle_mods)/source/talk/app/webrtc_dev/mediastreamhandler.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/mediastreamhandler.h', - '<(libjingle_mods)/source/talk/app/webrtc_dev/mediastreamprovider.h', - '<(libjingle_mods)/source/talk/app/webrtc_dev/mediastreamimpl.h', - '<(libjingle_mods)/source/talk/app/webrtc_dev/mediastreamimpl.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/mediastreamtrackproxy.h', - '<(libjingle_mods)/source/talk/app/webrtc_dev/mediastreamtrackproxy.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/mediastreamproxy.h', - '<(libjingle_mods)/source/talk/app/webrtc_dev/mediastreamproxy.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/mediatrackimpl.h', - '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnection.h', - '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionimpl.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionimpl.h', - '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionfactoryimpl.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionfactoryimpl.h', - '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionmessage.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionmessage.h', - '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionsignaling.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionsignaling.h', - '<(libjingle_mods)/source/talk/app/webrtc_dev/sessiondescriptionprovider.h' - '<(libjingle_mods)/source/talk/app/webrtc_dev/streamcollectionimpl.h', - '<(libjingle_mods)/source/talk/app/webrtc_dev/videorendererimpl.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/videotrackimpl.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/videotrackimpl.h', - '<(libjingle_mods)/source/talk/app/webrtc_dev/webrtcjson.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/webrtcjson.h', - '<(libjingle_mods)/source/talk/app/webrtc_dev/webrtcsessionobserver', - '<(libjingle_mods)/source/talk/app/webrtc_dev/webrtcsession.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/webrtcsession.h', - ], - }], # peer_connection_dev + } ], # inside_chromium_build ], # conditions - }, - { - 'target_name': 'peerconnection_unittests', - 'conditions': [ - ['peer_connection_dev==1', { - 'dependencies': [ - 'libjingle_app', - '../../testing/gmock.gyp:gmock', - '../../testing/gtest.gyp:gtest', - '../../testing/gtest.gyp:gtest_main', - # TODO(perkj): Temporary build the client app here to make sure - # nothing is broken. - 'source/talk/examples/peerconnection_client/' - 'peerconnection_client.gyp:peerconnection_client_dev', - ], - 'type': 'executable', - 'conditions': [ - ['inside_chromium_build==1', { - 'dependencies': [ - '../../third_party/webrtc/modules/modules.gyp:audio_device', - '../../third_party/webrtc/modules/modules.gyp:video_capture_module', - '../../third_party/webrtc/system_wrappers/source/system_wrappers.gyp:system_wrappers', - ], - }, { - 'dependencies': [ - '../../src/modules/modules.gyp:audio_device', - '../../src/modules/modules.gyp:video_capture_module', - '../../src/system_wrappers/source/system_wrappers.gyp:system_wrappers', - ], - }], - ['OS=="linux"', { - 'libraries': [ - '-lXext', - '-lX11', - ], - }], - ], #conditions - 'sources': [ - '<(libjingle_mods)/source/talk/app/webrtc_dev/test/filevideocapturemodule.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/mediastream_unittest.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/mediastreamhandler_unittest.cc', - # // TODO (henrike): Re add when there is no dependency to foreman.yuv - #'<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnection_unittest.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnection_unittests.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionimpl_unittest.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionfactory_unittest.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionmessage_unittest.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionsignaling_unittest.cc', - '<(libjingle_mods)/source/talk/app/webrtc_dev/webrtcsession_unittest.cc', - ], - } , { - 'type': 'none', - } - ], # peer_connection_dev - ], # conditions - }, + }, ], } diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/audiotrackimpl.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/audiotrackimpl.cc deleted file mode 100644 index 1adc55ecc..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/audiotrackimpl.cc +++ /dev/null @@ -1,71 +0,0 @@ -/* - * libjingle - * Copyright 2004--2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ -#include "talk/app/webrtc_dev/audiotrackimpl.h" - -#include - -namespace webrtc { - -static const char kAudioTrackKind[] = "audio"; - -AudioTrack::AudioTrack(const std::string& label) - : MediaTrack(label), - audio_device_(NULL) { -} - -AudioTrack::AudioTrack(const std::string& label, - AudioDeviceModule* audio_device) - : MediaTrack(label), - audio_device_(audio_device) { -} - - // Get the AudioDeviceModule associated with this track. -AudioDeviceModule* AudioTrack::GetAudioDevice() { - return audio_device_.get(); -} - - // Implement MediaStreamTrack -std::string AudioTrack::kind() const { - return std::string(kAudioTrackKind); -} - -talk_base::scoped_refptr AudioTrack::CreateRemote( - const std::string& label) { - talk_base::RefCountedObject* track = - new talk_base::RefCountedObject(label); - return track; -} - -talk_base::scoped_refptr AudioTrack::CreateLocal( - const std::string& label, - AudioDeviceModule* audio_device) { - talk_base::RefCountedObject* track = - new talk_base::RefCountedObject(label, audio_device); - return track; -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/audiotrackimpl.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/audiotrackimpl.h deleted file mode 100644 index bc89ffb2d..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/audiotrackimpl.h +++ /dev/null @@ -1,70 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#ifndef TALK_APP_WEBRTC_AUDIOTRACKIMPL_H_ -#define TALK_APP_WEBRTC_AUDIOTRACKIMPL_H_ - -#include "talk/app/webrtc_dev/mediastream.h" -#include "talk/app/webrtc_dev/mediatrackimpl.h" -#include "talk/app/webrtc_dev/notifierimpl.h" -#include "talk/base/scoped_refptr.h" - -#ifdef WEBRTC_RELATIVE_PATH -#include "modules/audio_device/main/interface/audio_device.h" -#else -#include "third_party/webrtc/files/include/audio_device.h" -#endif - -namespace webrtc { - -class AudioTrack : public MediaTrack { - public: - // Creates a remote audio track. - static talk_base::scoped_refptr CreateRemote( - const std::string& label); - // Creates a local audio track. - static talk_base::scoped_refptr CreateLocal( - const std::string& label, - AudioDeviceModule* audio_device); - - // Get the AudioDeviceModule associated with this track. - virtual AudioDeviceModule* GetAudioDevice(); - - // Implement MediaStreamTrack - virtual std::string kind() const; - - protected: - explicit AudioTrack(const std::string& label); - AudioTrack(const std::string& label, AudioDeviceModule* audio_device); - - private: - talk_base::scoped_refptr audio_device_; -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_AUDIOTRACKIMPL_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastream.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastream.h deleted file mode 100644 index 2f1b8cc82..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastream.h +++ /dev/null @@ -1,189 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -// This file contains interfaces for MediaStream and MediaTrack. These -// interfaces are used for implementing MediaStream and MediaTrack as defined -// in http://dev.w3.org/2011/webrtc/editor/webrtc.html#stream-api. These -// interfaces must be used only with PeerConnection. PeerConnectionManager -// interface provides the factory methods to create MediaStream and MediaTracks. - -#ifndef TALK_APP_WEBRTC_MEDIASTREAM_H_ -#define TALK_APP_WEBRTC_MEDIASTREAM_H_ - -#include - -#include "talk/base/basictypes.h" -#include "talk/base/refcount.h" -#include "talk/base/scoped_refptr.h" - -namespace cricket { -class VideoRenderer; -class MediaEngine; -} // namespace cricket - -namespace webrtc { - -class AudioDeviceModule; -class VideoCaptureModule; - -// Generic observer interface. -class ObserverInterface { - public: - virtual void OnChanged() = 0; - - protected: - virtual ~ObserverInterface() {} -}; - -class NotifierInterface { - public: - virtual void RegisterObserver(ObserverInterface* observer) = 0; - virtual void UnregisterObserver(ObserverInterface* observer) = 0; - - virtual ~NotifierInterface() {} -}; - -// Information about a track. -class MediaStreamTrackInterface : public talk_base::RefCountInterface, - public NotifierInterface { - public: - enum TrackState { - kInitializing, // Track is beeing negotiated. - kLive = 1, // Track alive - kEnded = 2, // Track have ended - kFailed = 3, // Track negotiation failed. - }; - - virtual std::string kind() const = 0; - virtual std::string label() const = 0; - virtual bool enabled() const = 0; - virtual TrackState state() const = 0; - virtual bool set_enabled(bool enable) = 0; - // These methods should be called by implementation only. - virtual bool set_state(TrackState new_state) = 0; -}; - -// Reference counted wrapper for a VideoRenderer. -class VideoRendererWrapperInterface : public talk_base::RefCountInterface { - public: - virtual cricket::VideoRenderer* renderer() = 0; - - protected: - virtual ~VideoRendererWrapperInterface() {} -}; - -// Creates a reference counted object of type cricket::VideoRenderer. -// webrtc::VideoRendererWrapperInterface take ownership of -// cricket::VideoRenderer. -talk_base::scoped_refptr CreateVideoRenderer( - cricket::VideoRenderer* renderer); - -class VideoTrackInterface : public MediaStreamTrackInterface { - public: - // Set the video renderer for a local or remote stream. - // This call will start decoding the received video stream and render it. - // The VideoRendererInterface is stored as a scoped_refptr. This means that - // it is not allowed to call delete renderer after this API has been called. - virtual void SetRenderer(VideoRendererWrapperInterface* renderer) = 0; - - // Get the VideoRenderer associated with this track. - virtual VideoRendererWrapperInterface* GetRenderer() = 0; - - protected: - virtual ~VideoTrackInterface() {} -}; - -class LocalVideoTrackInterface : public VideoTrackInterface { - public: - // Get the VideoCapture device associated with this track. - // TODO(mallinath) - Update with VideoCapturerWrapper to support both - // cricket and webrtc capture interface. - virtual VideoCaptureModule* GetVideoCapture() = 0; - - protected: - virtual ~LocalVideoTrackInterface() {} -}; - -class AudioTrackInterface : public MediaStreamTrackInterface { - public: - protected: - virtual ~AudioTrackInterface() {} -}; - -class LocalAudioTrackInterface : public AudioTrackInterface { - public: - // Get the AudioDeviceModule associated with this track. - virtual AudioDeviceModule* GetAudioDevice() = 0; - protected: - virtual ~LocalAudioTrackInterface() {} -}; - -// List of of tracks. -template -class MediaStreamTrackListInterface : public talk_base::RefCountInterface { - public: - virtual size_t count() = 0; - virtual TrackType* at(size_t index) = 0; - - protected: - virtual ~MediaStreamTrackListInterface() {} -}; - -typedef MediaStreamTrackListInterface AudioTracks; -typedef MediaStreamTrackListInterface VideoTracks; - -class MediaStreamInterface : public talk_base::RefCountInterface, - public NotifierInterface { - public: - virtual std::string label() const = 0; - virtual AudioTracks* audio_tracks() = 0; - virtual VideoTracks* video_tracks() = 0; - - enum ReadyState { - kInitializing, - kLive = 1, // Stream alive - kEnded = 2, // Stream have ended - }; - - virtual ReadyState ready_state() = 0; - - // These methods should be called by implementation only. - virtual void set_ready_state(ReadyState state) = 0; - - protected: - virtual ~MediaStreamInterface() {} -}; - -class LocalMediaStreamInterface : public MediaStreamInterface { - public: - virtual bool AddTrack(AudioTrackInterface* track) = 0; - virtual bool AddTrack(VideoTrackInterface* track) = 0; -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_MEDIASTREAM_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastream_unittest.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastream_unittest.cc deleted file mode 100644 index 9650590d2..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastream_unittest.cc +++ /dev/null @@ -1,389 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include - -#include "gmock/gmock.h" -#include "gtest/gtest.h" -#include "talk/app/webrtc_dev/mediastreamproxy.h" -#include "talk/app/webrtc_dev/mediastreamtrackproxy.h" -#include "talk/base/refcount.h" -#include "talk/base/scoped_ptr.h" -#include "talk/base/thread.h" - -static const char kStreamLabel1[] = "local_stream_1"; -static const char kVideoTrackLabel[] = "dummy_video_cam_1"; -static const char kAudioTrackLabel[] = "dummy_microphone_1"; - -using talk_base::scoped_refptr; -using ::testing::Exactly; - -namespace { - -class ReadyStateMessageData : public talk_base::MessageData { - public: - ReadyStateMessageData( - webrtc::MediaStreamInterface* stream, - webrtc::MediaStreamInterface::ReadyState new_state) - : stream_(stream), - ready_state_(new_state) { - } - - scoped_refptr stream_; - webrtc::MediaStreamInterface::ReadyState ready_state_; -}; - -class TrackStateMessageData : public talk_base::MessageData { - public: - TrackStateMessageData( - webrtc::MediaStreamTrackInterface* track, - webrtc::MediaStreamTrackInterface::TrackState state) - : track_(track), - state_(state) { - } - - scoped_refptr track_; - webrtc::MediaStreamTrackInterface::TrackState state_; -}; - -} // namespace anonymous - -namespace webrtc { - -// Helper class to test Observer. -class MockObserver : public ObserverInterface { - public: - explicit MockObserver(talk_base::Thread* signaling_thread) - : signaling_thread_(signaling_thread) { - } - - MOCK_METHOD0(DoOnChanged, void()); - virtual void OnChanged() { - ASSERT_TRUE(talk_base::Thread::Current() == signaling_thread_); - DoOnChanged(); - } - private: - talk_base::Thread* signaling_thread_; -}; - -class MockMediaStream: public LocalMediaStreamInterface { - public: - MockMediaStream(const std::string& label, talk_base::Thread* signaling_thread) - : stream_impl_(MediaStream::Create(label)), - signaling_thread_(signaling_thread) { - } - virtual void RegisterObserver(webrtc::ObserverInterface* observer) { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - stream_impl_->RegisterObserver(observer); - } - virtual void UnregisterObserver(webrtc::ObserverInterface* observer) { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - stream_impl_->UnregisterObserver(observer); - } - virtual std::string label() const { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - return stream_impl_->label(); - } - virtual AudioTracks* audio_tracks() { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - return stream_impl_->audio_tracks(); - } - virtual VideoTracks* video_tracks() { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - return stream_impl_->video_tracks(); - } - virtual ReadyState ready_state() { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - return stream_impl_->ready_state(); - } - virtual void set_ready_state(ReadyState state) { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - return stream_impl_->set_ready_state(state); - } - virtual bool AddTrack(AudioTrackInterface* audio_track) { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - return stream_impl_->AddTrack(audio_track); - } - virtual bool AddTrack(VideoTrackInterface* video_track) { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - return stream_impl_->AddTrack(video_track); - } - - private: - scoped_refptr stream_impl_; - talk_base::Thread* signaling_thread_; -}; - -template -class MockMediaStreamTrack: public T { - public: - MockMediaStreamTrack(T* implementation, - talk_base::Thread* signaling_thread) - : track_impl_(implementation), - signaling_thread_(signaling_thread) { - } - virtual void RegisterObserver(webrtc::ObserverInterface* observer) { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - track_impl_->RegisterObserver(observer); - } - virtual void UnregisterObserver(webrtc::ObserverInterface* observer) { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - track_impl_->UnregisterObserver(observer); - } - virtual std::string kind() const { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - return track_impl_->kind(); - } - virtual std::string label() const { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - return track_impl_->label(); - } - virtual bool enabled() const { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - return track_impl_->enabled(); - } - virtual MediaStreamTrackInterface::TrackState state() const { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - return track_impl_->state(); - } - virtual bool set_enabled(bool enabled) { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - return track_impl_->set_enabled(enabled); - } - virtual bool set_state(webrtc::MediaStreamTrackInterface::TrackState state) { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - return track_impl_->set_state(state); - } - - protected: - scoped_refptr track_impl_; - talk_base::Thread* signaling_thread_; -}; - -class MockLocalVideoTrack - : public MockMediaStreamTrack { - public: - MockLocalVideoTrack(LocalVideoTrackInterface* implementation, - talk_base::Thread* signaling_thread) - : MockMediaStreamTrack(implementation, - signaling_thread) { - } - virtual void SetRenderer(webrtc::VideoRendererWrapperInterface* renderer) { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - track_impl_->SetRenderer(renderer); - } - virtual VideoRendererWrapperInterface* GetRenderer() { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - return track_impl_->GetRenderer(); - } - virtual VideoCaptureModule* GetVideoCapture() { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - return track_impl_->GetVideoCapture(); - } -}; - -class MockLocalAudioTrack - : public MockMediaStreamTrack { - public: - MockLocalAudioTrack(LocalAudioTrackInterface* implementation, - talk_base::Thread* signaling_thread) - : MockMediaStreamTrack(implementation, - signaling_thread) { - } - - virtual AudioDeviceModule* GetAudioDevice() { - EXPECT_EQ(talk_base::Thread::Current(), signaling_thread_); - return track_impl_->GetAudioDevice(); - } -}; - -class MediaStreamTest: public testing::Test, - public talk_base::MessageHandler { - protected: - virtual void SetUp() { - signaling_thread_ .reset(new talk_base::Thread()); - ASSERT_TRUE(signaling_thread_->Start()); - - std::string label(kStreamLabel1); - // Create a stream proxy object that uses our mocked - // version of a LocalMediaStream. - scoped_refptr mock_stream( - new talk_base::RefCountedObject(label, - signaling_thread_.get())); - stream_ = MediaStreamProxy::Create(label, signaling_thread_.get(), - mock_stream); - ASSERT_TRUE(stream_.get()); - EXPECT_EQ(label, stream_->label()); - EXPECT_EQ(MediaStreamInterface::kInitializing, stream_->ready_state()); - - // Create a video track proxy object that uses our mocked - // version of a LocalVideoTrack - scoped_refptr video_track_impl( - VideoTrack::CreateLocal(kVideoTrackLabel, NULL)); - scoped_refptr mock_videotrack( - new talk_base::RefCountedObject(video_track_impl, - signaling_thread_.get())); - video_track_ = VideoTrackProxy::CreateLocal(mock_videotrack, - signaling_thread_.get()); - - ASSERT_TRUE(video_track_.get()); - EXPECT_EQ(MediaStreamTrackInterface::kInitializing, video_track_->state()); - - // Create an audio track proxy object that uses our mocked - // version of a LocalAudioTrack - scoped_refptr audio_track_impl( - AudioTrack::CreateLocal(kAudioTrackLabel, NULL)); - scoped_refptr mock_audiotrack( - new talk_base::RefCountedObject(audio_track_impl, - signaling_thread_.get())); - audio_track_ = AudioTrackProxy::CreateLocal(mock_audiotrack, - signaling_thread_.get()); - - ASSERT_TRUE(audio_track_.get()); - EXPECT_EQ(MediaStreamTrackInterface::kInitializing, audio_track_->state()); - } - - enum { - MSG_SET_READYSTATE, - MSG_SET_TRACKSTATE, - }; - - // Set the ready state on the signaling thread. - // State can only be changed on the signaling thread. - void SetReadyState(MediaStreamInterface* stream, - MediaStreamInterface::ReadyState new_state) { - ReadyStateMessageData state(stream, new_state); - signaling_thread_->Send(this, MSG_SET_READYSTATE, &state); - } - - // Set the track state on the signaling thread. - // State can only be changed on the signaling thread. - void SetTrackState(MediaStreamTrackInterface* track, - MediaStreamTrackInterface::TrackState new_state) { - TrackStateMessageData state(track, new_state); - signaling_thread_->Send(this, MSG_SET_TRACKSTATE, &state); - } - - talk_base::scoped_ptr signaling_thread_; - scoped_refptr stream_; - scoped_refptr video_track_; - scoped_refptr audio_track_; - - private: - // Implements talk_base::MessageHandler. - virtual void OnMessage(talk_base::Message* msg) { - switch (msg->message_id) { - case MSG_SET_READYSTATE: { - ReadyStateMessageData* state = - static_cast(msg->pdata); - state->stream_->set_ready_state(state->ready_state_); - break; - } - case MSG_SET_TRACKSTATE: { - TrackStateMessageData* state = - static_cast(msg->pdata); - state->track_->set_state(state->state_); - break; - } - default: - break; - } - } -}; - -TEST_F(MediaStreamTest, CreateLocalStream) { - EXPECT_TRUE(stream_->AddTrack(video_track_)); - EXPECT_TRUE(stream_->AddTrack(audio_track_)); - - ASSERT_EQ(1u, stream_->video_tracks()->count()); - ASSERT_EQ(1u, stream_->audio_tracks()->count()); - - // Verify the video track. - scoped_refptr track( - stream_->video_tracks()->at(0)); - EXPECT_EQ(0, track->label().compare(kVideoTrackLabel)); - EXPECT_TRUE(track->enabled()); - - // Verify the audio track. - track = stream_->audio_tracks()->at(0); - EXPECT_EQ(0, track->label().compare(kAudioTrackLabel)); - EXPECT_TRUE(track->enabled()); -} - -TEST_F(MediaStreamTest, ChangeStreamState) { - MockObserver observer(signaling_thread_.get()); - stream_->RegisterObserver(&observer); - - EXPECT_CALL(observer, DoOnChanged()) - .Times(Exactly(1)); - SetReadyState(stream_, MediaStreamInterface::kLive); - - EXPECT_EQ(MediaStreamInterface::kLive, stream_->ready_state()); - // It should not be possible to add - // streams when the state has changed to live. - EXPECT_FALSE(stream_->AddTrack(audio_track_)); - EXPECT_EQ(0u, stream_->audio_tracks()->count()); -} - -TEST_F(MediaStreamTest, ChangeVideoTrack) { - MockObserver observer(signaling_thread_.get()); - video_track_->RegisterObserver(&observer); - - EXPECT_CALL(observer, DoOnChanged()) - .Times(Exactly(1)); - video_track_->set_enabled(false); - EXPECT_FALSE(video_track_->state()); - - EXPECT_CALL(observer, DoOnChanged()) - .Times(Exactly(1)); - SetTrackState(video_track_, MediaStreamTrackInterface::kLive); - EXPECT_EQ(MediaStreamTrackInterface::kLive, video_track_->state()); - - EXPECT_CALL(observer, DoOnChanged()) - .Times(Exactly(1)); - scoped_refptr renderer( - CreateVideoRenderer(NULL)); - video_track_->SetRenderer(renderer.get()); - EXPECT_TRUE(renderer.get() == video_track_->GetRenderer()); -} - -TEST_F(MediaStreamTest, ChangeAudioTrack) { - MockObserver observer(signaling_thread_.get()); - audio_track_->RegisterObserver(&observer); - - EXPECT_CALL(observer, DoOnChanged()) - .Times(Exactly(1)); - audio_track_->set_enabled(false); - EXPECT_FALSE(audio_track_->enabled()); - - EXPECT_CALL(observer, DoOnChanged()) - .Times(Exactly(1)); - SetTrackState(audio_track_, MediaStreamTrackInterface::kLive); - EXPECT_EQ(MediaStreamTrackInterface::kLive, audio_track_->state()); -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamhandler.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamhandler.cc deleted file mode 100644 index 0778fb253..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamhandler.cc +++ /dev/null @@ -1,255 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include "talk/app/webrtc_dev/mediastreamhandler.h" - -#ifdef WEBRTC_RELATIVE_PATH -#include "modules/video_capture/main/interface/video_capture.h" -#else -#include "third_party/webrtc/files/include/video_capture.h" -#endif - -namespace webrtc { - -VideoTrackHandler::VideoTrackHandler(VideoTrackInterface* track, - MediaProviderInterface* provider) - : provider_(provider), - video_track_(track), - state_(track->state()), - enabled_(track->enabled()), - renderer_(track->GetRenderer()) { - video_track_->RegisterObserver(this); -} - -VideoTrackHandler::~VideoTrackHandler() { - video_track_->UnregisterObserver(this); -} - -void VideoTrackHandler::OnChanged() { - if (state_ != video_track_->state()) { - state_ = video_track_->state(); - OnStateChanged(); - } - if (renderer_.get() != video_track_->GetRenderer()) { - renderer_ = video_track_->GetRenderer(); - OnRendererChanged(); - } - if (enabled_ != video_track_->enabled()) { - enabled_ = video_track_->enabled(); - OnEnabledChanged(); - } -} - -LocalVideoTrackHandler::LocalVideoTrackHandler( - LocalVideoTrackInterface* track, - MediaProviderInterface* provider) - : VideoTrackHandler(track, provider), - local_video_track_(track) { -} - -LocalVideoTrackHandler::~LocalVideoTrackHandler() { - // Since cricket::VideoRenderer is not reference counted - // we need to remove the renderer before we are deleted. - provider_->SetLocalRenderer(video_track_->label(), NULL); -} - -void LocalVideoTrackHandler::OnRendererChanged() { - VideoRendererWrapperInterface* renderer(video_track_->GetRenderer()); - if (renderer) - provider_->SetLocalRenderer(video_track_->label(), renderer->renderer()); - else - provider_->SetLocalRenderer(video_track_->label(), NULL); -} - -void LocalVideoTrackHandler::OnStateChanged() { - if (local_video_track_->state() == VideoTrackInterface::kLive) { - provider_->SetCaptureDevice(local_video_track_->label(), - local_video_track_->GetVideoCapture()); - VideoRendererWrapperInterface* renderer(video_track_->GetRenderer()); - if (renderer) - provider_->SetLocalRenderer(video_track_->label(), renderer->renderer()); - else - provider_->SetLocalRenderer(video_track_->label(), NULL); - } -} - -void LocalVideoTrackHandler::OnEnabledChanged() { - // TODO(perkj) What should happen when enabled is changed? -} - -RemoteVideoTrackHandler::RemoteVideoTrackHandler( - VideoTrackInterface* track, - MediaProviderInterface* provider) - : VideoTrackHandler(track, provider), - remote_video_track_(track) { -} - -RemoteVideoTrackHandler::~RemoteVideoTrackHandler() { - // Since cricket::VideoRenderer is not reference counted - // we need to remove the renderer before we are deleted. - provider_->SetRemoteRenderer(video_track_->label(), NULL); -} - - -void RemoteVideoTrackHandler::OnRendererChanged() { - VideoRendererWrapperInterface* renderer(video_track_->GetRenderer()); - if (renderer) - provider_->SetRemoteRenderer(video_track_->label(), renderer->renderer()); - else - provider_->SetRemoteRenderer(video_track_->label(), NULL); -} - -void RemoteVideoTrackHandler::OnStateChanged() { -} - -void RemoteVideoTrackHandler::OnEnabledChanged() { - // TODO(perkj): What should happen when enabled is changed? -} - -MediaStreamHandler::MediaStreamHandler(MediaStreamInterface* stream, - MediaProviderInterface* provider) - : stream_(stream), - provider_(provider) { -} - -MediaStreamHandler::~MediaStreamHandler() { - for (VideoTrackHandlers::iterator it = video_handlers_.begin(); - it != video_handlers_.end(); ++it) { - delete *it; - } -} - -MediaStreamInterface* MediaStreamHandler::stream() { - return stream_.get(); -} - -void MediaStreamHandler::OnChanged() { - // TODO(perkj): Implement state change and enabled changed. -} - - -LocalMediaStreamHandler::LocalMediaStreamHandler( - MediaStreamInterface* stream, - MediaProviderInterface* provider) - : MediaStreamHandler(stream, provider) { - VideoTracks* tracklist(stream->video_tracks()); - - for (size_t j = 0; j < tracklist->count(); ++j) { - LocalVideoTrackInterface* track = - static_cast(tracklist->at(j)); - VideoTrackHandler* handler(new LocalVideoTrackHandler(track, provider)); - video_handlers_.push_back(handler); - } -} - -RemoteMediaStreamHandler::RemoteMediaStreamHandler( - MediaStreamInterface* stream, - MediaProviderInterface* provider) - : MediaStreamHandler(stream, provider) { - VideoTracks* tracklist(stream->video_tracks()); - - for (size_t j = 0; j < tracklist->count(); ++j) { - VideoTrackInterface* track = - static_cast(tracklist->at(j)); - VideoTrackHandler* handler(new RemoteVideoTrackHandler(track, provider)); - video_handlers_.push_back(handler); - } -} - -MediaStreamHandlers::MediaStreamHandlers(MediaProviderInterface* provider) - : provider_(provider) { -} - -MediaStreamHandlers::~MediaStreamHandlers() { - for (StreamHandlerList::iterator it = remote_streams_handlers_.begin(); - it != remote_streams_handlers_.end(); ++it) { - delete *it; - } - for (StreamHandlerList::iterator it = local_streams_handlers_.begin(); - it != local_streams_handlers_.end(); ++it) { - delete *it; - } -} - -void MediaStreamHandlers::AddRemoteStream(MediaStreamInterface* stream) { - RemoteMediaStreamHandler* handler = new RemoteMediaStreamHandler(stream, - provider_); - remote_streams_handlers_.push_back(handler); -} - -void MediaStreamHandlers::RemoveRemoteStream(MediaStreamInterface* stream) { - StreamHandlerList::iterator it = remote_streams_handlers_.begin(); - for (; it != remote_streams_handlers_.end(); ++it) { - if ((*it)->stream() == stream) { - delete *it; - break; - } - } - ASSERT(it != remote_streams_handlers_.end()); - remote_streams_handlers_.erase(it); -} - -void MediaStreamHandlers::CommitLocalStreams( - StreamCollectionInterface* streams) { - // Iterate the old list of local streams. - // If its not found in the new collection it have been removed. - // We can not erase from the old collection at the same time as we iterate. - // That is what the ugly while(1) fix. - while (1) { - StreamHandlerList::iterator it = local_streams_handlers_.begin(); - for (; it != local_streams_handlers_.end(); ++it) { - if (streams->find((*it)->stream()->label()) == NULL) { - delete *it; - break; - } - } - if (it != local_streams_handlers_.end()) { - local_streams_handlers_.erase(it); - continue; - } - break; - } - - // Iterate the new collection of local streams. - // If its not found in the old collection it have been added. - for (size_t j = 0; j < streams->count(); ++j) { - MediaStreamInterface* stream = streams->at(j); - StreamHandlerList::iterator it = local_streams_handlers_.begin(); - for (; it != local_streams_handlers_.end(); ++it) { - if (stream == (*it)->stream()) - break; - } - if (it == local_streams_handlers_.end()) { - LocalMediaStreamHandler* handler = new LocalMediaStreamHandler( - stream, provider_); - local_streams_handlers_.push_back(handler); - } - } -}; - - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamhandler.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamhandler.h deleted file mode 100644 index eabc552d5..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamhandler.h +++ /dev/null @@ -1,144 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -// This file contains classes for listening on changes on MediaStreams and -// MediaTracks and making sure appropriate action is taken. -// Example: If a user sets a rendererer on a local video track the renderer is -// connected to the appropriate camera. - -#ifndef TALK_APP_WEBRTC_DEV_MEDIASTREAMHANDLER_H_ -#define TALK_APP_WEBRTC_DEV_MEDIASTREAMHANDLER_H_ - -#include -#include - -#include "talk/app/webrtc_dev/mediastream.h" -#include "talk/app/webrtc_dev/mediastreamprovider.h" -#include "talk/app/webrtc_dev/peerconnection.h" -#include "talk/base/thread.h" - -namespace webrtc { - -// VideoTrackHandler listen to events on a VideoTrack instance and -// executes the requested change. -class VideoTrackHandler : public ObserverInterface { - public: - VideoTrackHandler(VideoTrackInterface* track, - MediaProviderInterface* provider); - virtual ~VideoTrackHandler(); - virtual void OnChanged(); - - protected: - virtual void OnRendererChanged() = 0; - virtual void OnStateChanged() = 0; - virtual void OnEnabledChanged() = 0; - - MediaProviderInterface* provider_; - VideoTrackInterface* video_track_; - - private: - MediaStreamTrackInterface::TrackState state_; - bool enabled_; - talk_base::scoped_refptr renderer_; -}; - -class LocalVideoTrackHandler : public VideoTrackHandler { - public: - LocalVideoTrackHandler(LocalVideoTrackInterface* track, - MediaProviderInterface* provider); - virtual ~LocalVideoTrackHandler(); - - protected: - virtual void OnRendererChanged(); - virtual void OnStateChanged(); - virtual void OnEnabledChanged(); - - private: - talk_base::scoped_refptr local_video_track_; -}; - -class RemoteVideoTrackHandler : public VideoTrackHandler { - public: - RemoteVideoTrackHandler(VideoTrackInterface* track, - MediaProviderInterface* provider); - virtual ~RemoteVideoTrackHandler(); - - protected: - virtual void OnRendererChanged(); - virtual void OnStateChanged(); - virtual void OnEnabledChanged(); - - private: - talk_base::scoped_refptr remote_video_track_; -}; - -class MediaStreamHandler : public ObserverInterface { - public: - MediaStreamHandler(MediaStreamInterface* stream, - MediaProviderInterface* provider); - ~MediaStreamHandler(); - MediaStreamInterface* stream(); - virtual void OnChanged(); - - protected: - MediaProviderInterface* provider_; - typedef std::vector VideoTrackHandlers; - VideoTrackHandlers video_handlers_; - talk_base::scoped_refptr stream_; -}; - -class LocalMediaStreamHandler : public MediaStreamHandler { - public: - LocalMediaStreamHandler(MediaStreamInterface* stream, - MediaProviderInterface* provider); -}; - -class RemoteMediaStreamHandler : public MediaStreamHandler { - public: - RemoteMediaStreamHandler(MediaStreamInterface* stream, - MediaProviderInterface* provider); -}; - -class MediaStreamHandlers { - public: - explicit MediaStreamHandlers(MediaProviderInterface* provider); - ~MediaStreamHandlers(); - void AddRemoteStream(MediaStreamInterface* stream); - void RemoveRemoteStream(MediaStreamInterface* stream); - void CommitLocalStreams(StreamCollectionInterface* streams); - - private: - typedef std::list StreamHandlerList; - StreamHandlerList local_streams_handlers_; - StreamHandlerList remote_streams_handlers_; - MediaProviderInterface* provider_; -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_ - diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamhandler_unittest.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamhandler_unittest.cc deleted file mode 100644 index f12677728..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamhandler_unittest.cc +++ /dev/null @@ -1,147 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include - -#include "gmock/gmock.h" -#include "gtest/gtest.h" -#include "talk/app/webrtc_dev/mediastreamimpl.h" -#include "talk/app/webrtc_dev/videotrackimpl.h" -#include "talk/app/webrtc_dev/mediastreamhandler.h" -#include "talk/app/webrtc_dev/streamcollectionimpl.h" -#include "talk/base/thread.h" - -using ::testing::Exactly; - -static const char kStreamLabel1[] = "local_stream_1"; -static const char kVideoDeviceName[] = "dummy_video_cam_1"; - -namespace webrtc { - -// Helper class to test MediaStreamHandler. -class MockMediaProvier : public MediaProviderInterface { - public: - MOCK_METHOD1(SetCaptureDevice, void(const std::string& name)); - MOCK_METHOD1(SetLocalRenderer, void(const std::string& name)); - MOCK_METHOD1(SetRemoteRenderer, void(const std::string& name)); - - virtual void SetCaptureDevice(const std::string& name, - VideoCaptureModule* camera) { - SetCaptureDevice(name); - } - virtual void SetLocalRenderer(const std::string& name, - cricket::VideoRenderer* renderer) { - SetLocalRenderer(name); - } - - virtual void SetRemoteRenderer(const std::string& name, - cricket::VideoRenderer* renderer) { - SetRemoteRenderer(name); - } - ~MockMediaProvier() {} -}; - -TEST(MediaStreamHandlerTest, LocalStreams) { - // Create a local stream. - std::string label(kStreamLabel1); - talk_base::scoped_refptr stream( - MediaStream::Create(label)); - talk_base::scoped_refptr - video_track(VideoTrack::CreateLocal(kVideoDeviceName, NULL)); - EXPECT_TRUE(stream->AddTrack(video_track)); - talk_base::scoped_refptr renderer( - CreateVideoRenderer(NULL)); - video_track->SetRenderer(renderer); - - MockMediaProvier provider; - MediaStreamHandlers handlers(&provider); - - talk_base::scoped_refptr collection( - StreamCollectionImpl::Create()); - collection->AddStream(stream); - - EXPECT_CALL(provider, SetLocalRenderer(kVideoDeviceName)) - .Times(Exactly(2)); // SetLocalRender will also be called from dtor of - // LocalVideoTrackHandler - EXPECT_CALL(provider, SetCaptureDevice(kVideoDeviceName)) - .Times(Exactly(1)); - handlers.CommitLocalStreams(collection); - - video_track->set_state(MediaStreamTrackInterface::kLive); - // Process posted messages. - talk_base::Thread::Current()->ProcessMessages(1); - - collection->RemoveStream(stream); - handlers.CommitLocalStreams(collection); - - video_track->set_state(MediaStreamTrackInterface::kEnded); - // Process posted messages. - talk_base::Thread::Current()->ProcessMessages(1); -} - -TEST(MediaStreamHandlerTest, RemoteStreams) { - // Create a local stream. We use local stream in this test as well because - // they are easier to create. - // LocalMediaStreams inherit from MediaStreams. - std::string label(kStreamLabel1); - talk_base::scoped_refptr stream( - MediaStream::Create(label)); - talk_base::scoped_refptr - video_track(VideoTrack::CreateLocal(kVideoDeviceName, NULL)); - EXPECT_TRUE(stream->AddTrack(video_track)); - - MockMediaProvier provider; - MediaStreamHandlers handlers(&provider); - - handlers.AddRemoteStream(stream); - - EXPECT_CALL(provider, SetRemoteRenderer(kVideoDeviceName)) - .Times(Exactly(3)); // SetRemoteRenderer is also called from dtor of - // RemoteVideoTrackHandler. - - // Set the renderer once. - talk_base::scoped_refptr renderer( - CreateVideoRenderer(NULL)); - video_track->SetRenderer(renderer); - talk_base::Thread::Current()->ProcessMessages(1); - - // Change the already set renderer. - renderer = CreateVideoRenderer(NULL); - video_track->SetRenderer(renderer); - talk_base::Thread::Current()->ProcessMessages(1); - - handlers.RemoveRemoteStream(stream); - - // Change the renderer after the stream have been removed from handler. - // This should not trigger a call to SetRemoteRenderer. - renderer = CreateVideoRenderer(NULL); - video_track->SetRenderer(renderer); - // Process posted messages. - talk_base::Thread::Current()->ProcessMessages(1); -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamimpl.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamimpl.cc deleted file mode 100644 index f6d30a75a..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamimpl.cc +++ /dev/null @@ -1,73 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include "talk/app/webrtc_dev/mediastreamimpl.h" -#include "talk/base/logging.h" - -namespace webrtc { - -talk_base::scoped_refptr MediaStream::Create( - const std::string& label) { - talk_base::RefCountedObject* stream = - new talk_base::RefCountedObject(label); - return stream; -} - -MediaStream::MediaStream(const std::string& label) - : label_(label), - ready_state_(MediaStreamInterface::kInitializing), - audio_track_list_( - new talk_base::RefCountedObject< - MediaStreamTrackList >()), - video_track_list_( - new talk_base::RefCountedObject< - MediaStreamTrackList >()) { -} - -void MediaStream::set_ready_state( - MediaStreamInterface::ReadyState new_state) { - if (ready_state_ != new_state) { - ready_state_ = new_state; - Notifier::FireOnChanged(); - } -} - -bool MediaStream::AddTrack(AudioTrackInterface* track) { - if (ready_state() != kInitializing) - return false; - audio_track_list_->AddTrack(track); - return true; -} - -bool MediaStream::AddTrack(VideoTrackInterface* track) { - if (ready_state() != kInitializing) - return false; - video_track_list_->AddTrack(track); - return true; -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamimpl.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamimpl.h deleted file mode 100644 index f5225c8fa..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamimpl.h +++ /dev/null @@ -1,90 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -// This file contains the implementation of MediaStreamInterface interface. - -#ifndef TALK_APP_WEBRTC_MEDIASTREAMIMPL_H_ -#define TALK_APP_WEBRTC_MEDIASTREAMIMPL_H_ - -#include -#include - -#include "talk/app/webrtc_dev/mediastream.h" -#include "talk/app/webrtc_dev/notifierimpl.h" - -namespace webrtc { -class AudioTrack; -class VideoTrack; - -class MediaStream : public Notifier { - public: - template - class MediaStreamTrackList : public MediaStreamTrackListInterface { - public: - void AddTrack(T* track) { - tracks_.push_back(track); - } - virtual size_t count() { return tracks_.size(); } - virtual T* at(size_t index) { - return tracks_.at(index); - } - - private: - std::vector > tracks_; - }; - - static talk_base::scoped_refptr Create(const std::string& label); - - // Implement LocalMediaStreamInterface. - virtual bool AddTrack(AudioTrackInterface* track); - virtual bool AddTrack(VideoTrackInterface* track); - // Implement MediaStreamInterface. - virtual std::string label() const { return label_; } - virtual MediaStreamTrackListInterface* audio_tracks() { - return audio_track_list_; - } - virtual MediaStreamTrackListInterface* video_tracks() { - return video_track_list_; - } - virtual ReadyState ready_state() { return ready_state_; } - virtual void set_ready_state(ReadyState new_state); - void set_state(ReadyState new_state); - - protected: - explicit MediaStream(const std::string& label); - - std::string label_; - MediaStreamInterface::ReadyState ready_state_; - talk_base::scoped_refptr > - audio_track_list_; - talk_base::scoped_refptr > - video_track_list_; -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_MEDIASTREAMIMPL_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamimpl_unittest.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamimpl_unittest.cc deleted file mode 100644 index cdf2e372f..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamimpl_unittest.cc +++ /dev/null @@ -1,95 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include - -#include "gtest/gtest.h" -#include "talk/app/webrtc_dev/mediastreamimpl.h" -#include "talk/app/webrtc_dev/videotrackimpl.h" - -static const char kStreamLabel1[] = "local_stream_1"; -static const char kVideoDeviceName[] = "dummy_video_cam_1"; - -namespace webrtc { - -// Helper class to test the Observer. -class TestObserver : public ObserverInterface { - public: - TestObserver() : changed_(0) {} - void OnChanged() { - ++changed_; - } - int NumChanges() { - return changed_; - } - void Reset() { - changed_ = 0; - } - - protected: - int changed_; -}; - -TEST(LocalStreamTest, Create) { - // Create a local stream. - std::string label(kStreamLabel1); - talk_base::scoped_refptr stream( - MediaStream::Create(label)); - - EXPECT_EQ(label, stream->label()); - // Check state. - EXPECT_EQ(MediaStreamInterface::kInitializing, stream->ready_state()); - - // Create a local Video track. - TestObserver tracklist_observer; - talk_base::scoped_refptr - video_track(VideoTrack::CreateLocal(kVideoDeviceName, NULL)); - // Add an observer to the track list. - talk_base::scoped_refptr > - track_list(stream->video_tracks()); - // Add the track to the local stream. - EXPECT_TRUE(stream->AddTrack(video_track)); - EXPECT_EQ(1u, stream->video_tracks()->count()); - - // Verify the track. - talk_base::scoped_refptr track( - stream->video_tracks()->at(0)); - EXPECT_EQ(0, track->label().compare(kVideoDeviceName)); - EXPECT_TRUE(track->enabled()); - - // Verify the Track observer. - TestObserver observer1; - TestObserver observer2; - track->RegisterObserver(&observer1); - track->RegisterObserver(&observer2); - track->set_enabled(false); - EXPECT_EQ(1u, observer1.NumChanges()); - EXPECT_EQ(1u, observer2.NumChanges()); - EXPECT_FALSE(track->enabled()); -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamprovider.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamprovider.h deleted file mode 100644 index 33247dbde..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamprovider.h +++ /dev/null @@ -1,52 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#ifndef TALK_APP_WEBRTC_DEV_MEDIASTREAMPROVIDER_H_ -#define TALK_APP_WEBRTC_DEV_MEDIASTREAMPROVIDER_H_ - -#include "talk/app/webrtc_dev/mediastream.h" - -namespace webrtc { - -// Interface for setting media devices on a certain MediaTrack. -// This interface is called by classes in mediastreamhandler.h to -// set new devices. -class MediaProviderInterface { - public: - virtual void SetCaptureDevice(const std::string& name, - VideoCaptureModule* camera) = 0; - virtual void SetLocalRenderer(const std::string& name, - cricket::VideoRenderer* renderer) = 0; - virtual void SetRemoteRenderer(const std::string& name, - cricket::VideoRenderer* renderer) = 0; - protected: - virtual ~MediaProviderInterface() {} -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_DEV_MEDIASTREAMPROVIDER_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamproxy.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamproxy.cc deleted file mode 100644 index dfb1c9a20..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamproxy.cc +++ /dev/null @@ -1,318 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include "talk/app/webrtc_dev/mediastreamproxy.h" -#include "talk/base/refcount.h" -#include "talk/base/scoped_refptr.h" - -namespace { - -enum { - MSG_SET_TRACKLIST_IMPLEMENTATION = 1, - MSG_REGISTER_OBSERVER, - MSG_UNREGISTER_OBSERVER, - MSG_LABEL, - MSG_ADD_AUDIO_TRACK, - MSG_ADD_VIDEO_TRACK, - MSG_READY_STATE, - MSG_COUNT, - MSG_AT -}; - -typedef talk_base::TypedMessageData LabelMessageData; -typedef talk_base::TypedMessageData SizeTMessageData; -typedef talk_base::TypedMessageData - ObserverMessageData; -typedef talk_base::TypedMessageData - ReadyStateMessageData; - -template -class MediaStreamTrackMessageData : public talk_base::MessageData { - public: - explicit MediaStreamTrackMessageData(T* track) - : track_(track), - result_(false) { - } - - talk_base::scoped_refptr track_; - bool result_; -}; - -typedef MediaStreamTrackMessageData - AudioTrackMsgData; -typedef MediaStreamTrackMessageData - VideoTrackMsgData; - -template -class MediaStreamTrackAtMessageData : public talk_base::MessageData { - public: - explicit MediaStreamTrackAtMessageData(size_t index) - : index_(index) { - } - - size_t index_; - talk_base::scoped_refptr track_; -}; - -class MediaStreamTrackListsMessageData : public talk_base::MessageData { - public: - talk_base::scoped_refptr audio_tracks_; - talk_base::scoped_refptr video_tracks_; -}; - -} // namespace anonymous - -namespace webrtc { - -talk_base::scoped_refptr MediaStreamProxy::Create( - const std::string& label, - talk_base::Thread* signaling_thread) { - ASSERT(signaling_thread); - talk_base::RefCountedObject* stream = - new talk_base::RefCountedObject( - label, signaling_thread, - reinterpret_cast(NULL)); - return stream; -} - -talk_base::scoped_refptr MediaStreamProxy::Create( - const std::string& label, - talk_base::Thread* signaling_thread, - LocalMediaStreamInterface* media_stream_impl) { - ASSERT(signaling_thread); - ASSERT(media_stream_impl); - talk_base::RefCountedObject* stream = - new talk_base::RefCountedObject(label, signaling_thread, - media_stream_impl); - return stream; -} - -MediaStreamProxy::MediaStreamProxy(const std::string& label, - talk_base::Thread* signaling_thread, - LocalMediaStreamInterface* media_stream_impl) - : signaling_thread_(signaling_thread), - media_stream_impl_(media_stream_impl), - audio_tracks_(new talk_base::RefCountedObject< - MediaStreamTrackListProxy >( - signaling_thread_)), - video_tracks_(new talk_base::RefCountedObject< - MediaStreamTrackListProxy >( - signaling_thread_)) { - if (media_stream_impl_ == NULL) { - media_stream_impl_ = MediaStream::Create(label); - } - - MediaStreamTrackListsMessageData tracklists; - Send(MSG_SET_TRACKLIST_IMPLEMENTATION, &tracklists); - audio_tracks_->SetImplementation(tracklists.audio_tracks_); - video_tracks_->SetImplementation(tracklists.video_tracks_); -} - -std::string MediaStreamProxy::label() const { - if (!signaling_thread_->IsCurrent()) { - std::string label; - LabelMessageData msg(&label); - Send(MSG_LABEL, &msg); - return label; - } - return media_stream_impl_->label(); -} - -MediaStreamInterface::ReadyState MediaStreamProxy::ready_state() { - if (!signaling_thread_->IsCurrent()) { - ReadyStateMessageData msg(MediaStreamInterface::kInitializing); - Send(MSG_READY_STATE, &msg); - return msg.data(); - } - return media_stream_impl_->ready_state(); -} - -void MediaStreamProxy::set_ready_state( - MediaStreamInterface::ReadyState new_state) { - if (!signaling_thread_->IsCurrent()) { - // State should only be allowed to be changed from the signaling thread. - ASSERT(!"Not Allowed!"); - return; - } - media_stream_impl_->set_ready_state(new_state); -} - -bool MediaStreamProxy::AddTrack(AudioTrackInterface* track) { - if (!signaling_thread_->IsCurrent()) { - AudioTrackMsgData msg(track); - Send(MSG_ADD_AUDIO_TRACK, &msg); - return msg.result_; - } - return media_stream_impl_->AddTrack(track); -} - -bool MediaStreamProxy::AddTrack(VideoTrackInterface* track) { - if (!signaling_thread_->IsCurrent()) { - VideoTrackMsgData msg(track); - Send(MSG_ADD_VIDEO_TRACK, &msg); - return msg.result_; - } - return media_stream_impl_->AddTrack(track); -} - -void MediaStreamProxy::RegisterObserver(ObserverInterface* observer) { - if (!signaling_thread_->IsCurrent()) { - ObserverMessageData msg(observer); - Send(MSG_REGISTER_OBSERVER, &msg); - return; - } - media_stream_impl_->RegisterObserver(observer); -} - -void MediaStreamProxy::UnregisterObserver(ObserverInterface* observer) { - if (!signaling_thread_->IsCurrent()) { - ObserverMessageData msg(observer); - Send(MSG_UNREGISTER_OBSERVER, &msg); - return; - } - media_stream_impl_->UnregisterObserver(observer); -} - -void MediaStreamProxy::Send(uint32 id, talk_base::MessageData* data) const { - signaling_thread_->Send(const_cast(this), id, - data); -} - -// Implement MessageHandler -void MediaStreamProxy::OnMessage(talk_base::Message* msg) { - talk_base::MessageData* data = msg->pdata; - switch (msg->message_id) { - case MSG_SET_TRACKLIST_IMPLEMENTATION: { - MediaStreamTrackListsMessageData* lists = - static_cast(data); - lists->audio_tracks_ = media_stream_impl_->audio_tracks(); - lists->video_tracks_ = media_stream_impl_->video_tracks(); - break; - } - case MSG_REGISTER_OBSERVER: { - ObserverMessageData* observer = static_cast(data); - media_stream_impl_->RegisterObserver(observer->data()); - break; - } - case MSG_UNREGISTER_OBSERVER: { - ObserverMessageData* observer = static_cast(data); - media_stream_impl_->UnregisterObserver(observer->data()); - break; - } - case MSG_LABEL: { - LabelMessageData * label = static_cast(data); - *(label->data()) = media_stream_impl_->label(); - break; - } - case MSG_ADD_AUDIO_TRACK: { - AudioTrackMsgData * track = - static_cast(data); - track->result_ = media_stream_impl_->AddTrack(track->track_.get()); - break; - } - case MSG_ADD_VIDEO_TRACK: { - VideoTrackMsgData * track = - static_cast(data); - track->result_ = media_stream_impl_->AddTrack(track->track_.get()); - break; - } - case MSG_READY_STATE: { - ReadyStateMessageData* state = static_cast(data); - state->data() = media_stream_impl_->ready_state(); - break; - } - default: - ASSERT(!"Not Implemented!"); - break; - } -} - -template -MediaStreamProxy::MediaStreamTrackListProxy::MediaStreamTrackListProxy( - talk_base::Thread* signaling_thread) - : signaling_thread_(signaling_thread) { -} - -template -void MediaStreamProxy::MediaStreamTrackListProxy::SetImplementation( - MediaStreamTrackListInterface* track_list) { - track_list_ = track_list; -} - -template -size_t MediaStreamProxy::MediaStreamTrackListProxy::count() { - if (!signaling_thread_->IsCurrent()) { - SizeTMessageData msg(0u); - Send(MSG_COUNT, &msg); - return msg.data(); - } - return track_list_->count(); -} - -template -T* MediaStreamProxy::MediaStreamTrackListProxy::at( - size_t index) { - if (!signaling_thread_->IsCurrent()) { - MediaStreamTrackAtMessageData msg(index); - Send(MSG_AT, &msg); - return msg.track_; - } - return track_list_->at(index); -} - -template -void MediaStreamProxy::MediaStreamTrackListProxy::Send( - uint32 id, talk_base::MessageData* data) const { - signaling_thread_->Send( - const_cast*>( - this), id, data); -} - -// Implement MessageHandler -template -void MediaStreamProxy::MediaStreamTrackListProxy::OnMessage( - talk_base::Message* msg) { - talk_base::MessageData* data = msg->pdata; - switch (msg->message_id) { - case MSG_COUNT: { - SizeTMessageData* count = static_cast(data); - count->data() = track_list_->count(); - break; - } - case MSG_AT: { - MediaStreamTrackAtMessageData* track = - static_cast*>(data); - track->track_ = track_list_->at(track->index_); - break; - } - default: - ASSERT(!"Not Implemented!"); - break; - } -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamproxy.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamproxy.h deleted file mode 100644 index cf4000923..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamproxy.h +++ /dev/null @@ -1,111 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#ifndef TALK_APP_WEBRTC_MEDIASTREAMPROXY_H_ -#define TALK_APP_WEBRTC_MEDIASTREAMPROXY_H_ - -#include -#include - -#include "talk/app/webrtc_dev/mediastreamimpl.h" -#include "talk/base/thread.h" - -namespace webrtc { -using talk_base::scoped_refptr; - -// MediaStreamProxy is a proxy for the MediaStream interface. The purpose is -// to make sure MediaStreamImpl is only accessed from the signaling thread. -// It can be used as a proxy for both local and remote MediaStreams. -class MediaStreamProxy : public LocalMediaStreamInterface, - public talk_base::MessageHandler { - public: - static scoped_refptr Create( - const std::string& label, - talk_base::Thread* signaling_thread); - - static scoped_refptr Create( - const std::string& label, - talk_base::Thread* signaling_thread, - LocalMediaStreamInterface* media_stream_impl); - - // Implement LocalStream. - virtual bool AddTrack(AudioTrackInterface* track); - virtual bool AddTrack(VideoTrackInterface* track); - - // Implement MediaStream. - virtual std::string label() const; - virtual AudioTracks* audio_tracks() { - return audio_tracks_; - } - virtual VideoTracks* video_tracks() { - return video_tracks_; - } - virtual ReadyState ready_state(); - virtual void set_ready_state(ReadyState new_state); - - // Implement Notifier - virtual void RegisterObserver(ObserverInterface* observer); - virtual void UnregisterObserver(ObserverInterface* observer); - - protected: - MediaStreamProxy(const std::string& label, - talk_base::Thread* signaling_thread, - LocalMediaStreamInterface* media_stream_impl); - - template - class MediaStreamTrackListProxy : public MediaStreamTrackListInterface, - public talk_base::MessageHandler { - public: - explicit MediaStreamTrackListProxy(talk_base::Thread* signaling_thread); - - void SetImplementation(MediaStreamTrackListInterface* track_list); - virtual size_t count(); - virtual T* at(size_t index); - - private: - void Send(uint32 id, talk_base::MessageData* data) const; - void OnMessage(talk_base::Message* msg); - - talk_base::scoped_refptr > track_list_; - mutable talk_base::Thread* signaling_thread_; - }; - typedef MediaStreamTrackListProxy AudioTrackListProxy; - typedef MediaStreamTrackListProxy VideoTrackListProxy; - - void Send(uint32 id, talk_base::MessageData* data) const; - // Implement MessageHandler. - virtual void OnMessage(talk_base::Message* msg); - - mutable talk_base::Thread* signaling_thread_; - scoped_refptr media_stream_impl_; - scoped_refptr audio_tracks_; - scoped_refptr video_tracks_; -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_MEDIASTREAMPROXY_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamtrackproxy.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamtrackproxy.cc deleted file mode 100644 index 9640832eb..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamtrackproxy.cc +++ /dev/null @@ -1,394 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include "talk/app/webrtc_dev/mediastreamtrackproxy.h" - -namespace { - -enum { - MSG_REGISTER_OBSERVER = 1, - MSG_UNREGISTER_OBSERVER, - MSG_LABEL, - MSG_ENABLED, - MSG_SET_ENABLED, - MSG_STATE, - MSG_GET_AUDIODEVICE, - MSG_GET_VIDEODEVICE, - MSG_GET_VIDEORENDERER, - MSG_SET_VIDEORENDERER, -}; - -typedef talk_base::TypedMessageData LabelMessageData; -typedef talk_base::TypedMessageData - ObserverMessageData; -typedef talk_base::TypedMessageData - TrackStateMessageData; -typedef talk_base::TypedMessageData EnableMessageData; - - -class AudioDeviceMessageData : public talk_base::MessageData { - public: - talk_base::scoped_refptr audio_device_; -}; - -class VideoDeviceMessageData : public talk_base::MessageData { - public: - talk_base::scoped_refptr video_device_; -}; - -class VideoRendererMessageData : public talk_base::MessageData { - public: - talk_base::scoped_refptr - video_renderer_; -}; - -} // namespace anonymous - -namespace webrtc { - -template -MediaStreamTrackProxy::MediaStreamTrackProxy( - talk_base::Thread* signaling_thread) - : signaling_thread_(signaling_thread) { -} - -template -void MediaStreamTrackProxy::Init(MediaStreamTrackInterface* track) { - track_ = track; -} - -template -std::string MediaStreamTrackProxy::kind() const { - return track_->kind(); -} - -template -std::string MediaStreamTrackProxy::label() const { - if (!signaling_thread_->IsCurrent()) { - std::string label; - LabelMessageData msg(&label); - Send(MSG_LABEL, &msg); - return label; - } - return track_->label(); -} - -template -MediaStreamTrackInterface::TrackState MediaStreamTrackProxy::state() const { - if (!signaling_thread_->IsCurrent()) { - TrackStateMessageData msg(MediaStreamTrackInterface::kInitializing); - Send(MSG_STATE, &msg); - return msg.data(); - } - return track_->state(); -} - -template -bool MediaStreamTrackProxy::enabled() const { - if (!signaling_thread_->IsCurrent()) { - EnableMessageData msg(false); - Send(MSG_ENABLED, &msg); - return msg.data(); - } - return track_->enabled(); -} - -template -bool MediaStreamTrackProxy::set_enabled(bool enable) { - if (!signaling_thread_->IsCurrent()) { - EnableMessageData msg(enable); - Send(MSG_SET_ENABLED, &msg); - return msg.data(); - } - return track_->set_enabled(enable); -} - -template -bool MediaStreamTrackProxy::set_state( - MediaStreamTrackInterface::TrackState new_state) { - if (!signaling_thread_->IsCurrent()) { - // State should only be allowed to be changed from the signaling thread. - ASSERT(!"Not Allowed!"); - return false; - } - return track_->set_state(new_state); -} - -template -void MediaStreamTrackProxy::RegisterObserver(ObserverInterface* observer) { - if (!signaling_thread_->IsCurrent()) { - ObserverMessageData msg(observer); - Send(MSG_REGISTER_OBSERVER, &msg); - return; - } - track_->RegisterObserver(observer); -} - -template -void MediaStreamTrackProxy::UnregisterObserver(ObserverInterface* observer) { - if (!signaling_thread_->IsCurrent()) { - ObserverMessageData msg(observer); - Send(MSG_UNREGISTER_OBSERVER, &msg); - return; - } - track_->UnregisterObserver(observer); -} - -template -void MediaStreamTrackProxy::Send(uint32 id, - talk_base::MessageData* data) const { - signaling_thread_->Send(const_cast*>(this), id, - data); -} - -template -bool MediaStreamTrackProxy::HandleMessage(talk_base::Message* msg) { - talk_base::MessageData* data = msg->pdata; - switch (msg->message_id) { - case MSG_REGISTER_OBSERVER: { - ObserverMessageData* observer = static_cast(data); - track_->RegisterObserver(observer->data()); - return true; - break; - } - case MSG_UNREGISTER_OBSERVER: { - ObserverMessageData* observer = static_cast(data); - track_->UnregisterObserver(observer->data()); - return true; - break; - } - case MSG_LABEL: { - LabelMessageData* label = static_cast(data); - *(label->data()) = track_->label(); - return true; - } - case MSG_SET_ENABLED: { - EnableMessageData* enabled = static_cast(data); - enabled->data() = track_->set_enabled(enabled->data()); - return true; - break; - } - case MSG_ENABLED: { - EnableMessageData* enabled = static_cast(data); - enabled->data() = track_->enabled(); - return true; - break; - } - case MSG_STATE: { - TrackStateMessageData* state = static_cast(data); - state->data() = track_->state(); - return true; - break; - } - default: - return false; - } -} - -AudioTrackProxy::AudioTrackProxy(const std::string& label, - talk_base::Thread* signaling_thread) - : MediaStreamTrackProxy(signaling_thread), - audio_track_(AudioTrack::CreateRemote(label)) { - Init(audio_track_); -} - -AudioTrackProxy::AudioTrackProxy(const std::string& label, - AudioDeviceModule* audio_device, - talk_base::Thread* signaling_thread) - : MediaStreamTrackProxy(signaling_thread), - audio_track_(AudioTrack::CreateLocal(label, audio_device)) { - Init(audio_track_); -} - -AudioTrackProxy::AudioTrackProxy(LocalAudioTrackInterface* implementation, - talk_base::Thread* signaling_thread) - : MediaStreamTrackProxy(signaling_thread), - audio_track_(implementation) { - Init(audio_track_); -} - -talk_base::scoped_refptr AudioTrackProxy::CreateRemote( - const std::string& label, - talk_base::Thread* signaling_thread) { - ASSERT(signaling_thread); - talk_base::RefCountedObject* track = - new talk_base::RefCountedObject(label, signaling_thread); - return track; -} - -talk_base::scoped_refptr AudioTrackProxy::CreateLocal( - const std::string& label, - AudioDeviceModule* audio_device, - talk_base::Thread* signaling_thread) { - ASSERT(signaling_thread); - talk_base::RefCountedObject* track = - new talk_base::RefCountedObject(label, - audio_device, - signaling_thread); - return track; -} - -talk_base::scoped_refptr AudioTrackProxy::CreateLocal( - LocalAudioTrackInterface* implementation, - talk_base::Thread* signaling_thread) { - ASSERT(signaling_thread); - talk_base::RefCountedObject* track = - new talk_base::RefCountedObject(implementation, - signaling_thread); - return track; -} - -AudioDeviceModule* AudioTrackProxy::GetAudioDevice() { - if (!signaling_thread_->IsCurrent()) { - AudioDeviceMessageData msg; - Send(MSG_GET_AUDIODEVICE, &msg); - return msg.audio_device_; - } - return audio_track_->GetAudioDevice(); -} - -void AudioTrackProxy::OnMessage(talk_base::Message* msg) { - if (!MediaStreamTrackProxy::HandleMessage(msg)) { - if (msg->message_id == MSG_GET_AUDIODEVICE) { - AudioDeviceMessageData* audio_device = - static_cast(msg->pdata); - audio_device->audio_device_ = audio_track_->GetAudioDevice(); - } else { - ASSERT(!"Not Implemented!"); - } - } -} - -VideoTrackProxy::VideoTrackProxy(const std::string& label, - talk_base::Thread* signaling_thread) - : MediaStreamTrackProxy(signaling_thread), - video_track_(VideoTrack::CreateRemote(label)) { - Init(video_track_); -} - -VideoTrackProxy::VideoTrackProxy(const std::string& label, - VideoCaptureModule* video_device, - talk_base::Thread* signaling_thread) - : MediaStreamTrackProxy(signaling_thread), - video_track_(VideoTrack::CreateLocal(label, video_device)) { - Init(video_track_); -} - -VideoTrackProxy::VideoTrackProxy(LocalVideoTrackInterface* implementation, - talk_base::Thread* signaling_thread) - : MediaStreamTrackProxy(signaling_thread), - video_track_(implementation) { - Init(video_track_); -} - -talk_base::scoped_refptr VideoTrackProxy::CreateRemote( - const std::string& label, - talk_base::Thread* signaling_thread) { - ASSERT(signaling_thread); - talk_base::RefCountedObject* track = - new talk_base::RefCountedObject(label, signaling_thread); - return track; -} - -talk_base::scoped_refptr VideoTrackProxy::CreateLocal( - const std::string& label, - VideoCaptureModule* video_device, - talk_base::Thread* signaling_thread) { - ASSERT(signaling_thread); - talk_base::RefCountedObject* track = - new talk_base::RefCountedObject(label, video_device, - signaling_thread); - return track; -} - -talk_base::scoped_refptr VideoTrackProxy::CreateLocal( - LocalVideoTrackInterface* implementation, - talk_base::Thread* signaling_thread) { - ASSERT(signaling_thread); - talk_base::RefCountedObject* track = - new talk_base::RefCountedObject(implementation, - signaling_thread); - return track; -} - -VideoCaptureModule* VideoTrackProxy::GetVideoCapture() { - if (!signaling_thread_->IsCurrent()) { - VideoDeviceMessageData msg; - Send(MSG_GET_VIDEODEVICE, &msg); - return msg.video_device_; - } - return video_track_->GetVideoCapture(); -} - -void VideoTrackProxy::SetRenderer(VideoRendererWrapperInterface* renderer) { - if (!signaling_thread_->IsCurrent()) { - VideoRendererMessageData msg; - msg.video_renderer_ = renderer; - Send(MSG_SET_VIDEORENDERER, &msg); - return; - } - return video_track_->SetRenderer(renderer); -} - -VideoRendererWrapperInterface* VideoTrackProxy::GetRenderer() { - if (!signaling_thread_->IsCurrent()) { - VideoRendererMessageData msg; - Send(MSG_GET_VIDEORENDERER, &msg); - return msg.video_renderer_; - } - return video_track_->GetRenderer(); -} - -void VideoTrackProxy::OnMessage(talk_base::Message* msg) { - if (!MediaStreamTrackProxy::HandleMessage(msg)) { - switch (msg->message_id) { - case MSG_GET_VIDEODEVICE: { - VideoDeviceMessageData* video_device = - static_cast(msg->pdata); - video_device->video_device_ = video_track_->GetVideoCapture(); - break; - } - case MSG_GET_VIDEORENDERER: { - VideoRendererMessageData* video_renderer = - static_cast(msg->pdata); - video_renderer->video_renderer_ = video_track_->GetRenderer(); - break; - } - case MSG_SET_VIDEORENDERER: { - VideoRendererMessageData* video_renderer = - static_cast(msg->pdata); - video_track_->SetRenderer(video_renderer->video_renderer_.get()); - break; - } - default: - ASSERT(!"Not Implemented!"); - break; - } - } -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamtrackproxy.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamtrackproxy.h deleted file mode 100644 index 8e00971b4..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastreamtrackproxy.h +++ /dev/null @@ -1,143 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -// This file includes proxy classes for tracks. The purpose is -// to make sure tracks are only accessed from the signaling thread. - -#ifndef TALK_APP_WEBRTC_MEDIASTREAMTRACKPROXY_H_ -#define TALK_APP_WEBRTC_MEDIASTREAMTRACKPROXY_H_ - -#include -#include - -#include "talk/app/webrtc_dev/audiotrackimpl.h" -#include "talk/app/webrtc_dev/mediastream.h" -#include "talk/app/webrtc_dev/videotrackimpl.h" -#include "talk/base/thread.h" - -namespace webrtc { - -template -class MediaStreamTrackProxy : public T, - talk_base::MessageHandler { - public: - void Init(MediaStreamTrackInterface* track); - // Implement MediaStreamTrack. - - virtual std::string kind() const; - virtual std::string label() const; - virtual bool enabled() const; - virtual MediaStreamTrackInterface::TrackState state() const; - virtual bool set_enabled(bool enable); - virtual bool set_state(MediaStreamTrackInterface::TrackState new_state); - - // Implement Notifier - virtual void RegisterObserver(ObserverInterface* observer); - virtual void UnregisterObserver(ObserverInterface* observer); - - protected: - explicit MediaStreamTrackProxy(talk_base::Thread* signaling_thread); - - void Send(uint32 id, talk_base::MessageData* data) const; - // Returns true if the message is handled. - bool HandleMessage(talk_base::Message* msg); - - mutable talk_base::Thread* signaling_thread_; - MediaStreamTrackInterface* track_; -}; - -// AudioTrackProxy is a proxy for the AudioTrackInterface. The purpose is -// to make sure AudioTrack is only accessed from the signaling thread. -// It can be used as a proxy for both local and remote audio tracks. -class AudioTrackProxy : public MediaStreamTrackProxy { - public: - static talk_base::scoped_refptr CreateRemote( - const std::string& label, - talk_base::Thread* signaling_thread); - static talk_base::scoped_refptr CreateLocal( - const std::string& label, - AudioDeviceModule* audio_device, - talk_base::Thread* signaling_thread); - static talk_base::scoped_refptr CreateLocal( - LocalAudioTrackInterface* implementation, - talk_base::Thread* signaling_thread); - - virtual AudioDeviceModule* GetAudioDevice(); - - protected: - AudioTrackProxy(const std::string& label, - talk_base::Thread* signaling_thread); - AudioTrackProxy(const std::string& label, - AudioDeviceModule* audio_device, - talk_base::Thread* signaling_thread); - AudioTrackProxy(LocalAudioTrackInterface* implementation, - talk_base::Thread* signaling_thread); - // Implement MessageHandler - virtual void OnMessage(talk_base::Message* msg); - - talk_base::scoped_refptr audio_track_; -}; - -// VideoTrackProxy is a proxy for the VideoTrackInterface and -// LocalVideoTrackInterface. The purpose is -// to make sure VideoTrack is only accessed from the signaling thread. -// It can be used as a proxy for both local and remote video tracks. -class VideoTrackProxy : public MediaStreamTrackProxy { - public: - static talk_base::scoped_refptr CreateRemote( - const std::string& label, - talk_base::Thread* signaling_thread); - static talk_base::scoped_refptr CreateLocal( - const std::string& label, - VideoCaptureModule* video_device, - talk_base::Thread* signaling_thread); - static talk_base::scoped_refptr CreateLocal( - LocalVideoTrackInterface* implementation, - talk_base::Thread* signaling_thread); - - virtual VideoCaptureModule* GetVideoCapture(); - virtual void SetRenderer(VideoRendererWrapperInterface* renderer); - VideoRendererWrapperInterface* GetRenderer(); - - protected: - VideoTrackProxy(const std::string& label, - talk_base::Thread* signaling_thread); - VideoTrackProxy(const std::string& label, - VideoCaptureModule* video_device, - talk_base::Thread* signaling_thread); - VideoTrackProxy(LocalVideoTrackInterface* implementation, - talk_base::Thread* signaling_thread); - - // Implement MessageHandler - virtual void OnMessage(talk_base::Message* msg); - - talk_base::scoped_refptr video_track_; -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_MEDIASTREAMTRACKPROXY_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediatrackimpl.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediatrackimpl.h deleted file mode 100644 index 189a2dfaa..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediatrackimpl.h +++ /dev/null @@ -1,78 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#ifndef TALK_APP_WEBRTC_MEDIATRACKIMPL_H_ -#define TALK_APP_WEBRTC_MEDIATRACKIMPL_H_ - -#include - -#include "talk/app/webrtc_dev/mediastream.h" -#include "talk/app/webrtc_dev/notifierimpl.h" - -namespace webrtc { - -// MediaTrack implements the interface common to AudioTrackInterface and -// VideoTrackInterface. -template -class MediaTrack : public Notifier { - public: - typedef typename T::TrackState TypedTrackState; - - virtual std::string label() const { return label_; } - virtual TypedTrackState state() const { return state_; } - virtual bool enabled() const { return enabled_; } - virtual bool set_enabled(bool enable) { - bool fire_on_change = (enable != enabled_); - enabled_ = enable; - if (fire_on_change) { - Notifier::FireOnChanged(); - } - } - virtual bool set_state(TypedTrackState new_state) { - bool fire_on_change = (state_ != new_state); - state_ = new_state; - if (fire_on_change) - Notifier::FireOnChanged(); - return true; - } - - protected: - explicit MediaTrack(const std::string& label) - : enabled_(true), - label_(label), - state_(T::kInitializing) { - } - - private: - bool enabled_; - std::string label_; - TypedTrackState state_; -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_MEDIATRACKIMPL_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/notifierimpl.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/notifierimpl.h deleted file mode 100644 index c8c9a549d..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/notifierimpl.h +++ /dev/null @@ -1,73 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#ifndef TALK_APP_WEBRTC_NOTIFIERIMPL_H_ -#define TALK_APP_WEBRTC_NOTIFIERIMPL_H_ - -#include - -#include "talk/base/common.h" -#include "talk/app/webrtc_dev/mediastream.h" - -namespace webrtc { - -// Implement a template version of a notifier. -template -class Notifier : public T { - public: - Notifier() { - } - - virtual void RegisterObserver(ObserverInterface* observer) { - ASSERT(observer != NULL); - observers_.push_back(observer); - } - - virtual void UnregisterObserver(ObserverInterface* observer) { - for (std::list::iterator it = observers_.begin(); - it != observers_.end(); it++) { - if (*it == observer) { - observers_.erase(it); - break; - } - } - } - - void FireOnChanged() { - for (std::list::iterator it = observers_.begin(); - it != observers_.end(); ++it) { - (*it)-> OnChanged(); - } - } - - protected: - std::list observers_; -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_NOTIFIERIMPL_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnection.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnection.h deleted file mode 100644 index 495b71474..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnection.h +++ /dev/null @@ -1,208 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -// This file contains the PeerConnection interface as defined in -// http://dev.w3.org/2011/webrtc/editor/webrtc.html#peer-to-peer-connections. -// Applications must use this interface to implement peerconnection. -// PeerConnectionFactory class provides factory methods to create -// peerconnection, mediastream and media tracks objects. -// -// The Following steps are needed to setup a typical call. -// 1. Create a PeerConnectionFactoryInterface. Check constructors for more -// information about input parameters. -// 2. Create a PeerConnection object. Provide a configuration string which -// points either to stun or turn server to generate ICE candidates and provide -// an object that implements the PeerConnectionObserver interface. -// Now PeerConnection will startcollecting ICE candidates. -// 3. Create local MediaStream and MediaTracks using the PeerConnectionFactory -// and add it to PeerConnection by calling AddStream. -// 4. Once all mediastreams are added to peerconnection, call -// CommitStreamChanges. Now PeerConnection starts generating an offer based on -// the local mediastreams. -// 5. When PeerConnection have generated the ICE candidates it will call the -// observer OnSignalingMessage callback with the initial offer. -// 6. When an Answer from peer received it must be supplied to the -// PeerConnection by calling ProcessSignalingMessage. -// At this point PeerConnection knows remote capabilities and ICE candidates. -// Media will start flowing to the remote peer. - -// The Receiver of a call can decide to accept or reject the call. -// This decision will be taken by the application not peerconnection. -// If application decides to accept the call -// 1. Create PeerConnectionFactoryInterface if it doesn't exist. -// 2. Create new PeerConnection -// 3. Provide the remote offer to the new PeerConnection object by calling -// ProcessSignalingMessage. -// 4. PeerConnection will call the observer function OnAddStream with remote -// MediaStream and tracks information. -// 5. PeerConnection will call the observer function OnSignalingMessage with -// local ICE candidates in a answer message. -// 6. Application can add it's own MediaStreams by calling AddStream. -// When all streams have been added the application must call -// CommitStreamChanges. Streams can be added at any time after the -// PeerConnection object have been created. - -#ifndef TALK_APP_WEBRTC_PEERCONNECTION_H_ -#define TALK_APP_WEBRTC_PEERCONNECTION_H_ - -#include - -#include "talk/app/webrtc_dev/mediastream.h" - -namespace talk_base { - class Thread; - class NetworkManager; - class PacketSocketFactory; -} - -namespace webrtc { -// MediaStream container interface. -class StreamCollectionInterface : public talk_base::RefCountInterface { - public: - virtual size_t count() = 0; - virtual MediaStreamInterface* at(size_t index) = 0; - virtual MediaStreamInterface* find(const std::string& label) = 0; - protected: - // Dtor protected as objects shouldn't be deleted via this interface. - ~StreamCollectionInterface() {} -}; - -// PeerConnection callback interface. Application should implement these -// methods. -class PeerConnectionObserver { - public: - enum Readiness { - kNegotiating, - kActive, - }; - - virtual void OnError() = 0; - - virtual void OnMessage(const std::string& msg) = 0; - - // Serialized signaling message - virtual void OnSignalingMessage(const std::string& msg) = 0; - - virtual void OnStateChange(Readiness state) = 0; - - // Triggered when media is received on a new stream from remote peer. - virtual void OnAddStream(MediaStreamInterface* stream) = 0; - - // Triggered when a remote peer close a stream. - virtual void OnRemoveStream(MediaStreamInterface* stream) = 0; - - protected: - // Dtor protected as objects shouldn't be deleted via this interface. - ~PeerConnectionObserver() {} -}; - - -class PeerConnectionInterface : public talk_base::RefCountInterface { - public: - // SignalingMessage in json format - virtual bool ProcessSignalingMessage(const std::string& msg) = 0; - - // Sends the msg over a data stream. - virtual bool Send(const std::string& msg) = 0; - - // Accessor methods to active local streams. - virtual talk_base::scoped_refptr - local_streams() = 0; - - // Accessor methods to remote streams. - virtual talk_base::scoped_refptr - remote_streams() = 0; - - // Add a new local stream. - // This function does not trigger any changes to the stream until - // CommitStreamChanges is called. - virtual void AddStream(LocalMediaStreamInterface* stream) = 0; - - // Remove a local stream and stop sending it. - // This function does not trigger any changes to the stream until - // CommitStreamChanges is called. - virtual void RemoveStream(LocalMediaStreamInterface* stream) = 0; - - // Commit Stream changes. This will start sending media on new streams - // and stop sending media on removed stream. - virtual void CommitStreamChanges() = 0; - - protected: - // Dtor protected as objects shouldn't be deleted via this interface. - ~PeerConnectionInterface() {} -}; - -// PeerConnectionFactoryInterface is the factory interface use for creating -// PeerConnection, MediaStream and media tracks. -// PeerConnectionFactoryInterface will create required libjingle threads, -// socket and network manager factory classes for networking. -// If application decides to provide its own implementation of these classes -// it should use alternate create method which accepts these parameters -// as input. - -class PeerConnectionFactoryInterface : public talk_base::RefCountInterface { - public: - virtual talk_base::scoped_refptr - CreatePeerConnection(const std::string& config, - PeerConnectionObserver* observer) = 0; - - virtual talk_base::scoped_refptr - CreateLocalMediaStream(const std::string& label) = 0; - - virtual talk_base::scoped_refptr - CreateLocalVideoTrack(const std::string& label, - VideoCaptureModule* video_device) = 0; - - virtual talk_base::scoped_refptr - CreateLocalAudioTrack(const std::string& label, - AudioDeviceModule* audio_device) = 0; - - protected: - // Dtor and ctor protected as objects shouldn't be created or deleted via - // this interface. - PeerConnectionFactoryInterface() {} - ~PeerConnectionFactoryInterface() {} // NOLINT -}; - -// Create a new instance of PeerConnectionFactoryInterface. -talk_base::scoped_refptr -CreatePeerConnectionFactory(); - -// Create a new instance of PeerConnectionFactoryInterface. -// Ownership of the arguments are not transfered to this object and must -// remain in scope for the lifetime of the PeerConnectionFactoryInterface. -talk_base::scoped_refptr -CreatePeerConnectionFactory( - talk_base::Thread* worker_thread, - talk_base::Thread* signaling_thread, - talk_base::NetworkManager* network_manager, - talk_base::PacketSocketFactory* packet_socket_factory, - AudioDeviceModule* default_adm); - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_PEERCONNECTION_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnection_unittest.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnection_unittest.cc deleted file mode 100644 index 356d3d8d7..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnection_unittest.cc +++ /dev/null @@ -1,348 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include - -#include - -#include "gtest/gtest.h" -#include "modules/video_capture/main/interface/video_capture_factory.h" -#include "talk/app/webrtc_dev/mediastream.h" -#include "talk/app/webrtc_dev/peerconnection.h" -#include "talk/app/webrtc_dev/test/filevideocapturemodule.h" -#include "talk/base/thread.h" -#include "talk/session/phone/videoframe.h" -#include "talk/session/phone/videorenderer.h" - -void GetAllVideoTracks(webrtc::MediaStreamInterface* media_stream, - std::list* video_tracks) { - webrtc::VideoTracks* track_list = media_stream->video_tracks(); - for (size_t i = 0; i < track_list->count(); ++i) { - webrtc::VideoTrackInterface* track = track_list->at(i); - video_tracks->push_back( - static_cast(track)); - } -} - -webrtc::VideoCaptureModule* OpenVideoCaptureDevice() { - webrtc::VideoCaptureModule::DeviceInfo* device_info( - webrtc::VideoCaptureFactory::CreateDeviceInfo(0)); - webrtc::VideoCaptureModule* video_device; - - const size_t kMaxDeviceNameLength = 128; - const size_t kMaxUniqueIdLength = 256; - uint8 device_name[kMaxDeviceNameLength]; - uint8 unique_id[kMaxUniqueIdLength]; - - const size_t device_count = device_info->NumberOfDevices(); - for (size_t i = 0; i < device_count; ++i) { - // Get the name of the video capture device. - device_info->GetDeviceName(i, device_name, kMaxDeviceNameLength, unique_id, - kMaxUniqueIdLength); - // Try to open this device. - video_device = - webrtc::VideoCaptureFactory::Create(0, unique_id); - if (video_device != NULL) - break; - } - delete device_info; - return video_device; -} - -class VideoRecorder : public cricket::VideoRenderer { - public: - static VideoRecorder* CreateVideoRecorder( - const char* file_name) { - VideoRecorder* renderer = new VideoRecorder(); - if (!renderer->Init(file_name)) { - delete renderer; - return NULL; - } - return renderer; - } - virtual ~VideoRecorder() { - if (output_file_ != NULL) { - fclose(output_file_); - } - } - - // Set up files so that recording can start immediately. - bool Init(const char* file_name) { - output_file_ = fopen(file_name, "wb"); - if (output_file_ == NULL) { - return false; - } - return true; - } - - virtual bool SetSize(int width, int height, int /*reserved*/) { - width_ = width; - height_ = height; - image_.reset(new uint8[buffersize()]); - } - - // |frame| is in I420 - virtual bool RenderFrame(const cricket::VideoFrame* frame) { - const int actual_size = frame->CopyToBuffer(image_.get(), - buffersize()); - if (actual_size > buffersize()) { - ASSERT(false); - // Skip frame. - return true; - } - // Write to file. - fwrite(image_.get(), sizeof(uint8), actual_size, output_file_); - return true; - } - - const uint8* image() const { - return image_.get(); - } - - int buffersize() const { - // I420 buffer size - return (width_ * height_ * 3) >> 1; - } - - int width() const { - return width_; - } - - int height() const { - return height_; - } - - protected: - VideoRecorder() - : width_(0), - height_(0), - output_file_(NULL) {} - - talk_base::scoped_array image_; - int width_; - int height_; - - // File to record to. - FILE* output_file_; -}; - -class SignalingMessageReceiver { - public: - virtual void ReceiveMessage(const std::string& msg) = 0; - - protected: - SignalingMessageReceiver() {} - virtual ~SignalingMessageReceiver() {} -}; - -class PeerConnectionP2PTestClient - : public webrtc::PeerConnectionObserver, - public SignalingMessageReceiver { - public: - static PeerConnectionP2PTestClient* CreateClient(int id) { - PeerConnectionP2PTestClient* client = new PeerConnectionP2PTestClient(id); - if (!client->Init()) { - delete client; - return NULL; - } - return client; - } - - ~PeerConnectionP2PTestClient() { - } - - void StartSession() { - // Audio track doesn't seem to be implemented yet. No need to pass a device - // to it. - talk_base::scoped_refptr audio_track( - peer_connection_factory_->CreateLocalAudioTrack("audio_track", NULL)); - - talk_base::scoped_refptr video_track( - peer_connection_factory_->CreateLocalVideoTrack( - "video_track", OpenVideoFileCaptureDevice())); - - talk_base::scoped_refptr stream = - peer_connection_factory_->CreateLocalMediaStream("stream_label"); - - stream->AddTrack(audio_track); - stream->AddTrack(video_track); - - peer_connection_->AddStream(stream); - peer_connection_->CommitStreamChanges(); - } - - void set_signaling_message_receiver( - SignalingMessageReceiver* signaling_message_receiver) { - signaling_message_receiver_ = signaling_message_receiver; - } - - // SignalingMessageReceiver callback. - virtual void ReceiveMessage(const std::string& msg) { - peer_connection_->ProcessSignalingMessage(msg); - } - - // PeerConnectionObserver callbacks. - virtual void OnError() {} - virtual void OnMessage(const std::string&) {} - virtual void OnSignalingMessage(const std::string& msg) { - if (signaling_message_receiver_ == NULL) { - // Remote party may be deleted. - return; - } - signaling_message_receiver_->ReceiveMessage(msg); - } - virtual void OnStateChange(Readiness) {} - virtual void OnAddStream(webrtc::MediaStreamInterface* media_stream) { - std::list video_tracks; - GetAllVideoTracks(media_stream, &video_tracks); - int track_id = 0; - for (std::list::iterator iter = - video_tracks.begin(); - iter != video_tracks.end(); - ++iter) { - char file_name[256]; - GenerateRecordingFileName(track_id, file_name); - talk_base::scoped_refptr - video_renderer = webrtc::CreateVideoRenderer( - VideoRecorder::CreateVideoRecorder(file_name)); - if (video_renderer == NULL) { - ADD_FAILURE(); - continue; - } - (*iter)->SetRenderer(video_renderer); - track_id++; - } - } - virtual void OnRemoveStream(webrtc::MediaStreamInterface*) { - } - - private: - static const int kFileNameSize = 256; - - explicit PeerConnectionP2PTestClient(int id) - : id_(id), - peer_connection_(), - peer_connection_factory_(), - signaling_message_receiver_(NULL) { - } - - bool Init() { - EXPECT_TRUE(peer_connection_.get() == NULL); - EXPECT_TRUE(peer_connection_factory_.get() == NULL); - peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(); - if (peer_connection_factory_.get() == NULL) { - ADD_FAILURE(); - return false; - } - - const char server_configuration[] = "STUN stun.l.google.com:19302"; - peer_connection_ = peer_connection_factory_->CreatePeerConnection( - server_configuration, this); - return peer_connection_.get() != NULL; - } - - void GenerateRecordingFileName(int track, char file_name[kFileNameSize]) { - if (file_name == NULL) { - return; - } - snprintf(file_name, kFileNameSize, - "p2p_test_client_%d_videotrack_%d.yuv", id_, track); - } - - webrtc::VideoCaptureModule* OpenVideoFileCaptureDevice() { - const char filename[] = "foreman_cif.yuv"; - webrtc::VideoCaptureModule* video_device = - FileVideoCaptureModule::CreateFileVideoCaptureModule(filename); - return video_device; - } - - int id_; - talk_base::scoped_refptr peer_connection_; - talk_base::scoped_refptr - peer_connection_factory_; - - // Remote peer communication. - SignalingMessageReceiver* signaling_message_receiver_; -}; - -class P2PTestConductor { - public: - static P2PTestConductor* CreateConductor() { - P2PTestConductor* conductor = new P2PTestConductor(); - if (!conductor->Init()) { - delete conductor; - return NULL; - } - return conductor; - } - ~P2PTestConductor() { - clients[0]->set_signaling_message_receiver(NULL); - clients[1]->set_signaling_message_receiver(NULL); - for (int i = 0; i < kClients; ++i) { - if (clients[i] != NULL) { - // TODO(hellner): currently deleting the clients will trigger an assert - // in cricket::BaseChannel::DisableMedia_w (not due to the unit test). - // Fix that problem and remove the below comment. - delete clients[i]; - } - } - } - - void StartSession() { - PeerConnectionP2PTestClient* initiating_client = clients[0]; - initiating_client->StartSession(); - } - - private: - static const int kClients = 2; - P2PTestConductor() { - clients[0] = NULL; - clients[1] = NULL; - } - - bool Init() { - for (int i = 0; i < kClients; ++i) { - clients[i] = PeerConnectionP2PTestClient::CreateClient(i); - if (clients[i] == NULL) { - return false; - } - } - clients[0]->set_signaling_message_receiver(clients[1]); - clients[1]->set_signaling_message_receiver(clients[0]); - return true; - } - - PeerConnectionP2PTestClient* clients[kClients]; -}; - -TEST(PeerConnection2, LocalP2PTest) { - P2PTestConductor* test = P2PTestConductor::CreateConductor(); - ASSERT_TRUE(test != NULL); - test->StartSession(); - talk_base::Thread::Current()->ProcessMessages(10000); - delete test; -} diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnection_unittests.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnection_unittests.cc deleted file mode 100644 index e9bc50a21..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnection_unittests.cc +++ /dev/null @@ -1,36 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ -#include "gtest/gtest.h" - -int main(int argc, char **argv) { - ::testing::InitGoogleTest(&argc, argv); - // Added return_value so that it's convenient to put a breakpoint before - // exiting please note that the return value from RUN_ALL_TESTS() must - // be returned by the main function. - const int return_value = RUN_ALL_TESTS(); - return return_value; -} diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionfactory_unittest.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionfactory_unittest.cc deleted file mode 100644 index 37d81bd6d..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionfactory_unittest.cc +++ /dev/null @@ -1,111 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include - -#include "gtest/gtest.h" -#include "talk/app/webrtc_dev/mediastreamimpl.h" -#include "talk/app/webrtc_dev/peerconnectionfactoryimpl.h" -#include "talk/base/basicpacketsocketfactory.h" -#include "talk/base/scoped_ptr.h" -#include "talk/base/thread.h" -#include "talk/session/phone/webrtccommon.h" -#include "talk/session/phone/webrtcvoe.h" - -#ifdef WEBRTC_RELATIVE_PATH -#include "modules/audio_device/main/source/audio_device_impl.h" -#else -#include "third_party/webrtc/files/include/audio_device_impl.h" -#endif - -static const char kAudioDeviceLabel[] = "dummy_audio_device"; -static const char kStunConfiguration[] = "STUN stun.l.google.com:19302"; - -namespace webrtc { - -class MockPeerConnectionObserver : public PeerConnectionObserver { - public: - virtual void OnError() {} - virtual void OnMessage(const std::string& msg) {} - virtual void OnSignalingMessage(const std::string& msg) {} - virtual void OnStateChange(Readiness state) {} - virtual void OnAddStream(MediaStreamInterface* stream) {} - virtual void OnRemoveStream(MediaStreamInterface* stream) {} -}; - -// TODO(mallinath) - Fix drash when components are created in factory. -TEST(PeerConnectionFactory, DISABLED_CreatePCUsingInternalModules) { - MockPeerConnectionObserver observer; - talk_base::scoped_refptr factory( - CreatePeerConnectionFactory()); - ASSERT_TRUE(factory.get() != NULL); - talk_base::scoped_refptr pc1( - factory->CreatePeerConnection("", &observer)); - EXPECT_TRUE(pc1.get() == NULL); - - talk_base::scoped_refptr pc2( - factory->CreatePeerConnection(kStunConfiguration, &observer)); - - EXPECT_TRUE(pc2.get() != NULL); -} - -TEST(PeerConnectionFactory, CreatePCUsingExternalModules) { - // Create an audio device. Use the default sound card. - talk_base::scoped_refptr audio_device( - AudioDeviceModuleImpl::Create(0)); - - // Creata a libjingle thread used as internal worker thread. - talk_base::scoped_ptr w_thread(new talk_base::Thread); - EXPECT_TRUE(w_thread->Start()); - - // Ownership of these pointers is handed over to the PeerConnectionFactory. - // TODO(henrike): add a check that ensures that the destructor is called for - // these classes. E.g. by writing a wrapper and set a flag in the wrappers - // destructor, or e.g. add a callback. - talk_base::NetworkManager* network_manager = - new talk_base::BasicNetworkManager(); - talk_base::PacketSocketFactory* socket_factory = - new talk_base::BasicPacketSocketFactory(); - - talk_base::scoped_refptr factory = - CreatePeerConnectionFactory(talk_base::Thread::Current(), - talk_base::Thread::Current(), - network_manager, - socket_factory, - audio_device); - ASSERT_TRUE(factory.get() != NULL); - - MockPeerConnectionObserver observer; - talk_base::scoped_refptr pc1( - factory->CreatePeerConnection("", &observer)); - - EXPECT_TRUE(pc1.get() == NULL); - - talk_base::scoped_refptr pc2( - factory->CreatePeerConnection(kStunConfiguration, &observer)); - EXPECT_TRUE(pc2.get() != NULL); -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionfactoryimpl.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionfactoryimpl.cc deleted file mode 100644 index 7812b831d..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionfactoryimpl.cc +++ /dev/null @@ -1,241 +0,0 @@ -/* - * libjingle - * Copyright 2004--2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include "talk/app/webrtc_dev/peerconnectionfactoryimpl.h" - -#include "talk/app/webrtc_dev/mediastreamproxy.h" -#include "talk/app/webrtc_dev/mediastreamtrackproxy.h" -#include "talk/app/webrtc_dev/peerconnectionimpl.h" -#include "talk/base/basicpacketsocketfactory.h" -#include "talk/session/phone/dummydevicemanager.h" -#include "talk/session/phone/webrtcmediaengine.h" - -#ifdef WEBRTC_RELATIVE_PATH -#include "modules/audio_device/main/interface/audio_device.h" -#else -#include "third_party/webrtc/files/include/audio_device.h" -#endif - -using talk_base::scoped_refptr; - -namespace { - -typedef talk_base::TypedMessageData InitMessageData; - -struct CreatePeerConnectionParams : public talk_base::MessageData { - CreatePeerConnectionParams(const std::string& configuration, - webrtc::PeerConnectionObserver* observer) - : configuration(configuration), observer(observer) { - } - scoped_refptr peerconnection; - const std::string& configuration; - webrtc::PeerConnectionObserver* observer; -}; - -enum { - MSG_INIT_FACTORY = 1, - MSG_CREATE_PEERCONNECTION = 2, -}; - -} // namespace anonymous - - -namespace webrtc { -scoped_refptr -CreatePeerConnectionFactory() { - talk_base::RefCountedObject* pc_factory = - new talk_base::RefCountedObject(); - - if (!pc_factory->Initialize()) { - delete pc_factory; - pc_factory = NULL; - } - return pc_factory; -} - -scoped_refptr -CreatePeerConnectionFactory( - talk_base::Thread* worker_thread, - talk_base::Thread* signaling_thread, - talk_base::NetworkManager* network_manager, - talk_base::PacketSocketFactory* socket_factory, - AudioDeviceModule* default_adm) { - talk_base::RefCountedObject* pc_factory = - new talk_base::RefCountedObject( - worker_thread, signaling_thread, network_manager, socket_factory, - default_adm); - if (!pc_factory->Initialize()) { - delete pc_factory; - pc_factory = NULL; - } - return pc_factory; -} - -PeerConnectionFactoryImpl::PeerConnectionFactoryImpl() - : worker_thread_(new talk_base::Thread), - signaling_thread_(new talk_base::Thread) { - worker_thread_ptr_ = worker_thread_.get(); - signaling_thread_ptr_ = signaling_thread_.get(); - bool result = worker_thread_->Start(); - ASSERT(result); - result = signaling_thread_->Start(); - ASSERT(result); -} - -PeerConnectionFactoryImpl::PeerConnectionFactoryImpl( - talk_base::Thread* worker_thread, - talk_base::Thread* signaling_thread, - talk_base::NetworkManager* network_manager, - talk_base::PacketSocketFactory* socket_factory, - AudioDeviceModule* default_adm) - : worker_thread_ptr_(worker_thread), - signaling_thread_ptr_(signaling_thread), - network_manager_(network_manager), - socket_factory_(socket_factory), - default_adm_(default_adm) { - ASSERT(worker_thread); - ASSERT(signaling_thread); - ASSERT(network_manager); - ASSERT(socket_factory); - ASSERT(default_adm); -} - -PeerConnectionFactoryImpl::~PeerConnectionFactoryImpl() { -} - -bool PeerConnectionFactoryImpl::Initialize() { - InitMessageData result(false); - signaling_thread_ptr_->Send(this, MSG_INIT_FACTORY, &result); - return result.data(); -} - -void PeerConnectionFactoryImpl::OnMessage(talk_base::Message* msg) { - switch (msg->message_id) { - case MSG_INIT_FACTORY: { - InitMessageData* pdata = static_cast (msg->pdata); - pdata->data() = Initialize_s(); - break; - } - case MSG_CREATE_PEERCONNECTION: { - CreatePeerConnectionParams* pdata = - static_cast (msg->pdata); - pdata->peerconnection = CreatePeerConnection_s(pdata->configuration, - pdata->observer); - break; - } - } -} - -bool PeerConnectionFactoryImpl::Initialize_s() { - if (!network_manager_.get()) - network_manager_.reset(new talk_base::BasicNetworkManager()); - if (!socket_factory_.get()) - socket_factory_.reset( - new talk_base::BasicPacketSocketFactory(worker_thread_ptr_)); - - cricket::DummyDeviceManager* device_manager( - new cricket::DummyDeviceManager()); - // TODO(perkj): Need to make sure only one VoE is created inside - // WebRtcMediaEngine. - cricket::WebRtcMediaEngine* webrtc_media_engine( - new cricket::WebRtcMediaEngine(default_adm_.get(), - NULL, // No secondary adm. - NULL)); // No vcm available. - - channel_manager_.reset(new cricket::ChannelManager( - webrtc_media_engine, device_manager, worker_thread_ptr_)); - if (!channel_manager_->Init()) { - return false; - } - return true; -} - -scoped_refptr -PeerConnectionFactoryImpl::CreatePeerConnection( - const std::string& configuration, - PeerConnectionObserver* observer) { - CreatePeerConnectionParams params(configuration, observer); - signaling_thread_ptr_->Send(this, MSG_CREATE_PEERCONNECTION, ¶ms); - return params.peerconnection; -} - -scoped_refptr -PeerConnectionFactoryImpl::CreatePeerConnection_s( - const std::string& configuration, - PeerConnectionObserver* observer) { - talk_base::RefCountedObject* pc( - new talk_base::RefCountedObject(this)); - if (!pc->Initialize(configuration, observer)) { - delete pc; - pc = NULL; - } - return pc; -} - -scoped_refptr -PeerConnectionFactoryImpl::CreateLocalMediaStream( - const std::string& label) { - return MediaStreamProxy::Create(label, signaling_thread_ptr_); -} - -scoped_refptr -PeerConnectionFactoryImpl::CreateLocalVideoTrack( - const std::string& label, - VideoCaptureModule* video_device) { - return VideoTrackProxy::CreateLocal(label, video_device, - signaling_thread_ptr_); -} - -scoped_refptr -PeerConnectionFactoryImpl::CreateLocalAudioTrack( - const std::string& label, - AudioDeviceModule* audio_device) { - return AudioTrackProxy::CreateLocal(label, audio_device, - signaling_thread_ptr_); -} - -cricket::ChannelManager* PeerConnectionFactoryImpl::channel_manager() { - return channel_manager_.get(); -} - -talk_base::Thread* PeerConnectionFactoryImpl::signaling_thread() { - return signaling_thread_ptr_; -} - -talk_base::Thread* PeerConnectionFactoryImpl::worker_thread() { - return worker_thread_ptr_; -} - -talk_base::NetworkManager* PeerConnectionFactoryImpl::network_manager() { - return network_manager_.get(); -} - -talk_base::PacketSocketFactory* PeerConnectionFactoryImpl::socket_factory() { - return socket_factory_.get(); -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionfactoryimpl.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionfactoryimpl.h deleted file mode 100644 index 71048c8e1..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionfactoryimpl.h +++ /dev/null @@ -1,96 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ -#ifndef TALK_APP_WEBRTC_PEERCONNECTIONFACTORYIMPL_H_ -#define TALK_APP_WEBRTC_PEERCONNECTIONFACTORYIMPL_H_ - -#include - -#include "talk/base/scoped_ptr.h" -#include "talk/app/webrtc_dev/peerconnection.h" -#include "talk/app/webrtc_dev/mediastream.h" -#include "talk/base/thread.h" -#include "talk/session/phone/channelmanager.h" - -namespace webrtc { - -class PeerConnectionFactoryImpl : public PeerConnectionFactoryInterface, - public talk_base::MessageHandler { - public: - talk_base::scoped_refptr CreatePeerConnection( - const std::string& config, - PeerConnectionObserver* observer); - bool Initialize(); - - virtual talk_base::scoped_refptr - CreateLocalMediaStream(const std::string& label); - - virtual talk_base::scoped_refptr - CreateLocalVideoTrack(const std::string& label, - VideoCaptureModule* video_device); - - virtual talk_base::scoped_refptr - CreateLocalAudioTrack(const std::string& label, - AudioDeviceModule* audio_device); - - virtual cricket::ChannelManager* channel_manager(); - virtual talk_base::Thread* signaling_thread(); - virtual talk_base::Thread* worker_thread(); - virtual talk_base::NetworkManager* network_manager(); - virtual talk_base::PacketSocketFactory* socket_factory(); - - protected: - PeerConnectionFactoryImpl(); - PeerConnectionFactoryImpl(talk_base::Thread* worker_thread, - talk_base::Thread* signaling_thread, - talk_base::NetworkManager* network_manager, - talk_base::PacketSocketFactory* socket_factory, - AudioDeviceModule* default_adm); - virtual ~PeerConnectionFactoryImpl(); - - - private: - bool Initialize_s(); - talk_base::scoped_refptr CreatePeerConnection_s( - const std::string& configuration, - PeerConnectionObserver* observer); - // Implements talk_base::MessageHandler. - void OnMessage(talk_base::Message* msg); - - talk_base::scoped_ptr signaling_thread_; - talk_base::Thread* signaling_thread_ptr_; - talk_base::scoped_ptr worker_thread_; - talk_base::Thread* worker_thread_ptr_; - talk_base::scoped_ptr network_manager_; - talk_base::scoped_ptr socket_factory_; - // External Audio device used for audio playback. - talk_base::scoped_refptr default_adm_; - talk_base::scoped_ptr channel_manager_; -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_PEERCONNECTIONFACTORYIMPL_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionimpl.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionimpl.cc deleted file mode 100644 index 18f32f19f..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionimpl.cc +++ /dev/null @@ -1,290 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include "talk/app/webrtc_dev/peerconnectionimpl.h" - -#include - -#include "talk/app/webrtc_dev/mediastreamhandler.h" -#include "talk/app/webrtc_dev/streamcollectionimpl.h" -#include "talk/base/logging.h" -#include "talk/session/phone/channelmanager.h" - -namespace { - -// The number of the tokens in the config string. -static const size_t kConfigTokens = 2; -static const int kServiceCount = 5; -// The default stun port. -static const int kDefaultPort = 3478; - -// NOTE: Must be in the same order as the ServiceType enum. -static const char* kValidServiceTypes[kServiceCount] = { - "STUN", "STUNS", "TURN", "TURNS", "INVALID" }; - -static const char kUserAgent[] = "PeerConnection User Agent"; - -enum ServiceType { - STUN, // Indicates a STUN server. - STUNS, // Indicates a STUN server used with a TLS session. - TURN, // Indicates a TURN server - TURNS, // Indicates a TURN server used with a TLS session. - INVALID, // Unknown. -}; - -enum { - MSG_COMMITSTREAMCHANGES = 1, - MSG_PROCESSSIGNALINGMESSAGE = 2, - MSG_RETURNREMOTEMEDIASTREAMS = 3, - MSG_TERMINATE = 4 -}; - -bool static ParseConfigString(const std::string& config, - talk_base::SocketAddress* addr, - ServiceType* service_type) { - std::vector tokens; - talk_base::tokenize(config, ' ', &tokens); - - if (tokens.size() != kConfigTokens) { - LOG(WARNING) << "Invalid config string"; - return false; - } - - *service_type = INVALID; - - const std::string& type = tokens[0]; - for (size_t i = 0; i < kServiceCount; ++i) { - if (type.compare(kValidServiceTypes[i]) == 0) { - *service_type = static_cast(i); - break; - } - } - - if (*service_type == INVALID) { - LOG(WARNING) << "Invalid service type: " << type; - return false; - } - std::string service_address = tokens[1]; - - int port; - tokens.clear(); - talk_base::tokenize(service_address, ':', &tokens); - if (tokens.size() != kConfigTokens) { - port = kDefaultPort; - } else { - port = atoi(tokens[1].c_str()); - if (port <= 0 || port > 0xffff) { - LOG(WARNING) << "Invalid port: " << tokens[1]; - return false; - } - } - addr->SetIP(service_address); - addr->SetPort(port); - return true; -} - -struct SignalingParams : public talk_base::MessageData { - SignalingParams(const std::string& msg, - webrtc::StreamCollectionInterface* local_streams) - : msg(msg), - local_streams(local_streams) {} - const std::string msg; - talk_base::scoped_refptr local_streams; -}; - -struct StreamCollectionParams : public talk_base::MessageData { - explicit StreamCollectionParams(webrtc::StreamCollectionInterface* streams) - : streams(streams) {} - talk_base::scoped_refptr streams; -}; - -} // namespace - -namespace webrtc { - -PeerConnectionImpl::PeerConnectionImpl( - PeerConnectionFactoryImpl* factory) - : factory_(factory), - observer_(NULL), - local_media_streams_(StreamCollectionImpl::Create()), - remote_media_streams_(StreamCollectionImpl::Create()), - port_allocator_(new cricket::HttpPortAllocator( - factory->network_manager(), - factory->socket_factory(), - std::string(kUserAgent))), - session_(new WebRtcSession(factory->channel_manager(), - factory->signaling_thread(), - factory->worker_thread(), - port_allocator_.get())), - signaling_(new PeerConnectionSignaling(factory->signaling_thread(), - session_.get())), - stream_handler_(new MediaStreamHandlers(session_.get())) { - signaling_->SignalNewPeerConnectionMessage.connect( - this, &PeerConnectionImpl::OnNewPeerConnectionMessage); - signaling_->SignalRemoteStreamAdded.connect( - this, &PeerConnectionImpl::OnRemoteStreamAdded); - signaling_->SignalRemoteStreamRemoved.connect( - this, &PeerConnectionImpl::OnRemoteStreamRemoved); - // Register with WebRtcSession - session_->RegisterObserver(signaling_.get()); -} - -PeerConnectionImpl::~PeerConnectionImpl() { - signaling_thread()->Clear(this); - signaling_thread()->Send(this, MSG_TERMINATE); -} - -// Clean up what needs to be cleaned up on the signaling thread. -void PeerConnectionImpl::Terminate_s() { - stream_handler_.reset(); - signaling_.reset(); - session_.reset(); - port_allocator_.reset(); -} - -bool PeerConnectionImpl::Initialize(const std::string& configuration, - PeerConnectionObserver* observer) { - ASSERT(observer); - if (!observer) - return false; - observer_ = observer; - talk_base::SocketAddress address; - ServiceType service; - if (!ParseConfigString(configuration, &address, &service)) - return false; - - switch (service) { - case STUN: { - std::vector address_vector; - address_vector.push_back(address); - port_allocator_->SetStunHosts(address_vector); - break; - } - case TURN: { - std::vector address_vector; - address_vector.push_back(address.ToString()); - port_allocator_->SetRelayHosts(address_vector); - break; - } - default: - ASSERT(!"NOT SUPPORTED"); - return false; - } - - // Initialize the WebRtcSession. It creates transport channels etc. - return session_->Initialize(); -} - -talk_base::scoped_refptr -PeerConnectionImpl::local_streams() { - return local_media_streams_; -} - -talk_base::scoped_refptr -PeerConnectionImpl::remote_streams() { - StreamCollectionParams msg(NULL); - signaling_thread()->Send(this, MSG_RETURNREMOTEMEDIASTREAMS, &msg); - return msg.streams; -} - -bool PeerConnectionImpl::ProcessSignalingMessage(const std::string& msg) { - SignalingParams* parameter(new SignalingParams( - msg, StreamCollectionImpl::Create(local_media_streams_))); - signaling_thread()->Post(this, MSG_PROCESSSIGNALINGMESSAGE, parameter); -} - -void PeerConnectionImpl::AddStream(LocalMediaStreamInterface* local_stream) { - local_media_streams_->AddStream(local_stream); -} - -void PeerConnectionImpl::RemoveStream( - LocalMediaStreamInterface* remove_stream) { - local_media_streams_->RemoveStream(remove_stream); -} - -void PeerConnectionImpl::CommitStreamChanges() { - StreamCollectionParams* msg(new StreamCollectionParams( - StreamCollectionImpl::Create(local_media_streams_))); - signaling_thread()->Post(this, MSG_COMMITSTREAMCHANGES, msg); -} - -void PeerConnectionImpl::OnMessage(talk_base::Message* msg) { - talk_base::MessageData* data = msg->pdata; - switch (msg->message_id) { - case MSG_COMMITSTREAMCHANGES: { - StreamCollectionParams* param( - static_cast (data)); - signaling_->CreateOffer(param->streams); - stream_handler_->CommitLocalStreams(param->streams); - delete data; // Because it is Posted. - break; - } - case MSG_PROCESSSIGNALINGMESSAGE: { - SignalingParams* params(static_cast (data)); - signaling_->ProcessSignalingMessage(params->msg, params->local_streams); - delete data; // Because it is Posted. - break; - } - case MSG_RETURNREMOTEMEDIASTREAMS: { - StreamCollectionParams* param( - static_cast (data)); - param->streams = StreamCollectionImpl::Create(remote_media_streams_); - break; - } - case MSG_TERMINATE: { - Terminate_s(); - break; - } - } -} - -void PeerConnectionImpl::OnNewPeerConnectionMessage( - const std::string& message) { - observer_->OnSignalingMessage(message); -} - -void PeerConnectionImpl::OnRemoteStreamAdded( - MediaStreamInterface* remote_stream) { - // TODO(perkj): add function in pc signaling to return a collection of - // remote streams. - // This way we can avoid keeping a separate list of remote_media_streams_. - remote_media_streams_->AddStream(remote_stream); - stream_handler_->AddRemoteStream(remote_stream); - observer_->OnAddStream(remote_stream); -} - -void PeerConnectionImpl::OnRemoteStreamRemoved( - MediaStreamInterface* remote_stream) { - // TODO(perkj): add function in pc signaling to return a collection of - // remote streams. - // This way we can avoid keeping a separate list of remote_media_streams_. - remote_media_streams_->RemoveStream(remote_stream); - stream_handler_->RemoveRemoteStream(remote_stream); - observer_->OnRemoveStream(remote_stream); -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionimpl.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionimpl.h deleted file mode 100644 index 48435edcd..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionimpl.h +++ /dev/null @@ -1,109 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#ifndef TALK_APP_WEBRTC_PEERCONNECTIONIMPL_H_ -#define TALK_APP_WEBRTC_PEERCONNECTIONIMPL_H_ - -#include -#include - -#include "talk/app/webrtc_dev/peerconnection.h" -#include "talk/app/webrtc_dev/peerconnectionfactoryimpl.h" -#include "talk/app/webrtc_dev/peerconnectionsignaling.h" -#include "talk/app/webrtc_dev/webrtcsession.h" -#include "talk/base/scoped_ptr.h" -#include "talk/p2p/client/httpportallocator.h" - -namespace cricket { -class ChannelManager; -} - -namespace webrtc { -class MediaStreamHandlers; -class StreamCollectionImpl; - - -// PeerConnectionImpl implements the PeerConnection interface. -// It uses PeerConnectionSignaling and WebRtcSession to implement -// the PeerConnection functionality. -class PeerConnectionImpl : public PeerConnectionInterface, - public talk_base::MessageHandler, - public sigslot::has_slots<> { - public: - explicit PeerConnectionImpl(PeerConnectionFactoryImpl* factory); - - bool Initialize(const std::string& configuration, - PeerConnectionObserver* observer); - - virtual ~PeerConnectionImpl(); - - virtual bool ProcessSignalingMessage(const std::string& msg); - virtual bool Send(const std::string& msg) { - // TODO(perkj): implement - ASSERT(false); - } - virtual talk_base::scoped_refptr local_streams(); - virtual talk_base::scoped_refptr remote_streams(); - virtual void AddStream(LocalMediaStreamInterface* stream); - virtual void RemoveStream(LocalMediaStreamInterface* stream); - virtual void CommitStreamChanges(); - - private: - // Implement talk_base::MessageHandler. - void OnMessage(talk_base::Message* msg); - - // Signals from PeerConnectionSignaling. - void OnNewPeerConnectionMessage(const std::string& message); - void OnRemoteStreamAdded(MediaStreamInterface* remote_stream); - void OnRemoteStreamRemoved(MediaStreamInterface* remote_stream); - - void Terminate_s(); - - talk_base::Thread* signaling_thread() { - return factory_->signaling_thread(); - } - - // Storing the factory as a scoped reference pointer ensures that the memory - // in the PeerConnectionFactoryImpl remains available as long as the - // PeerConnection is running. It is passed to PeerConnection as a raw pointer. - // However, since the reference counting is done in the - // PeerConnectionFactoryInteface all instances created using the raw pointer - // will refer to the same reference count. - talk_base::scoped_refptr factory_; - PeerConnectionObserver* observer_; - talk_base::scoped_refptr local_media_streams_; - talk_base::scoped_refptr remote_media_streams_; - - talk_base::scoped_ptr port_allocator_; - talk_base::scoped_ptr session_; - talk_base::scoped_ptr signaling_; - talk_base::scoped_ptr stream_handler_; -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_PEERCONNECTIONIMPL_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionimpl_unittest.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionimpl_unittest.cc deleted file mode 100644 index b5a57776b..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionimpl_unittest.cc +++ /dev/null @@ -1,81 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include - -#include "gtest/gtest.h" -#include "talk/app/webrtc_dev/mediastreamimpl.h" -#include "talk/app/webrtc_dev/peerconnection.h" -#include "talk/app/webrtc_dev/peerconnectionimpl.h" -#include "talk/base/scoped_ptr.h" -#include "talk/base/thread.h" - -static const char kStreamLabel1[] = "local_stream_1"; -static const char kStunConfiguration[] = "STUN stun.l.google.com:19302"; - -namespace webrtc { - -class MockPeerConnectionObserver : public PeerConnectionObserver { - public: - virtual void OnError() {} - virtual void OnMessage(const std::string& msg) {} - virtual void OnSignalingMessage(const std::string& msg) {} - virtual void OnStateChange(Readiness state) {} - virtual void OnAddStream(MediaStreamInterface* stream) {} - virtual void OnRemoveStream(MediaStreamInterface* stream) {} -}; - -class PeerConnectionImplTest : public testing::Test { - public: - protected: - virtual void SetUp() { - pc_factory_ = webrtc::CreatePeerConnectionFactory(); - ASSERT_TRUE(pc_factory_.get() != NULL); - pc_ = pc_factory_->CreatePeerConnection(kStunConfiguration, &observer_); - ASSERT_TRUE(pc_.get() != NULL); - } - - talk_base::scoped_refptr pc_factory_; - talk_base::scoped_refptr pc_; - MockPeerConnectionObserver observer_; -}; - -TEST_F(PeerConnectionImplTest, DISABLED_AddRemoveStream) { - // Create a local stream. - std::string label(kStreamLabel1); - talk_base::scoped_refptr stream( - pc_factory_->CreateLocalMediaStream(label)); - - pc_->AddStream(stream); - pc_->CommitStreamChanges(); - EXPECT_EQ(1l, pc_->local_streams()->count()); - if (pc_->local_streams()->count() <= 0) - return; - EXPECT_EQ(0, pc_->local_streams()->at(0)->label().compare(kStreamLabel1)); -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionmessage.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionmessage.cc deleted file mode 100644 index 29507d88f..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionmessage.cc +++ /dev/null @@ -1,95 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include "talk/app/webrtc_dev/peerconnectionmessage.h" - -#include -#include - -#include "talk/app/webrtc_dev/webrtcjson.h" - -namespace webrtc { - -PeerConnectionMessage* PeerConnectionMessage::Create( - PeerConnectionMessageType type, - const cricket::SessionDescription* desc, - const std::vector& candidates) { - return new PeerConnectionMessage(type, desc, candidates); -} - -PeerConnectionMessage* PeerConnectionMessage::Create( - const std::string& message) { - PeerConnectionMessage* pc_message(new PeerConnectionMessage()); - if (!pc_message->Deserialize(message)) - return NULL; - return pc_message; -} - -PeerConnectionMessage* PeerConnectionMessage::CreateErrorMessage( - ErrorCode error) { - return new PeerConnectionMessage(error); -} - -PeerConnectionMessage::PeerConnectionMessage( - PeerConnectionMessageType type, - const cricket::SessionDescription* desc, - const std::vector& candidates) - : type_(type), - error_code_(kNoError), - desc_(desc), - candidates_(candidates) { -} - -PeerConnectionMessage::PeerConnectionMessage() - : type_(kOffer), - error_code_(kNoError), - desc_(NULL) { -} - -PeerConnectionMessage::PeerConnectionMessage(ErrorCode error) - : type_(kError), - desc_(NULL), - error_code_(error) { -} - -std::string PeerConnectionMessage::Serialize() { - return JsonSerialize(type_, error_code_, desc_, candidates_); -} - -bool PeerConnectionMessage::Deserialize(std::string message) { - cricket::SessionDescription* desc(new cricket::SessionDescription()); - bool result = JsonDeserialize(&type_, &error_code_, desc, - &candidates_, message); - if (!result) { - delete desc; - desc = NULL; - } - desc_ = desc; - return result; -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionmessage.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionmessage.h deleted file mode 100644 index 1d29d8527..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionmessage.h +++ /dev/null @@ -1,102 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -// This file contains classes used for handling signaling between -// two PeerConnections. - -#ifndef TALK_APP_WEBRTC_DEV_PEERCONNECTIONMESSAGE_H_ -#define TALK_APP_WEBRTC_DEV_PEERCONNECTIONMESSAGE_H_ - -#include -#include - -#include "talk/base/refcount.h" -#include "talk/base/basictypes.h" -#include "talk/base/scoped_ptr.h" -#include "talk/base/scoped_refptr.h" -#include "talk/session/phone/mediasession.h" -#include "talk/p2p/base/sessiondescription.h" - -namespace webrtc { - -// PeerConnectionMessage represent an SDP offer or an answer. -// Instances of this class can be serialized / deserialized and are used for -// signaling between PeerConnection objects. -// Each instance has a type and a session description. -class PeerConnectionMessage { - public: - enum PeerConnectionMessageType { - kOffer, - kAnswer, - kError - }; - - enum ErrorCode { - kNoError = 0, - kWrongState = 10, // Offer received when Answer was expected. - kParseError = 20, // Can't parse / process offer. - kOfferNotAcceptable = 30, // The offer have been rejected. - kMessageNotDeliverable = 40 // The signaling channel is broken. - }; - - static PeerConnectionMessage* Create( - PeerConnectionMessageType type, - const cricket::SessionDescription* desc, - const std::vector& candidates); - - static PeerConnectionMessage* Create( - const std::string& message); - - static PeerConnectionMessage* CreateErrorMessage( - ErrorCode error); - - PeerConnectionMessageType type() {return type_;} - ErrorCode error() {return error_code_;} - const cricket::SessionDescription* desc() {return desc_;} - - std::string Serialize(); - std::vector& candidates() { return candidates_; } - - protected: - PeerConnectionMessage(PeerConnectionMessageType type, - const cricket::SessionDescription* desc, - const std::vector& candidates); - explicit PeerConnectionMessage(ErrorCode error); - PeerConnectionMessage(); - - bool Deserialize(std::string message); - - private: - PeerConnectionMessageType type_; - ErrorCode error_code_; - const cricket::SessionDescription* desc_; // Weak ref. - std::vector candidates_; -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_DEV_PEERCONNECTIONMESSAGE_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionmessage_unittest.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionmessage_unittest.cc deleted file mode 100644 index c793c595e..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionmessage_unittest.cc +++ /dev/null @@ -1,169 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include - -#include "gtest/gtest.h" -#include "talk/app/webrtc_dev/peerconnectionmessage.h" -#include "talk/base/logging.h" -#include "talk/base/scoped_ptr.h" -#include "talk/session/phone/channelmanager.h" - -using webrtc::PeerConnectionMessage; - -static const char kStreamLabel1[] = "local_stream_1"; -static const char kAudioTrackLabel1[] = "local_audio_1"; -static const char kVideoTrackLabel1[] = "local_video_1"; -static const char kVideoTrackLabel2[] = "local_video_2"; - -static const char kStreamLabel2[] = "local_stream_2"; -static const char kAudioTrackLabel2[] = "local_audio_2"; -static const char kVideoTrackLabel3[] = "local_video_3"; - -class PeerConnectionMessageTest: public testing::Test { - public: - PeerConnectionMessageTest() { - channel_manager_.reset(new cricket::ChannelManager( - talk_base::Thread::Current())); - EXPECT_TRUE(channel_manager_->Init()); - session_description_factory_.reset( - new cricket::MediaSessionDescriptionFactory(channel_manager_.get())); - options_.AddStream(cricket::MEDIA_TYPE_AUDIO, kAudioTrackLabel1, - kStreamLabel1); - options_.AddStream(cricket::MEDIA_TYPE_VIDEO, kVideoTrackLabel1, - kStreamLabel1); - options_.AddStream(cricket::MEDIA_TYPE_VIDEO, kVideoTrackLabel2, - kStreamLabel1); - - // kStreamLabel2 with 1 audio track and 1 video track - options_.AddStream(cricket::MEDIA_TYPE_AUDIO, kAudioTrackLabel2, - kStreamLabel2); - options_.AddStream(cricket::MEDIA_TYPE_VIDEO, kVideoTrackLabel3, - kStreamLabel2); - - int port = 1234; - talk_base::SocketAddress address("127.0.0.1", port++); - cricket::Candidate candidate1("video_rtcp", "udp", address, 1, - "user_video_rtcp", "password_video_rtcp", "local", "eth0", 0); - address.SetPort(port++); - cricket::Candidate candidate2("video_rtp", "udp", address, 1, - "user_video_rtp", "password_video_rtp", "local", "eth0", 0); - address.SetPort(port++); - cricket::Candidate candidate3("rtp", "udp", address, 1, - "user_rtp", "password_rtp", "local", "eth0", 0); - address.SetPort(port++); - cricket::Candidate candidate4("rtcp", "udp", address, 1, - "user_rtcp", "password_rtcp", "local", "eth0", 0); - - candidates_.push_back(candidate1); - candidates_.push_back(candidate2); - candidates_.push_back(candidate3); - candidates_.push_back(candidate4); - } - - protected: - talk_base::scoped_ptr channel_manager_; - talk_base::scoped_ptr - session_description_factory_; - cricket::MediaSessionOptions options_; - cricket::Candidates candidates_; -}; - -TEST_F(PeerConnectionMessageTest, Serialize) { - std::string message; - talk_base::scoped_ptr pc_message; - - // Offer - talk_base::scoped_ptr offer( - session_description_factory_->CreateOffer(options_)); - pc_message.reset(PeerConnectionMessage::Create(PeerConnectionMessage::kOffer, - offer.get(), candidates_)); - message = pc_message->Serialize(); - LOG(LS_INFO) << message; - - // Answer - talk_base::scoped_ptr answer( - session_description_factory_->CreateAnswer(offer.get(), options_)); - pc_message.reset(PeerConnectionMessage::Create(PeerConnectionMessage::kAnswer, - answer.get(), candidates_)); - message = pc_message->Serialize(); - LOG(LS_INFO) << message; - - // Error - pc_message.reset(PeerConnectionMessage::CreateErrorMessage( - PeerConnectionMessage::kParseError)); - message = pc_message->Serialize(); - LOG(LS_INFO) << message; - - // TODO(ronghuawu): Verify the serialized message. -} - -TEST_F(PeerConnectionMessageTest, Deserialize) { - std::string message_ref; - std::string message_result; - talk_base::scoped_ptr pc_message; - talk_base::scoped_ptr offer( - session_description_factory_->CreateOffer(options_)); - talk_base::scoped_ptr answer( - session_description_factory_->CreateAnswer(offer.get(), options_)); - - // Offer - pc_message.reset(PeerConnectionMessage::Create(PeerConnectionMessage::kOffer, - offer.get(), candidates_)); - message_ref = pc_message->Serialize(); - LOG(LS_INFO) << "The reference message: " << message_ref; - - // Deserialize Offer - pc_message.reset(PeerConnectionMessage::Create(message_ref)); - message_result = pc_message->Serialize(); - LOG(LS_INFO) << "The result message: " << message_result; - EXPECT_EQ(message_ref, message_result); - - // Answer - pc_message.reset(PeerConnectionMessage::Create(PeerConnectionMessage::kAnswer, - answer.get(), candidates_)); - message_ref = pc_message->Serialize(); - LOG(LS_INFO) << "The reference message: " << message_ref; - - // Deserialize Answer - pc_message.reset(PeerConnectionMessage::Create(message_ref)); - message_result = pc_message->Serialize(); - LOG(LS_INFO) << "The result message: " << message_result; - EXPECT_EQ(message_ref, message_result); - - // Error - pc_message.reset(PeerConnectionMessage::CreateErrorMessage( - PeerConnectionMessage::kParseError)); - message_ref = pc_message->Serialize(); - LOG(LS_INFO) << "The reference message: " << message_ref; - - // Deserialize Error - pc_message.reset(PeerConnectionMessage::Create(message_ref)); - message_result = pc_message->Serialize(); - LOG(LS_INFO) << "The result message: " << message_result; - EXPECT_EQ(message_ref, message_result); -} diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionsignaling.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionsignaling.cc deleted file mode 100644 index 9e785da3d..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionsignaling.cc +++ /dev/null @@ -1,535 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include "talk/app/webrtc_dev/peerconnectionsignaling.h" - -#include - -#include "talk/app/webrtc_dev/mediastreamtrackproxy.h" -#include "talk/app/webrtc_dev/sessiondescriptionprovider.h" -#include "talk/base/helpers.h" -#include "talk/base/messagequeue.h" -#include "talk/session/phone/channelmanager.h" - -using talk_base::scoped_refptr; - -namespace webrtc { - -enum { - MSG_SEND_QUEUED_OFFER = 1, - MSG_GENERATE_ANSWER = 2, -}; -static const int kGlareMinWaitTime = 2 * 1000; // 2 sec -static const int kGlareWaitIntervall = 1 * 1000; // 1 sec - -// Verifies that a SessionDescription contains as least one valid media content -// and a valid codec. -static bool VerifyAnswer(const cricket::SessionDescription* answer_desc) { - // We need to verify that at least one media content with - // a codec is available. - const cricket::ContentInfo* audio_content = - GetFirstAudioContent(answer_desc); - if (audio_content) { - const cricket::AudioContentDescription* audio_desc = - static_cast( - audio_content->description); - if (audio_desc->codecs().size() > 0) { - return true; - } - } - const cricket::ContentInfo* video_content = - GetFirstVideoContent(answer_desc); - if (video_content) { - const cricket::VideoContentDescription* video_desc = - static_cast( - video_content->description); - if (video_desc->codecs().size() > 0) { - return true; - } - } - return false; -} - -PeerConnectionSignaling::PeerConnectionSignaling( - talk_base::Thread* signaling_thread, - SessionDescriptionProvider* provider) - : signaling_thread_(signaling_thread), - provider_(provider), - state_(kInitializing) { -} - -PeerConnectionSignaling::~PeerConnectionSignaling() { - // Cleanup the queued_received_offer_ if this object is - // deleted before the offer can be processed. - // That means we have parsed an offer and created the remote - // session description but we have not transfered the ownership - // to the provider yet. - if (queued_received_offer_.first) { - const cricket::SessionDescription* remote_desc = - queued_received_offer_.first->desc(); - delete remote_desc; - delete queued_received_offer_.first; - } -} - -void PeerConnectionSignaling::OnCandidatesReady( - const cricket::Candidates& candidates) { - ASSERT(state_ == kInitializing); - if (state_ != kInitializing) - return; - // Store the candidates. - candidates_ = candidates; - // If we have a queued remote offer we need to handle this first. - if (queued_received_offer_.first != NULL) { - state_ = kIdle; - signaling_thread_->Post(this, MSG_GENERATE_ANSWER); - } else if (queued_offers_.size() >0) { - // Else if we have local queued offers. - state_ = PeerConnectionSignaling::kWaitingForAnswer; - signaling_thread_->Post(this, MSG_SEND_QUEUED_OFFER); - } else { - state_ = kIdle; - } -} - -void PeerConnectionSignaling::ProcessSignalingMessage( - const std::string& message, - StreamCollectionInterface* local_streams) { - ASSERT(talk_base::Thread::Current() == signaling_thread_); - - talk_base::scoped_ptr signaling_message( - PeerConnectionMessage::Create(message)); - if (!signaling_message.get()) { - signaling_message.reset(PeerConnectionMessage::CreateErrorMessage( - PeerConnectionMessage::kParseError)); - SignalNewPeerConnectionMessage(signaling_message->Serialize()); - } - - switch (signaling_message->type()) { - case PeerConnectionMessage::kOffer: { - queued_received_offer_ = - RemoteOfferPair(signaling_message.release(), local_streams); - // If we are still Initializing we need to wait before we can handle - // the offer. Queue it and handle it when the state change. - if (state_ == kInitializing) { - break; - } - // Don't handle offers when we are waiting for an answer. - if (state_ == kWaitingForAnswer) { - state_ = kGlare; - // Resends our last offer in 2 to 3s. - const int timeout = kGlareMinWaitTime + - talk_base::CreateRandomId() % kGlareWaitIntervall; - signaling_thread_->PostDelayed( - timeout, this, MSG_SEND_QUEUED_OFFER, NULL); - talk_base::scoped_ptr msg( - PeerConnectionMessage::CreateErrorMessage( - PeerConnectionMessage::kWrongState)); - SignalNewPeerConnectionMessage(msg->Serialize()); - break; - } - if (state_ == kGlare) { - state_ = kIdle; - } - // Clear the MSG_SEND_QUEUED_OFFER we posted delayed. - signaling_thread_->Clear(this, MSG_SEND_QUEUED_OFFER); - signaling_thread_->Post(this, MSG_GENERATE_ANSWER); - break; - } - case PeerConnectionMessage::kAnswer: { - ASSERT(state_ != PeerConnectionSignaling::kIdle); - if (state_ == PeerConnectionSignaling::kIdle) - return; - const cricket::SessionDescription* remote_desc = - provider_->SetRemoteSessionDescription( - signaling_message->desc(), - signaling_message->candidates()); - provider_->NegotiationDone(); - UpdateRemoteStreams(remote_desc); - scoped_refptr streams(queued_offers_.front()); - queued_offers_.pop_front(); - UpdateSendingLocalStreams(remote_desc, streams); - // Check if we have more offers waiting in the queue. - if (queued_offers_.size() > 0) { - // Send the next offer. - signaling_thread_->Post(this, MSG_SEND_QUEUED_OFFER); - } else { - state_ = PeerConnectionSignaling::kIdle; - } - break; - } - case PeerConnectionMessage::kError: { - if (signaling_message->error() != PeerConnectionMessage::kWrongState) { - SignalErrorMessageReceived(signaling_message->error()); - // An error have occurred that we can't do anything about. - // Reset the state and wait for user action. - queued_offers_.clear(); - state_ = kIdle; - } - break; - } - } -} - -void PeerConnectionSignaling::CreateOffer( - StreamCollectionInterface* local_streams) { - ASSERT(talk_base::Thread::Current() == signaling_thread_); - queued_offers_.push_back(local_streams); - if (state_ == kIdle) { - // Check if we can sent a new offer. - // Only one offer is allowed at the time. - state_ = PeerConnectionSignaling::kWaitingForAnswer; - signaling_thread_->Post(this, MSG_SEND_QUEUED_OFFER); - } -} - -// Implement talk_base::MessageHandler. -void PeerConnectionSignaling::OnMessage(talk_base::Message* msg) { - switch (msg->message_id) { - case MSG_SEND_QUEUED_OFFER: - CreateOffer_s(); - break; - case MSG_GENERATE_ANSWER: - CreateAnswer_s(); - break; - } -} - -void PeerConnectionSignaling::CreateOffer_s() { - ASSERT(queued_offers_.size() > 0); - scoped_refptr local_streams( - queued_offers_.front()); - cricket::MediaSessionOptions options; - InitMediaSessionOptions(&options, local_streams); - - const cricket::SessionDescription* local_desc = - provider_->ProvideOffer(options); - - talk_base::scoped_ptr offer_message( - PeerConnectionMessage::Create(PeerConnectionMessage::kOffer, - local_desc, - candidates_)); - SignalNewPeerConnectionMessage(offer_message->Serialize()); -} - -PeerConnectionSignaling::State PeerConnectionSignaling::GetState() { - return state_; -} - -void PeerConnectionSignaling::CreateAnswer_s() { - talk_base::scoped_ptr message( - queued_received_offer_.first); - queued_received_offer_.first = NULL; - scoped_refptr local_streams( - queued_received_offer_.second.release()); - - // Reset all pending offers. Instead, send the new streams in the answer. - signaling_thread_->Clear(this, MSG_SEND_QUEUED_OFFER, NULL); - queued_offers_.clear(); - - // Let the provider know about the remote offer. - // The provider takes ownership and return a pointer for us to use. - const cricket::SessionDescription* remote_desc = - provider_->SetRemoteSessionDescription(message->desc(), - message->candidates()); - - // Create a MediaSessionOptions object with the sources we want to send. - cricket::MediaSessionOptions options; - InitMediaSessionOptions(&options, local_streams); - // Create an local session description based on this. - const cricket::SessionDescription* local_desc = - provider_->ProvideAnswer(options); - - talk_base::scoped_ptr answer_message; - if (!VerifyAnswer(local_desc)) { - answer_message.reset(PeerConnectionMessage::CreateErrorMessage( - PeerConnectionMessage::kOfferNotAcceptable)); - - // Signal that the new answer is ready to be sent. - SignalNewPeerConnectionMessage(answer_message->Serialize()); - return; - } - - answer_message.reset(PeerConnectionMessage::Create( - PeerConnectionMessage::kAnswer, local_desc, candidates_)); - - // Let the provider know the negotiation is done. - provider_->NegotiationDone(); - - SignalNewPeerConnectionMessage(answer_message->Serialize()); - - UpdateRemoteStreams(message->desc()); - - // Update the state of the local streams. - UpdateSendingLocalStreams(answer_message->desc(), local_streams); -} - -// Fills a MediaSessionOptions struct with the MediaTracks we want to sent given -// the local MediaStreams. -// MediaSessionOptions contains the ssrc of the media track, the cname -// corresponding to the MediaStream and a label of the track. -void PeerConnectionSignaling::InitMediaSessionOptions( - cricket::MediaSessionOptions* options, - StreamCollectionInterface* local_streams) { - // In order to be able to receive video, - // the has_video should always be true even if there are not video tracks. - options->has_video = true; - for (size_t i = 0; i < local_streams->count(); ++i) { - MediaStreamInterface* stream = local_streams->at(i); - - scoped_refptr audio_tracks(stream->audio_tracks()); - // For each audio track in the stream, add it to the MediaSessionOptions. - for (size_t j = 0; j < audio_tracks->count(); ++j) { - scoped_refptr track(audio_tracks->at(j)); - options->AddStream(cricket::MEDIA_TYPE_AUDIO, track->label(), - stream->label()); - } - - scoped_refptr video_tracks(stream->video_tracks()); - // For each video track in the stream, add it to the MediaSessionOptions. - for (size_t j = 0; j < video_tracks->count(); ++j) { - scoped_refptr track(video_tracks->at(j)); - options->AddStream(cricket::MEDIA_TYPE_VIDEO, track->label(), - stream->label()); - } - } -} - -// Updates or Creates remote MediaStream objects given a -// remote SessionDesription. -// If the remote SessionDesription contain new remote MediaStreams -// SignalRemoteStreamAdded is triggered. If a remote MediaStream is missing from -// the remote SessionDescription SignalRemoteStreamRemoved is triggered. -void PeerConnectionSignaling::UpdateRemoteStreams( - const cricket::SessionDescription* remote_desc) { - RemoteStreamMap current_streams; - typedef std::pair > - MediaStreamPair; - - const cricket::ContentInfo* audio_content = GetFirstAudioContent(remote_desc); - if (audio_content) { - const cricket::AudioContentDescription* audio_desc = - static_cast( - audio_content->description); - - for (cricket::StreamParamsVec::const_iterator it = - audio_desc->streams().begin(); - it != audio_desc->streams().end(); ++it) { - RemoteStreamMap::iterator old_streams_it = - remote_streams_.find(it->sync_label); - RemoteStreamMap::iterator new_streams_it = - current_streams.find(it->sync_label); - - if (old_streams_it == remote_streams_.end()) { - if (new_streams_it == current_streams.end()) { - // New stream - scoped_refptr stream( - MediaStreamProxy::Create(it->sync_label, signaling_thread_)); - current_streams.insert(MediaStreamPair(stream->label(), stream)); - new_streams_it = current_streams.find(it->sync_label); - } - scoped_refptr track( - AudioTrackProxy::CreateRemote(it->name, signaling_thread_)); - track->set_state(MediaStreamTrackInterface::kLive); - new_streams_it->second->AddTrack(track); - } else { - scoped_refptr stream(old_streams_it->second); - current_streams.insert(MediaStreamPair(stream->label(), stream)); - } - } - } - - const cricket::ContentInfo* video_content = GetFirstVideoContent(remote_desc); - if (video_content) { - const cricket::VideoContentDescription* video_desc = - static_cast( - video_content->description); - - for (cricket::StreamParamsVec::const_iterator it = - video_desc->streams().begin(); - it != video_desc->streams().end(); ++it) { - RemoteStreamMap::iterator old_streams_it = - remote_streams_.find(it->cname); - RemoteStreamMap::iterator new_streams_it = - current_streams.find(it->cname); - - if (old_streams_it == remote_streams_.end()) { - if (new_streams_it == current_streams.end()) { - // New stream - scoped_refptr stream( - MediaStreamProxy::Create(it->cname, signaling_thread_)); - current_streams.insert(MediaStreamPair(stream->label(), stream)); - new_streams_it = current_streams.find(it->cname); - } - scoped_refptr track( - VideoTrackProxy::CreateRemote(it->name, signaling_thread_)); - new_streams_it->second->AddTrack(track); - track->set_state(MediaStreamTrackInterface::kLive); - - } else { - scoped_refptr stream(old_streams_it->second); - current_streams.insert(MediaStreamPair(stream->label(), stream)); - } - } - } - - // Iterate current_streams to find all new streams. - // Change the state of the new stream and SignalRemoteStreamAdded. - for (RemoteStreamMap::iterator it = current_streams.begin(); - it != current_streams.end(); - ++it) { - scoped_refptr new_stream(it->second); - RemoteStreamMap::iterator old_streams_it = - remote_streams_.find(new_stream->label()); - if (old_streams_it == remote_streams_.end()) { - new_stream->set_ready_state(MediaStreamInterface::kLive); - SignalRemoteStreamAdded(new_stream); - } - } - - // Iterate the old list of remote streams. - // If a stream is not found in the new list it have been removed. - // Change the state of the removed stream and SignalRemoteStreamRemoved. - for (RemoteStreamMap::iterator it = remote_streams_.begin(); - it != remote_streams_.end(); - ++it) { - scoped_refptr old_stream(it->second); - RemoteStreamMap::iterator new_streams_it = - current_streams.find(old_stream->label()); - if (new_streams_it == current_streams.end()) { - old_stream->set_ready_state(MediaStreamInterface::kEnded); - scoped_refptr audio_tracklist(old_stream->audio_tracks()); - for (size_t j = 0; j < audio_tracklist->count(); ++j) { - audio_tracklist->at(j)->set_state(MediaStreamTrackInterface::kEnded); - } - scoped_refptr video_tracklist(old_stream->video_tracks()); - for (size_t j = 0; j < video_tracklist->count(); ++j) { - video_tracklist->at(j)->set_state(MediaStreamTrackInterface::kEnded); - } - SignalRemoteStreamRemoved(old_stream); - } - } - // Set the remote_streams_ map to the map of MediaStreams we just created to - // be prepared for the next offer. - remote_streams_ = current_streams; -} - -// Update the state of all local streams we have just negotiated. If the -// negotiation succeeded the state is changed to kLive, if the negotiation -// failed the state is changed to kEnded. -void PeerConnectionSignaling::UpdateSendingLocalStreams( - const cricket::SessionDescription* answer_desc, - StreamCollectionInterface* negotiated_streams) { - typedef std::pair > - MediaStreamPair; - LocalStreamMap current_local_streams; - - for (size_t i = 0; i < negotiated_streams->count(); ++i) { - scoped_refptr stream(negotiated_streams->at(i)); - scoped_refptr audiotracklist(stream->audio_tracks()); - scoped_refptr videotracklist(stream->video_tracks()); - - bool stream_ok = false; // A stream is ok if at least one track succeed. - // Update tracks based on its type. - for (size_t j = 0; j < audiotracklist->count(); ++j) { - scoped_refptr track(audiotracklist->at(j)); - const cricket::ContentInfo* audio_content = - GetFirstAudioContent(answer_desc); - if (!audio_content) { // The remote does not accept audio. - track->set_state(MediaStreamTrackInterface::kFailed); - continue; - } - - const cricket::AudioContentDescription* audio_desc = - static_cast( - audio_content->description); - if (audio_desc->codecs().size() <= 0) { - // No common codec. - track->set_state(MediaStreamTrackInterface::kFailed); - } - track->set_state(MediaStreamTrackInterface::kLive); - stream_ok = true; - } - - for (size_t j = 0; j < videotracklist->count(); ++j) { - scoped_refptr track(videotracklist->at(j)); - const cricket::ContentInfo* video_content = - GetFirstVideoContent(answer_desc); - if (!video_content) { // The remote does not accept video. - track->set_state(MediaStreamTrackInterface::kFailed); - continue; - } - - const cricket::VideoContentDescription* video_desc = - static_cast( - video_content->description); - // TODO(perkj): Do we need to store the codec in the track? - if (video_desc->codecs().size() <= 0) { - // No common codec. - track->set_state(MediaStreamTrackInterface::kFailed); - } - track->set_state(MediaStreamTrackInterface::kLive); - stream_ok = true; - } - - if (stream_ok) { - // We have successfully negotiated to send this stream. - // Change the stream and store it as successfully negotiated. - stream->set_ready_state(MediaStreamInterface::kLive); - current_local_streams.insert(MediaStreamPair(stream->label(), stream)); - } else { - stream->set_ready_state(MediaStreamInterface::kEnded); - } - } - - // Iterate the old list of remote streams. - // If a stream is not found in the new list it have been removed. - // Change the state of the removed stream and all its tracks to kEnded. - for (LocalStreamMap::iterator it = local_streams_.begin(); - it != local_streams_.end(); - ++it) { - scoped_refptr old_stream(it->second); - MediaStreamInterface* new_streams = - negotiated_streams->find(old_stream->label()); - if (new_streams == NULL) { - old_stream->set_ready_state(MediaStreamInterface::kEnded); - scoped_refptr audio_tracklist(old_stream->audio_tracks()); - for (size_t j = 0; j < audio_tracklist->count(); ++j) { - audio_tracklist->at(j)->set_state(MediaStreamTrackInterface::kEnded); - } - scoped_refptr video_tracklist(old_stream->video_tracks()); - for (size_t j = 0; j < video_tracklist->count(); ++j) { - video_tracklist->at(j)->set_state(MediaStreamTrackInterface::kEnded); - } - } - } - - // Update the local_streams_ for next update. - local_streams_ = current_local_streams; -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionsignaling.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionsignaling.h deleted file mode 100644 index ed2eafe44..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionsignaling.h +++ /dev/null @@ -1,163 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -// This file contains classes used for handling signaling between -// two PeerConnections. - -#ifndef TALK_APP_WEBRTC_PEERCONNECTIONSIGNALING_H_ -#define TALK_APP_WEBRTC_PEERCONNECTIONSIGNALING_H_ - -#include -#include -#include -#include -#include - -#include "talk/app/webrtc_dev/mediastreamproxy.h" -#include "talk/app/webrtc_dev/peerconnection.h" -#include "talk/app/webrtc_dev/peerconnectionmessage.h" -#include "talk/app/webrtc_dev/sessiondescriptionprovider.h" -#include "talk/app/webrtc_dev/webrtcsessionobserver.h" -#include "talk/base/basictypes.h" -#include "talk/base/messagehandler.h" -#include "talk/base/refcount.h" -#include "talk/base/scoped_ptr.h" -#include "talk/base/scoped_refptr.h" -#include "talk/base/thread.h" -#include "talk/session/phone/mediasession.h" -#include "talk/p2p/base/sessiondescription.h" - -namespace cricket { -class ChannelManager; -class Candidate; -typedef std::vector Candidates; -} - -namespace webrtc { - -// PeerConnectionSignaling is a class responsible for handling signaling -// between PeerConnection objects. -// It creates remote MediaStream objects when the remote peer signals it wants -// to send a new MediaStream. -// It changes the state of local MediaStreams and tracks -// when a remote peer is ready to receive media. -// Call Initialize when local Candidates are ready. -// Call CreateOffer to negotiate new local streams to send. -// Call ProcessSignalingMessage when a new PeerConnectionMessage have been -// received from the remote peer. -// Before PeerConnectionSignaling can process an answer or create an offer, -// Initialize have to be called. The last request to create an offer or process -// an answer will be processed after Initialize have been called. -class PeerConnectionSignaling : public WebRtcSessionObserver, - public talk_base::MessageHandler { - public: - enum State { - // Awaiting the local candidates. - kInitializing, - // Ready to sent new offer or receive a new offer. - kIdle, - // We have sent an offer and expect an answer, or we want to update - // our own offer. - kWaitingForAnswer, - // While waiting for an answer to our offer we received an offer from - // the remote peer. - kGlare - }; - - PeerConnectionSignaling(talk_base::Thread* signaling_thread, - SessionDescriptionProvider* provider); - ~PeerConnectionSignaling(); - - // Process a received offer/answer from the remote peer. - void ProcessSignalingMessage(const std::string& message, - StreamCollectionInterface* local_streams); - - // Creates an offer containing all tracks in local_streams. - // When the offer is ready it is signaled by SignalNewPeerConnectionMessage. - // When the remote peer is ready to receive media on a stream , the state of - // the local stream will change to kAlive. - void CreateOffer(StreamCollectionInterface* local_streams); - - // Returns the current state. - State GetState(); - - // New PeerConnectionMessage with an SDP offer/answer is ready to be sent. - // The listener to this signal is expected to serialize and send the - // PeerConnectionMessage to the remote peer. - sigslot::signal1 SignalNewPeerConnectionMessage; - - // A new remote stream have been discovered. - sigslot::signal1 SignalRemoteStreamAdded; - - // Remote stream is no longer available. - sigslot::signal1 SignalRemoteStreamRemoved; - - // Remote PeerConnection sent an error message. - sigslot::signal1 SignalErrorMessageReceived; - - // Implements WebRtcSessionObserver - virtual void OnCandidatesReady(const cricket::Candidates& candidates); - // Implements talk_base::MessageHandler. - virtual void OnMessage(talk_base::Message* msg); - - private: - void CreateOffer_s(); - void CreateAnswer_s(); - - void InitMediaSessionOptions(cricket::MediaSessionOptions* options, - StreamCollectionInterface* local_streams); - - void UpdateRemoteStreams(const cricket::SessionDescription* remote_desc); - void UpdateSendingLocalStreams( - const cricket::SessionDescription* answer_desc, - StreamCollectionInterface* negotiated_streams); - - typedef std::list > - StreamCollectionList; - StreamCollectionList queued_offers_; - - typedef std::pair > - RemoteOfferPair; - RemoteOfferPair queued_received_offer_; - - talk_base::Thread* signaling_thread_; - SessionDescriptionProvider* provider_; - State state_; - - typedef std::map > - RemoteStreamMap; - RemoteStreamMap remote_streams_; - typedef std::map > - LocalStreamMap; - LocalStreamMap local_streams_; - cricket::Candidates candidates_; -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_PEERCONNECTIONSIGNALING_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionsignaling_unittest.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionsignaling_unittest.cc deleted file mode 100644 index 65111b0f7..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnectionsignaling_unittest.cc +++ /dev/null @@ -1,455 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include -#include -#include - -#include "gtest/gtest.h" -#include "talk/app/webrtc_dev/mediastreamimpl.h" -#include "talk/app/webrtc_dev/videotrackimpl.h" -#include "talk/app/webrtc_dev/audiotrackimpl.h" -#include "talk/app/webrtc_dev/peerconnectionsignaling.h" -#include "talk/app/webrtc_dev/streamcollectionimpl.h" -#include "talk/base/scoped_ptr.h" -#include "talk/base/thread.h" -#include "talk/session/phone/channelmanager.h" - -static const char kStreamLabel1[] = "local_stream_1"; -static const char kAudioTrackLabel1[] = "local_audio_1"; -static const char kVideoTrackLabel1[] = "local_video_1"; -static const int kWaitTime = 5000; - -namespace webrtc { - -typedef std::map > - MediaStreamMap; -typedef std::pair > - RemotePair; - -class MockMediaTrackObserver : public webrtc::ObserverInterface { - public: - explicit MockMediaTrackObserver(MediaStreamTrackInterface* track) - : track_(track) { - track_state = track->state(); - track->RegisterObserver(this); - } - - virtual void OnChanged() { - track_state = track_->state(); - } - - webrtc::MediaStreamTrackInterface::TrackState track_state; - private: - talk_base::scoped_refptr track_; -}; - -class MockMediaStreamObserver : public webrtc::ObserverInterface { - public: - explicit MockMediaStreamObserver(MediaStreamInterface* stream) - : stream_(stream) { - ready_state = stream->ready_state(); - stream_->RegisterObserver(this); - } - - virtual void OnChanged() { - ready_state = stream_->ready_state(); - } - - webrtc::MediaStreamInterface::ReadyState ready_state; - private: - talk_base::scoped_refptr stream_; -}; - -class MockSignalingObserver : public sigslot::has_slots<> { - public: - MockSignalingObserver() - : remote_peer_(NULL) { - } - - // New remote stream have been discovered. - virtual void OnRemoteStreamAdded(MediaStreamInterface* remote_stream) { - EXPECT_EQ(MediaStreamInterface::kLive, remote_stream->ready_state()); - remote_media_streams_.insert(RemotePair(remote_stream->label(), - remote_stream)); - } - - // Remote stream is no longer available. - virtual void OnRemoteStreamRemoved(MediaStreamInterface* remote_stream) { - EXPECT_NE(remote_media_streams_.find(remote_stream->label()), - remote_media_streams_.end()); - remote_media_streams_.erase(remote_stream->label()); - } - - // New answer ready to be sent. - void OnSignalingMessage(const std::string& smessage) { - if (remote_peer_) { - remote_peer_->ProcessSignalingMessage(smessage, remote_local_collection_); - // Process posted messages to allow the remote peer to process - // the message. - talk_base::Thread::Current()->ProcessMessages(1); - } - talk_base::scoped_ptr message( - PeerConnectionMessage::Create(smessage)); - if (message.get() != NULL && - message->type() != PeerConnectionMessage::kError) { - last_message = smessage; - } - } - - // Tell this object to answer the remote_peer. - // remote_local_collection is the local collection the remote peer want to - // send in an answer. - void AnswerPeer(PeerConnectionSignaling* remote_peer, - StreamCollectionImpl* remote_local_collection) { - remote_peer_ = remote_peer; - remote_local_collection_ = remote_local_collection; - } - - void CancelAnswerPeer() { - remote_peer_ = NULL; - remote_local_collection_.release(); - } - - MediaStreamInterface* RemoteStream(const std::string& label) { - MediaStreamMap::iterator it = remote_media_streams_.find(label); - if (it != remote_media_streams_.end()) - return it->second; - return NULL; - } - - virtual ~MockSignalingObserver() {} - - std::string last_message; - - private: - MediaStreamMap remote_media_streams_; - talk_base::scoped_refptr remote_local_collection_; - PeerConnectionSignaling* remote_peer_; -}; - -class MockSessionDescriptionProvider : public SessionDescriptionProvider { - public: - explicit MockSessionDescriptionProvider( - cricket::ChannelManager* channel_manager) - : update_session_description_counter_(0), - session_description_factory_( - new cricket::MediaSessionDescriptionFactory(channel_manager)) { - } - virtual const cricket::SessionDescription* ProvideOffer( - const cricket::MediaSessionOptions& options) { - offer_.reset(session_description_factory_->CreateOffer(options)); - return offer_.get(); - } - - // Transfer ownership of remote_offer. - virtual const cricket::SessionDescription* SetRemoteSessionDescription( - const cricket::SessionDescription* remote_offer, - const cricket::Candidates& remote_candidates) { - remote_desc_.reset(remote_offer); - return remote_desc_.get(); - } - - virtual const cricket::SessionDescription* ProvideAnswer( - const cricket::MediaSessionOptions& options) { - answer_.reset(session_description_factory_->CreateAnswer(remote_desc_.get(), - options)); - return answer_.get(); - } - - virtual void NegotiationDone() { - ++update_session_description_counter_; - } - - int update_session_description_counter_; - - protected: - talk_base::scoped_ptr - session_description_factory_; - talk_base::scoped_ptr offer_; - talk_base::scoped_ptr answer_; - talk_base::scoped_ptr remote_desc_; -}; - -class PeerConnectionSignalingTest: public testing::Test { - protected: - virtual void SetUp() { - channel_manager_.reset(new cricket::ChannelManager( - talk_base::Thread::Current())); - EXPECT_TRUE(channel_manager_->Init()); - provider1_.reset(new MockSessionDescriptionProvider( - channel_manager_.get())); - provider2_.reset(new MockSessionDescriptionProvider( - channel_manager_.get())); - - signaling1_.reset(new PeerConnectionSignaling( - talk_base::Thread::Current(), provider1_.get())); - observer1_.reset(new MockSignalingObserver()); - signaling1_->SignalNewPeerConnectionMessage.connect( - observer1_.get(), &MockSignalingObserver::OnSignalingMessage); - signaling1_->SignalRemoteStreamAdded.connect( - observer1_.get(), &MockSignalingObserver::OnRemoteStreamAdded); - signaling1_->SignalRemoteStreamRemoved.connect( - observer1_.get(), &MockSignalingObserver::OnRemoteStreamRemoved); - - signaling2_.reset(new PeerConnectionSignaling( - talk_base::Thread::Current(), provider2_.get())); - observer2_.reset(new MockSignalingObserver()); - signaling2_->SignalNewPeerConnectionMessage.connect( - observer2_.get(), &MockSignalingObserver::OnSignalingMessage); - signaling2_->SignalRemoteStreamAdded.connect( - observer2_.get(), &MockSignalingObserver::OnRemoteStreamAdded); - signaling2_->SignalRemoteStreamRemoved.connect( - observer2_.get(), &MockSignalingObserver::OnRemoteStreamRemoved); - } - - cricket::Candidates candidates_; - talk_base::scoped_ptr observer1_; - talk_base::scoped_ptr observer2_; - talk_base::scoped_ptr provider1_; - talk_base::scoped_ptr provider2_; - talk_base::scoped_ptr signaling1_; - talk_base::scoped_ptr signaling2_; - talk_base::scoped_ptr channel_manager_; -}; - -TEST_F(PeerConnectionSignalingTest, SimpleOneWayCall) { - // Create a local stream. - std::string label(kStreamLabel1); - talk_base::scoped_refptr stream( - MediaStream::Create(label)); - MockMediaStreamObserver stream_observer1(stream); - - // Add a local audio track. - talk_base::scoped_refptr - audio_track(AudioTrack::CreateLocal(kAudioTrackLabel1, NULL)); - stream->AddTrack(audio_track); - MockMediaTrackObserver track_observer1(audio_track); - - // Peer 1 create an offer with only one audio track. - talk_base::scoped_refptr local_collection1( - StreamCollectionImpl::Create()); - local_collection1->AddStream(stream); - // Verify that the local stream is now initializing. - EXPECT_EQ(MediaStreamInterface::kInitializing, stream_observer1.ready_state); - // Verify that the audio track is now initializing. - EXPECT_EQ(MediaStreamTrackInterface::kInitializing, - track_observer1.track_state); - - // Peer 2 only receive. Create an empty collection - talk_base::scoped_refptr local_collection2( - StreamCollectionImpl::Create()); - - // Connect all messages sent from Peer1 to be received on Peer2 - observer1_->AnswerPeer(signaling2_.get(), local_collection2); - // Connect all messages sent from Peer2 to be received on Peer1 - observer2_->AnswerPeer(signaling1_.get(), local_collection1); - - // Peer 1 generates the offer. It is not sent since there is no - // local candidates ready. - signaling1_->CreateOffer(local_collection1); - - // Process posted messages. - talk_base::Thread::Current()->ProcessMessages(1); - EXPECT_EQ(PeerConnectionSignaling::kInitializing, signaling1_->GetState()); - - // Initialize signaling1_ by providing the candidates. - signaling1_->OnCandidatesReady(candidates_); - EXPECT_EQ(PeerConnectionSignaling::kWaitingForAnswer, - signaling1_->GetState()); - // Process posted messages to allow signaling_1 to send the offer. - talk_base::Thread::Current()->ProcessMessages(1); - - // Verify that signaling_2 is still not initialized. - // Even though it have received an offer. - EXPECT_EQ(PeerConnectionSignaling::kInitializing, signaling2_->GetState()); - - // Provide the candidates to signaling_2 and let it process the offer. - signaling2_->OnCandidatesReady(candidates_); - talk_base::Thread::Current()->ProcessMessages(1); - - // Verify that the offer/answer have been exchanged and the state is good. - EXPECT_EQ(PeerConnectionSignaling::kIdle, signaling1_->GetState()); - EXPECT_EQ(PeerConnectionSignaling::kIdle, signaling2_->GetState()); - - // Verify that the local stream is now sending. - EXPECT_EQ(MediaStreamInterface::kLive, stream_observer1.ready_state); - // Verify that the local audio track is now sending. - EXPECT_EQ(MediaStreamTrackInterface::kLive, track_observer1.track_state); - - // Verify that PeerConnection2 is aware of the sending stream. - EXPECT_TRUE(observer2_->RemoteStream(label) != NULL); - - // Verify that both peers have updated the session descriptions. - EXPECT_EQ(1u, provider1_->update_session_description_counter_); - EXPECT_EQ(1u, provider2_->update_session_description_counter_); -} - -TEST_F(PeerConnectionSignalingTest, Glare) { - // Initialize signaling1_ and signaling_2 by providing the candidates. - signaling1_->OnCandidatesReady(candidates_); - signaling2_->OnCandidatesReady(candidates_); - // Create a local stream. - std::string label(kStreamLabel1); - talk_base::scoped_refptr stream( - MediaStream::Create(label)); - - // Add a local audio track. - talk_base::scoped_refptr - audio_track(AudioTrack::CreateLocal(kAudioTrackLabel1, NULL)); - stream->AddTrack(audio_track); - - // Peer 1 create an offer with only one audio track. - talk_base::scoped_refptr local_collection1( - StreamCollectionImpl::Create()); - local_collection1->AddStream(stream); - signaling1_->CreateOffer(local_collection1); - EXPECT_EQ(PeerConnectionSignaling::kWaitingForAnswer, - signaling1_->GetState()); - // Process posted messages. - talk_base::Thread::Current()->ProcessMessages(1); - - // Peer 2 only receive. Create an empty collection. - talk_base::scoped_refptr local_collection2( - StreamCollectionImpl::Create()); - // Peer 2 create an empty offer. - signaling2_->CreateOffer(local_collection2); - - // Process posted messages. - talk_base::Thread::Current()->ProcessMessages(1); - - // Peer 2 sends the offer to Peer1 and Peer1 sends its offer to Peer2. - ASSERT_TRUE(!observer1_->last_message.empty()); - ASSERT_TRUE(!observer2_->last_message.empty()); - signaling2_->ProcessSignalingMessage(observer1_->last_message, - local_collection2); - - signaling1_->ProcessSignalingMessage(observer2_->last_message, - local_collection1); - - EXPECT_EQ(PeerConnectionSignaling::kGlare, signaling1_->GetState()); - EXPECT_EQ(PeerConnectionSignaling::kGlare, signaling2_->GetState()); - - // Make sure all messages are send between - // the two PeerConnectionSignaling objects. - observer1_->AnswerPeer(signaling2_.get(), local_collection2); - observer2_->AnswerPeer(signaling1_.get(), local_collection1); - - // Process all delayed posted messages. - talk_base::Thread::Current()->ProcessMessages(kWaitTime); - - EXPECT_EQ(PeerConnectionSignaling::kIdle, signaling1_->GetState()); - EXPECT_EQ(PeerConnectionSignaling::kIdle, signaling2_->GetState()); - - // Verify that PeerConnection2 is aware of the sending stream. - EXPECT_TRUE(observer2_->RemoteStream(label) != NULL); - - // Verify that both peers have updated the session descriptions. - EXPECT_EQ(1u, provider1_->update_session_description_counter_); - EXPECT_EQ(1u, provider2_->update_session_description_counter_); -} - -TEST_F(PeerConnectionSignalingTest, AddRemoveStream) { - // Initialize signaling1_ and signaling_2 by providing the candidates. - signaling1_->OnCandidatesReady(candidates_); - signaling2_->OnCandidatesReady(candidates_); - // Create a local stream. - std::string label(kStreamLabel1); - talk_base::scoped_refptr stream( - MediaStream::Create(label)); - MockMediaStreamObserver stream_observer1(stream); - - // Add a local audio track. - talk_base::scoped_refptr - audio_track(AudioTrack::CreateLocal(kAudioTrackLabel1, NULL)); - stream->AddTrack(audio_track); - MockMediaTrackObserver track_observer1(audio_track); - audio_track->RegisterObserver(&track_observer1); - - // Add a local video track. - talk_base::scoped_refptr - video_track(VideoTrack::CreateLocal(kVideoTrackLabel1, NULL)); - stream->AddTrack(video_track); - - // Peer 1 create an empty collection - talk_base::scoped_refptr local_collection1( - StreamCollectionImpl::Create()); - - // Peer 2 create an empty collection - talk_base::scoped_refptr local_collection2( - StreamCollectionImpl::Create()); - - // Connect all messages sent from Peer1 to be received on Peer2 - observer1_->AnswerPeer(signaling2_.get(), local_collection2); - // Connect all messages sent from Peer2 to be received on Peer1 - observer2_->AnswerPeer(signaling1_.get(), local_collection1); - - // Peer 1 creates an empty offer and send it to Peer2. - signaling1_->CreateOffer(local_collection1); - // Process posted messages. - talk_base::Thread::Current()->ProcessMessages(1); - - // Verify that both peers have updated the session descriptions. - EXPECT_EQ(1u, provider1_->update_session_description_counter_); - EXPECT_EQ(1u, provider2_->update_session_description_counter_); - - // Peer2 add a stream. - local_collection2->AddStream(stream); - - signaling2_->CreateOffer(local_collection2); - talk_base::Thread::Current()->ProcessMessages(1); - - // Verify that the PeerConnection 2 local stream is now sending. - EXPECT_EQ(MediaStreamInterface::kLive, stream_observer1.ready_state); - EXPECT_EQ(MediaStreamTrackInterface::kLive, track_observer1.track_state); - - // Verify that PeerConnection1 is aware of the sending stream. - EXPECT_TRUE(observer1_->RemoteStream(label) != NULL); - - // Verify that both peers have updated the session descriptions. - EXPECT_EQ(2u, provider1_->update_session_description_counter_); - EXPECT_EQ(2u, provider2_->update_session_description_counter_); - - // Remove the stream - local_collection2->RemoveStream(stream); - - signaling2_->CreateOffer(local_collection2); - talk_base::Thread::Current()->ProcessMessages(1); - - // Verify that PeerConnection1 is not aware of the sending stream. - EXPECT_TRUE(observer1_->RemoteStream(label) == NULL); - - // Verify that the PeerConnection 2 local stream is now ended. - EXPECT_EQ(MediaStreamInterface::kEnded, stream_observer1.ready_state); - EXPECT_EQ(MediaStreamTrackInterface::kEnded, track_observer1.track_state); - - // Verify that both peers have updated the session descriptions. - EXPECT_EQ(3u, provider1_->update_session_description_counter_); - EXPECT_EQ(3u, provider2_->update_session_description_counter_); -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/sessiondescriptionprovider.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/sessiondescriptionprovider.h deleted file mode 100644 index f0a764542..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/sessiondescriptionprovider.h +++ /dev/null @@ -1,58 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#ifndef TALK_APP_WEBRTC_SESSIONDESCRIPTIONPROVIDER_H_ -#define TALK_APP_WEBRTC_SESSIONDESCRIPTIONPROVIDER_H_ - -#include "talk/session/phone/mediasession.h" -#include "talk/p2p/base/candidate.h" -#include "talk/p2p/base/sessiondescription.h" - -namespace webrtc { - -class SessionDescriptionProvider { - public: - virtual const cricket::SessionDescription* ProvideOffer( - const cricket::MediaSessionOptions& options) = 0; - - // Transfer ownership of remote_offer. - virtual const cricket::SessionDescription* SetRemoteSessionDescription( - const cricket::SessionDescription* remote_offer, - const std::vector& remote_candidates) = 0; - - virtual const cricket::SessionDescription* ProvideAnswer( - const cricket::MediaSessionOptions& options) = 0; - - virtual void NegotiationDone() = 0; - - protected: - virtual ~SessionDescriptionProvider() {} -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_SESSIONDESCRIPTIONPROVIDER_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/streamcollectionimpl.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/streamcollectionimpl.h deleted file mode 100644 index 4ab9012e6..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/streamcollectionimpl.h +++ /dev/null @@ -1,103 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#ifndef TALK_APP_WEBRTC_STREAMCOLLECTIONIMPL_H_ -#define TALK_APP_WEBRTC_STREAMCOLLECTIONIMPL_H_ - -#include -#include - -#include "talk/app/webrtc_dev/peerconnection.h" - -namespace webrtc { - -// Implementation of StreamCollection. -class StreamCollectionImpl : public StreamCollectionInterface { - public: - static talk_base::scoped_refptr Create() { - talk_base::RefCountedObject* implementation = - new talk_base::RefCountedObject(); - return implementation; - } - - static talk_base::scoped_refptr Create( - StreamCollectionImpl* streams) { - talk_base::RefCountedObject* implementation = - new talk_base::RefCountedObject(streams); - return implementation; - } - - virtual size_t count() { - return media_streams_.size(); - } - - virtual MediaStreamInterface* at(size_t index) { - return media_streams_.at(index); - } - - virtual MediaStreamInterface* find(const std::string& label) { - for (StreamVector::iterator it = media_streams_.begin(); - it != media_streams_.end(); ++it) { - if ((*it)->label().compare(label) == 0) { - return (*it); - } - } - return NULL; - } - - void AddStream(MediaStreamInterface* stream) { - for (StreamVector::iterator it = media_streams_.begin(); - it != media_streams_.end(); ++it) { - if ((*it)->label().compare(stream->label()) == 0) - return; - } - media_streams_.push_back(stream); - } - - void RemoveStream(MediaStreamInterface* remove_stream) { - for (StreamVector::iterator it = media_streams_.begin(); - it != media_streams_.end(); ++it) { - if ((*it)->label().compare(remove_stream->label()) == 0) { - media_streams_.erase(it); - break; - } - } - } - - protected: - StreamCollectionImpl() {} - explicit StreamCollectionImpl(StreamCollectionImpl* original) - : media_streams_(original->media_streams_) { - } - typedef std::vector > - StreamVector; - StreamVector media_streams_; -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_STREAMCOLLECTIONIMPL_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/test/filevideocapturemodule.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/test/filevideocapturemodule.cc deleted file mode 100644 index 23a8087c1..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/test/filevideocapturemodule.cc +++ /dev/null @@ -1,165 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include "talk/app/webrtc_dev/test/filevideocapturemodule.h" - -#ifdef WEBRTC_RELATIVE_PATH -#include "system_wrappers/interface/ref_count.h" -#else -#include "third_party/webrtc/files/include/ref_count.h" -#endif - -FileVideoCaptureModule::~FileVideoCaptureModule() { - camera_thread_->Stop(); - if (i420_file_ != NULL) fclose(i420_file_); - // The memory associated with video_capture_ is owned by impl_. -} - -webrtc::VideoCaptureModule* -FileVideoCaptureModule::CreateFileVideoCaptureModule(const char* file_name) { - webrtc::RefCountImpl* capture_module = - new webrtc::RefCountImpl(); - if (!capture_module->Init(file_name)) { - capture_module->Release(); - return NULL; - } - return capture_module; -} - -// TODO(henrike): deal with the rounding error. -bool FileVideoCaptureModule::SetFrameRate(int fps) { - fps_ = fps; - time_per_frame_ms_ = 1000 / fps; - return true; -} - -void FileVideoCaptureModule::SetSize(int width, int height) { - width_ = width; - height_ = height; - image_.reset(new uint8[GetI420FrameLength()]); -} - -FileVideoCaptureModule::FileVideoCaptureModule() - : impl_(), - i420_file_(NULL), - camera_thread_(new talk_base::Thread()), - video_capture_(NULL), - started_(false), - sent_frames_(0), - next_frame_time_(0), - time_per_frame_ms_(0), - fps_(0), - width_(0), - height_(0), - image_() {} - -bool FileVideoCaptureModule::Init(const char* file_name) { - impl_ = webrtc::VideoCaptureFactory::Create(0, // id - video_capture_); - if (impl_.get() == NULL) { - return false; - } - if (video_capture_ == NULL) { - return false; - } - if (!SetFrameRate(kStartFrameRate)) { - return false; - } - SetSize(kStartWidth, kStartHeight); - i420_file_ = fopen(file_name, "rb"); - if (i420_file_ == NULL) { - // Not generally unexpected but for this class it is. - ASSERT(false); - return false; - } - if (!camera_thread_->Start()) { - return false; - } - // Only one post, no need to add any data to post. - camera_thread_->Post(this); - return true; -} - -// TODO(henrike): handle time wrapparound. -void FileVideoCaptureModule::GenerateNewFrame() { - if (!started_) { - next_frame_time_ = talk_base::Time(); - started_ = true; - } - // Read from file. - int read = fread(image_.get(), sizeof(uint8), GetI420FrameLength(), - i420_file_); - // Loop file if end is reached. - if (read != GetI420FrameLength()) { - fseek(i420_file_, 0, SEEK_SET); - read = fread(image_.get(), sizeof(uint8), GetI420FrameLength(), - i420_file_); - if (read != GetI420FrameLength()) { - ASSERT(false); - return; - } - } - - webrtc::VideoCaptureCapability capability; - capability.width = width_; - capability.height = height_; - capability.maxFPS = 0; - capability.expectedCaptureDelay = 0; - capability.rawType =webrtc:: kVideoI420; - capability.codecType = webrtc::kVideoCodecUnknown; - capability.interlaced = false; - video_capture_->IncomingFrame(image_.get(), GetI420FrameLength(), - capability, GetTimestamp()); - ++sent_frames_; - next_frame_time_ += time_per_frame_ms_; - const uint32 current_time = talk_base::Time(); - const uint32 wait_time = (next_frame_time_ > current_time) ? - next_frame_time_ - current_time : 0; - camera_thread_->PostDelayed(wait_time, this); -} - -int FileVideoCaptureModule::GetI420FrameLength() { - return width_ * height_ * 3 >> 1; -} - -// TODO(henrike): use this function instead of/in addition to reading from a -// file. -void FileVideoCaptureModule::SetFrame(uint8* image) { - // Set Y plane. - memset(image, 128, width_ * height_); - // Set U plane. - int write_position = width_ * height_; - memset(&image[write_position], 64, width_ * height_ / 4); - // Set V plane. - write_position += width_ * height_ / 4; - memset(&image[write_position], 32, width_ * height_ / 4); -} - -// TODO(henrike): handle timestamp wrapparound. -uint32 FileVideoCaptureModule::GetTimestamp() { - return kStartTimeStamp + sent_frames_ * time_per_frame_ms_; -} diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/test/filevideocapturemodule.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/test/filevideocapturemodule.h deleted file mode 100644 index fa8f6924b..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/test/filevideocapturemodule.h +++ /dev/null @@ -1,203 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -// This class implements the VideoCaptureModule interface. Instead of capturing -// frames from a camera it captures frames from a file. - -#ifndef TALK_APP_WEBRTC_TEST_FILEVIDEOCAPTUREMODULE_H_ -#define TALK_APP_WEBRTC_TEST_FILEVIDEOCAPTUREMODULE_H_ - -#include -#include "talk/base/common.h" -#include "talk/base/scoped_ptr.h" -#include "talk/base/thread.h" -#include "talk/base/time.h" - -#ifdef WEBRTC_RELATIVE_PATH -#include "common_types.h" -#include "modules/video_capture/main/interface/video_capture.h" -#include "modules/video_capture/main/interface/video_capture_defines.h" -#include "modules/video_capture/main/interface/video_capture_factory.h" -#include "system_wrappers/interface/ref_count.h" -#include "system_wrappers/interface/scoped_refptr.h" -#else -#include "third_party/webrtc/files/include/common_types.h" -#include "third_party/webrtc/files/include/video_capture.h" -#include "third_party/webrtc/files/include/video_capture_defines.h" -#include "third_party/webrtc/files/include/video_capture_factory.h" -#include "third_party/webrtc/files/include/ref_count.h" -#include "third_party/webrtc/files/include/scoped_refptr.h" -#endif - -// TODO(henrike): replace playing file with playing a buffer. -class FileVideoCaptureModule - : public webrtc::VideoCaptureModule, - public talk_base::MessageHandler { - public: - virtual ~FileVideoCaptureModule(); - static VideoCaptureModule* CreateFileVideoCaptureModule( - const char* file_name); - - bool SetFrameRate(int fps); - void SetSize(int width, int height); - - virtual int32_t Version(char* version, - uint32_t& remaining_buffer_in_bytes, - uint32_t& position) const { - return impl_->Version(version, remaining_buffer_in_bytes, - position); - } - - virtual int32_t ChangeUniqueId(const int32_t id) { - return impl_->ChangeUniqueId(id); - } - - virtual int32_t TimeUntilNextProcess() { - return impl_->TimeUntilNextProcess(); - } - - virtual int32_t Process() { - return impl_->Process(); - } - - virtual WebRtc_Word32 RegisterCaptureDataCallback( - webrtc::VideoCaptureDataCallback& dataCallback) { - return impl_->RegisterCaptureDataCallback(dataCallback); - } - - virtual WebRtc_Word32 DeRegisterCaptureDataCallback() { - return impl_->DeRegisterCaptureDataCallback(); - } - - virtual WebRtc_Word32 RegisterCaptureCallback( - webrtc::VideoCaptureFeedBack& callBack) { - return impl_->RegisterCaptureCallback(callBack); - } - - virtual WebRtc_Word32 DeRegisterCaptureCallback() { - return impl_->DeRegisterCaptureCallback(); - } - - virtual WebRtc_Word32 StartCapture( - const webrtc::VideoCaptureCapability& capability) { - return impl_->StartCapture(capability); - } - - virtual WebRtc_Word32 StopCapture() { - return impl_->StopCapture(); - } - - virtual WebRtc_Word32 StartSendImage(const webrtc::VideoFrame& videoFrame, - WebRtc_Word32 frameRate = 1) { - return impl_->StartSendImage(videoFrame, frameRate = 1); - } - - virtual WebRtc_Word32 StopSendImage() { - return impl_->StopSendImage(); - } - - virtual const WebRtc_UWord8* CurrentDeviceName() const { - return impl_->CurrentDeviceName(); - } - - virtual bool CaptureStarted() { - return impl_->CaptureStarted(); - } - - virtual WebRtc_Word32 CaptureSettings( - webrtc::VideoCaptureCapability& settings) { - return impl_->CaptureSettings(settings); - } - - virtual WebRtc_Word32 SetCaptureDelay(WebRtc_Word32 delayMS) { - return impl_->SetCaptureDelay(delayMS); - } - - virtual WebRtc_Word32 CaptureDelay() { - return impl_->CaptureDelay(); - } - - virtual WebRtc_Word32 SetCaptureRotation( - webrtc::VideoCaptureRotation rotation) { - return impl_->SetCaptureRotation(rotation); - } - - virtual VideoCaptureEncodeInterface* GetEncodeInterface( - const webrtc::VideoCodec& codec) { - return impl_->GetEncodeInterface(codec); - } - - virtual WebRtc_Word32 EnableFrameRateCallback(const bool enable) { - return impl_->EnableFrameRateCallback(enable); - } - virtual WebRtc_Word32 EnableNoPictureAlarm(const bool enable) { - return impl_->EnableNoPictureAlarm(enable); - } - - // Inherited from MesageHandler. - virtual void OnMessage(talk_base::Message* msg) { - GenerateNewFrame(); - } - - protected: - FileVideoCaptureModule(); - - private: - bool Init(const char* file_name); - void GenerateNewFrame(); - int GetI420FrameLength(); - // Generate an arbitrary frame. (Will be used when file reading is replaced - // with reading a buffer). - void SetFrame(uint8* image); - uint32 GetTimestamp(); - - // Module interface implementation. - webrtc::scoped_refptr impl_; - - // File playing implementation. - static const int kStartFrameRate = 30; - // CIF - static const int kStartWidth = 352; - static const int kStartHeight = 288; - static const uint32 kStartTimeStamp = 2000; - - FILE* i420_file_; - talk_base::scoped_ptr camera_thread_; - webrtc::VideoCaptureExternal* video_capture_; - - bool started_; - int sent_frames_; - uint32 next_frame_time_; - uint32 time_per_frame_ms_; - - int fps_; - int width_; - int height_; - talk_base::scoped_array image_; -}; - -#endif // TALK_APP_WEBRTC_TEST_FILEVIDEOCAPTUREMODULE_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/videorendererimpl.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/videorendererimpl.cc deleted file mode 100644 index 49e57df87..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/videorendererimpl.cc +++ /dev/null @@ -1,58 +0,0 @@ -/* - * libjingle - * Copyright 2004--2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include "talk/app/webrtc_dev/mediastreamimpl.h" - -#include "talk/session/phone/videorenderer.h" - -namespace webrtc { - -// VideoRendererImpl take ownership of cricket::VideoRenderer. -class VideoRendererImpl : public VideoRendererWrapperInterface { - public: - explicit VideoRendererImpl(cricket::VideoRenderer* renderer) - : renderer_(renderer) { - } - virtual cricket::VideoRenderer* renderer() { - return renderer_; - } - protected: - ~VideoRendererImpl() { - delete renderer_; - } - private: - cricket::VideoRenderer* renderer_; -}; - -talk_base::scoped_refptr CreateVideoRenderer( - cricket::VideoRenderer* renderer) { - talk_base::RefCountedObject* r = - new talk_base::RefCountedObject(renderer); - return r; -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/videotrackimpl.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/videotrackimpl.cc deleted file mode 100644 index d9f9e524e..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/videotrackimpl.cc +++ /dev/null @@ -1,79 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ -#include "talk/app/webrtc_dev/videotrackimpl.h" - -#include - -namespace webrtc { - -static const char kVideoTrackKind[] = "video"; - -VideoTrack::VideoTrack(const std::string& label) - : MediaTrack(label), - video_device_(NULL) { -} - -VideoTrack::VideoTrack(const std::string& label, - VideoCaptureModule* video_device) - : MediaTrack(label), - video_device_(video_device) { -} - -void VideoTrack::SetRenderer(VideoRendererWrapperInterface* renderer) { - video_renderer_ = renderer; - Notifier::FireOnChanged(); -} - -VideoRendererWrapperInterface* VideoTrack::GetRenderer() { - return video_renderer_.get(); -} - - // Get the VideoCapture device associated with this track. -VideoCaptureModule* VideoTrack::GetVideoCapture() { - return video_device_.get(); -} - -std::string VideoTrack::kind() const { - return std::string(kVideoTrackKind); -} - -talk_base::scoped_refptr VideoTrack::CreateRemote( - const std::string& label) { - talk_base::RefCountedObject* track = - new talk_base::RefCountedObject(label); - return track; -} - -talk_base::scoped_refptr VideoTrack::CreateLocal( - const std::string& label, - VideoCaptureModule* video_device) { - talk_base::RefCountedObject* track = - new talk_base::RefCountedObject(label, video_device); - return track; -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/videotrackimpl.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/videotrackimpl.h deleted file mode 100644 index 33d982034..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/videotrackimpl.h +++ /dev/null @@ -1,73 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#ifndef TALK_APP_WEBRTC_VIDEOTRACKIMPL_H_ -#define TALK_APP_WEBRTC_VIDEOTRACKIMPL_H_ - -#include - -#include "talk/app/webrtc_dev/mediastream.h" -#include "talk/app/webrtc_dev/mediatrackimpl.h" -#include "talk/app/webrtc_dev/notifierimpl.h" -#include "talk/base/scoped_refptr.h" - -#ifdef WEBRTC_RELATIVE_PATH -#include "modules/video_capture/main/interface/video_capture.h" -#else -#include "third_party/webrtc/files/include/video_capture.h" -#endif - -namespace webrtc { - -class VideoTrack : public MediaTrack { - public: - // Create a video track used for remote video tracks. - static talk_base::scoped_refptr CreateRemote( - const std::string& label); - // Create a video track used for local video tracks. - static talk_base::scoped_refptr CreateLocal( - const std::string& label, - VideoCaptureModule* video_device); - - virtual VideoCaptureModule* GetVideoCapture(); - virtual void SetRenderer(VideoRendererWrapperInterface* renderer); - VideoRendererWrapperInterface* GetRenderer(); - - virtual std::string kind() const; - - protected: - explicit VideoTrack(const std::string& label); - VideoTrack(const std::string& label, VideoCaptureModule* video_device); - - private: - talk_base::scoped_refptr video_device_; - talk_base::scoped_refptr video_renderer_; -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_VIDEOTRACKIMPL_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcjson.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcjson.cc deleted file mode 100644 index 313876ba5..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcjson.cc +++ /dev/null @@ -1,639 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include "talk/app/webrtc_dev/webrtcjson.h" - -#include -#include -#include - -#include "talk/base/json.h" -#include "talk/base/logging.h" -#include "talk/base/stringutils.h" -#include "talk/session/phone/codec.h" -#include "talk/session/phone/cryptoparams.h" -#include "talk/session/phone/mediasession.h" -#include "talk/session/phone/mediasessionclient.h" - -namespace webrtc { -static const int kIceComponent = 1; -static const int kIceFoundation = 1; -static const char* kMessageType[] = { - "OFFER", - "ANSWER", - "ERROR", -}; - -static std::vector ReadValues(const Json::Value& value, - const std::string& key); - -static void BuildContent( - const cricket::SessionDescription* sdp, - const cricket::ContentInfo& content_info, - const std::vector& candidates, - bool video, - Json::Value* content); - -static void BuildCandidate(const std::vector& candidates, - bool video, - std::vector* jcandidates); - -static void BuildRtpMapParams(const cricket::ContentInfo& content_info, - bool video, - std::vector* rtpmap); - -static void BuildCrypto(const cricket::ContentInfo& content_info, - bool video, - std::vector* cryptos); - -static void BuildTrack(const cricket::SessionDescription* sdp, - bool video, - std::vector* track); - -static std::string Serialize(const Json::Value& value); - -static bool Deserialize(const std::string& message, Json::Value& value); - -bool ParseContent(const Json::Value& jmessage, - cricket::SessionDescription* sdp, - std::vector* candidates); - -static bool ParseAudioCodec(const Json::Value& value, - cricket::AudioContentDescription* content); -static bool ParseVideoCodec(const Json::Value& value, - cricket::VideoContentDescription* content); -static bool ParseCrypto(const Json::Value& content, - cricket::MediaContentDescription* desc); - -static bool ParseCandidates(const Json::Value& content, - std::vector* candidates); - -static bool ParseTrack(const Json::Value& content, - cricket::MediaContentDescription* content_desc); - -static void Append(Json::Value* object, const std::string& key, bool value); -static void Append(Json::Value* object, const std::string& key, - const char* value); -static void Append(Json::Value* object, const std::string& key, int value); -static void Append(Json::Value* object, const std::string& key, - const std::string& value); -static void Append(Json::Value* object, const std::string& key, uint32 value); -static void Append(Json::Value* object, const std::string& key, - const Json::Value& value); -static void Append(Json::Value* object, - const std::string& key, - const std::vector& values); - -std::string JsonSerialize( - const webrtc::PeerConnectionMessage::PeerConnectionMessageType type, - int error_code, - const cricket::SessionDescription* sdp, - const std::vector& candidates) { - Json::Value media; - // TODO(ronghuawu): Replace magic strings. - Append(&media, "SDP", kMessageType[type]); - - if (type == webrtc::PeerConnectionMessage::kError) { - Append(&media, "error_code", error_code); - return Serialize(media); - } - - const cricket::ContentInfo* audio_content = GetFirstAudioContent(sdp); - const cricket::ContentInfo* video_content = GetFirstVideoContent(sdp); - - std::vector together; - together.push_back("audio"); - together.push_back("video"); - - std::vector contents; - - if (audio_content) { - Json::Value content; - BuildContent(sdp, *audio_content, candidates, false, &content); - contents.push_back(content); - } - - if (video_content) { - Json::Value content; - BuildContent(sdp, *video_content, candidates, true, &content); - contents.push_back(content); - } - - Append(&media, "content", contents); - Append(&media, "TOGETHER", together); - - // Now serialize. - return Serialize(media); -} - -void BuildContent( - const cricket::SessionDescription* sdp, - const cricket::ContentInfo& content_info, - const std::vector& candidates, - bool video, - Json::Value* content) { - std::string label("media"); - // TODO(ronghuawu): Use enum instead of bool video to prepare for other - // media types such as the data media stream. - if (video) { - Append(content, label, "video"); - } else { - Append(content, label, "audio"); - } - - const cricket::MediaContentDescription* media_info = - static_cast ( - content_info.description); - if (media_info->rtcp_mux()) { - Append(content, "rtcp_mux", true); - } - - // rtpmap - std::vector rtpmap; - BuildRtpMapParams(content_info, video, &rtpmap); - Append(content, "rtpmap", rtpmap); - - // crypto - std::vector crypto; - BuildCrypto(content_info, video, &crypto); - Append(content, "crypto", crypto); - - // candidate - std::vector jcandidates; - BuildCandidate(candidates, video, &jcandidates); - Append(content, "candidate", jcandidates); - - // track - std::vector track; - BuildTrack(sdp, video, &track); - Append(content, "track", track); -} - -void BuildRtpMapParams(const cricket::ContentInfo& content_info, - bool video, - std::vector* rtpmap) { - if (!video) { - const cricket::AudioContentDescription* audio = - static_cast( - content_info.description); - - std::vector::const_iterator iter = - audio->codecs().begin(); - std::vector::const_iterator iter_end = - audio->codecs().end(); - for (; iter != iter_end; ++iter) { - Json::Value codec; - std::string codec_str(std::string("audio/").append(iter->name)); - // adding clockrate - Append(&codec, "clockrate", iter->clockrate); - Append(&codec, "codec", codec_str); - Json::Value codec_id; - Append(&codec_id, talk_base::ToString(iter->id), codec); - rtpmap->push_back(codec_id); - } - } else { - const cricket::VideoContentDescription* video = - static_cast( - content_info.description); - - std::vector::const_iterator iter = - video->codecs().begin(); - std::vector::const_iterator iter_end = - video->codecs().end(); - for (; iter != iter_end; ++iter) { - Json::Value codec; - std::string codec_str(std::string("video/").append(iter->name)); - Append(&codec, "codec", codec_str); - Json::Value codec_id; - Append(&codec_id, talk_base::ToString(iter->id), codec); - rtpmap->push_back(codec_id); - } - } -} - -void BuildCrypto(const cricket::ContentInfo& content_info, - bool video, - std::vector* cryptos) { - const cricket::MediaContentDescription* content_desc = - static_cast( - content_info.description); - std::vector::const_iterator iter = - content_desc->cryptos().begin(); - std::vector::const_iterator iter_end = - content_desc->cryptos().end(); - for (; iter != iter_end; ++iter) { - Json::Value crypto; - Append(&crypto, "cipher_suite", iter->cipher_suite); - Append(&crypto, "key_params", iter->key_params); - cryptos->push_back(crypto); - } -} - -void BuildCandidate(const std::vector& candidates, - bool video, - std::vector* jcandidates) { - std::vector::const_iterator iter = - candidates.begin(); - std::vector::const_iterator iter_end = - candidates.end(); - for (; iter != iter_end; ++iter) { - if ((video && (!iter->name().compare("video_rtcp") || - (!iter->name().compare("video_rtp")))) || - (!video && (!iter->name().compare("rtp") || - (!iter->name().compare("rtcp"))))) { - Json::Value jcandidate; - Append(&jcandidate, "component", kIceComponent); - Append(&jcandidate, "foundation", kIceFoundation); - Append(&jcandidate, "generation", iter->generation()); - Append(&jcandidate, "proto", iter->protocol()); - Append(&jcandidate, "priority", iter->preference_str()); - Append(&jcandidate, "ip", iter->address().IPAsString()); - Append(&jcandidate, "port", iter->address().PortAsString()); - Append(&jcandidate, "type", iter->type()); - Append(&jcandidate, "name", iter->name()); - Append(&jcandidate, "network_name", iter->network_name()); - Append(&jcandidate, "username", iter->username()); - Append(&jcandidate, "password", iter->password()); - jcandidates->push_back(jcandidate); - } - } -} - -void BuildTrack(const cricket::SessionDescription* sdp, - bool video, - std::vector* tracks) { - const cricket::ContentInfo* content; - if (video) - content = GetFirstVideoContent(sdp); - else - content = GetFirstAudioContent(sdp); - - if (!content) - return; - - const cricket::MediaContentDescription* desc = - static_cast( - content->description); - for (cricket::StreamParamsVec::const_iterator it = desc->streams().begin(); - it != desc->streams().end(); - ++it) { - // TODO(ronghuawu): Support ssrcsgroups. - Json::Value track; - ASSERT(it->ssrcs.size() == 1); - Append(&track, "ssrc", it->ssrcs[0]); - Append(&track, "cname", it->cname); - Append(&track, "stream_label", it->sync_label); - Append(&track, "label", it->name); - tracks->push_back(track); - } -} - -std::string Serialize(const Json::Value& value) { - Json::StyledWriter writer; - return writer.write(value); -} - -bool Deserialize(const std::string& message, Json::Value* value) { - Json::Reader reader; - return reader.parse(message, *value); -} - -bool JsonDeserialize( - webrtc::PeerConnectionMessage::PeerConnectionMessageType* type, - webrtc::PeerConnectionMessage::ErrorCode* error_code, - cricket::SessionDescription* sdp, - std::vector* candidates, - const std::string& signaling_message) { - ASSERT(type); - ASSERT(error_code); - ASSERT(sdp); - ASSERT(candidates); - if (type == NULL || error_code == NULL || sdp == NULL || candidates == NULL) - return false; - - // first deserialize message - Json::Value jmessage; - if (!Deserialize(signaling_message, &jmessage)) { - return false; - } - - // Get the message type - std::string message_type; - bool valid_message_type = false; - if (!GetStringFromJsonObject(jmessage, "SDP", &message_type)) - return false; - for (int i = 0; i < ARRAY_SIZE(kMessageType); i++) { - if (message_type.compare(kMessageType[i]) == 0) { - *type = static_cast< - webrtc::PeerConnectionMessage::PeerConnectionMessageType>(i); - valid_message_type = true; - break; - } - } - if (!valid_message_type) - return false; - if (*type == webrtc::PeerConnectionMessage::kError) { - int code; - if (!GetIntFromJsonObject(jmessage, "error_code", &code)) - return false; - *error_code = static_cast(code); - return true; - } - - return ParseContent(jmessage, sdp, candidates); -} - -bool ParseContent(const Json::Value& jmessage, - cricket::SessionDescription* sdp, - std::vector* candidates) { - // Get content - std::vector contents = ReadValues(jmessage, "content"); - if (contents.size() == 0) - return false; - for (size_t i = 0; i < contents.size(); ++i) { - Json::Value content = contents[i]; - // candidates - if (!ParseCandidates(content, candidates)) - return false; - // rtcp_mux - bool rtcp_mux; - if (!GetBoolFromJsonObject(content, "rtcp_mux", &rtcp_mux)) - rtcp_mux = false; - // rtpmap - if (content["media"].asString().compare("audio") == 0) { - cricket::AudioContentDescription* audio_content = - new cricket::AudioContentDescription(); - if (!ParseAudioCodec(content, audio_content)) - return false; - audio_content->set_rtcp_mux(rtcp_mux); - audio_content->SortCodecs(); - // crypto - if (!ParseCrypto(content, audio_content)) - return false; - // tracks - if (!ParseTrack(content, audio_content)) - return false; - (sdp)->AddContent(cricket::CN_AUDIO, - cricket::NS_JINGLE_RTP, audio_content); - } else if (content["media"].asString().compare("video") == 0) { - cricket::VideoContentDescription* video_content = - new cricket::VideoContentDescription(); - if (!ParseVideoCodec(content, video_content)) - return false; - video_content->set_rtcp_mux(rtcp_mux); - video_content->SortCodecs(); - // crypto - if (!ParseCrypto(content, video_content)) - return false; - if (!ParseTrack(content, video_content)) - return false; - (sdp)->AddContent(cricket::CN_VIDEO, - cricket::NS_JINGLE_RTP, video_content); - } - } - return true; -} - -bool ParseAudioCodec(const Json::Value& value, - cricket::AudioContentDescription* content) { - std::vector rtpmap(ReadValues(value, "rtpmap")); - // When there's no codecs in common, rtpmap can be empty. - if (rtpmap.empty()) - return true; - - std::vector::const_iterator iter = - rtpmap.begin(); - std::vector::const_iterator iter_end = - rtpmap.end(); - for (; iter != iter_end; ++iter) { - cricket::AudioCodec codec; - std::string pltype(iter->begin().memberName()); - talk_base::FromString(pltype, &codec.id); - Json::Value codec_info((*iter)[pltype]); - std::string codec_name; - if (!GetStringFromJsonObject(codec_info, "codec", &codec_name)) - continue; - std::vector tokens; - talk_base::split(codec_name, '/', &tokens); - codec.name = tokens[1]; - GetIntFromJsonObject(codec_info, "clockrate", &codec.clockrate); - content->AddCodec(codec); - } - - return true; -} - -bool ParseVideoCodec(const Json::Value& value, - cricket::VideoContentDescription* content) { - std::vector rtpmap(ReadValues(value, "rtpmap")); - // When there's no codecs in common, rtpmap can be empty. - if (rtpmap.empty()) - return true; - - std::vector::const_iterator iter = - rtpmap.begin(); - std::vector::const_iterator iter_end = - rtpmap.end(); - for (; iter != iter_end; ++iter) { - cricket::VideoCodec codec; - std::string pltype(iter->begin().memberName()); - talk_base::FromString(pltype, &codec.id); - Json::Value codec_info((*iter)[pltype]); - std::vector tokens; - talk_base::split(codec_info["codec"].asString(), '/', &tokens); - codec.name = tokens[1]; - content->AddCodec(codec); - } - return true; -} - -bool ParseCandidates(const Json::Value& content, - std::vector* candidates) { - std::vector jcandidates(ReadValues(content, "candidate")); - std::vector::const_iterator iter = - jcandidates.begin(); - std::vector::const_iterator iter_end = - jcandidates.end(); - for (; iter != iter_end; ++iter) { - cricket::Candidate cand; - - unsigned int generation; - if (!GetUIntFromJsonObject(*iter, "generation", &generation)) - return false; - cand.set_generation_str(talk_base::ToString(generation)); - - std::string proto; - if (!GetStringFromJsonObject(*iter, "proto", &proto)) - return false; - cand.set_protocol(proto); - - std::string priority; - if (!GetStringFromJsonObject(*iter, "priority", &priority)) - return false; - cand.set_preference_str(priority); - - std::string str; - talk_base::SocketAddress addr; - if (!GetStringFromJsonObject(*iter, "ip", &str)) - return false; - addr.SetIP(str); - if (!GetStringFromJsonObject(*iter, "port", &str)) - return false; - int port; - if (!talk_base::FromString(str, &port)) - return false; - addr.SetPort(port); - cand.set_address(addr); - - if (!GetStringFromJsonObject(*iter, "type", &str)) - return false; - cand.set_type(str); - - if (!GetStringFromJsonObject(*iter, "name", &str)) - return false; - cand.set_name(str); - - if (!GetStringFromJsonObject(*iter, "network_name", &str)) - return false; - cand.set_network_name(str); - - if (!GetStringFromJsonObject(*iter, "username", &str)) - return false; - cand.set_username(str); - - if (!GetStringFromJsonObject(*iter, "password", &str)) - return false; - cand.set_password(str); - - candidates->push_back(cand); - } - return true; -} - -bool ParseCrypto(const Json::Value& content, - cricket::MediaContentDescription* desc) { - std::vector jcryptos(ReadValues(content, "crypto")); - std::vector::const_iterator iter = - jcryptos.begin(); - std::vector::const_iterator iter_end = - jcryptos.end(); - for (; iter != iter_end; ++iter) { - cricket::CryptoParams crypto; - - std::string cipher_suite; - if (!GetStringFromJsonObject(*iter, "cipher_suite", &cipher_suite)) - return false; - crypto.cipher_suite = cipher_suite; - - std::string key_params; - if (!GetStringFromJsonObject(*iter, "key_params", &key_params)) - return false; - crypto.key_params= key_params; - - desc->AddCrypto(crypto); - } - return true; -} - -bool ParseTrack(const Json::Value& content, - cricket::MediaContentDescription* content_desc) { - ASSERT(content_desc); - if (!content_desc) - return false; - - std::vector tracks(ReadValues(content, "track")); - std::vector::const_iterator iter = - tracks.begin(); - std::vector::const_iterator iter_end = - tracks.end(); - for (; iter != iter_end; ++iter) { - uint32 ssrc; - std::string label; - std::string cname; - std::string stream_label; - if (!GetUIntFromJsonObject(*iter, "ssrc", &ssrc)) - return false; - // label is optional, it will be empty string if doesn't exist - GetStringFromJsonObject(*iter, "label", &label); - if (!GetStringFromJsonObject(*iter, "cname", &cname)) - return false; - // stream_label is optional, it will be the same as cname if it - // doesn't exist. - GetStringFromJsonObject(*iter, "stream_label", &stream_label); - if (stream_label.empty()) - stream_label = cname; - content_desc->AddStream(cricket::StreamParams(label, ssrc, cname, - stream_label)); - } - return true; -} - -std::vector ReadValues( - const Json::Value& value, const std::string& key) { - std::vector objects; - for (Json::Value::ArrayIndex i = 0; i < value[key].size(); ++i) { - objects.push_back(value[key][i]); - } - return objects; -} - -void Append(Json::Value* object, const std::string& key, bool value) { - (*object)[key] = Json::Value(value); -} - -void Append(Json::Value* object, const std::string& key, const char* value) { - (*object)[key] = Json::Value(value); -} - -void Append(Json::Value* object, const std::string& key, int value) { - (*object)[key] = Json::Value(value); -} - -void Append(Json::Value* object, const std::string& key, - const std::string& value) { - (*object)[key] = Json::Value(value); -} - -void Append(Json::Value* object, const std::string& key, uint32 value) { - (*object)[key] = Json::Value(value); -} - -void Append(Json::Value* object, const std::string& key, - const Json::Value& value) { - (*object)[key] = value; -} - -void Append(Json::Value* object, - const std::string & key, - const std::vector& values) { - for (std::vector::const_iterator iter = values.begin(); - iter != values.end(); ++iter) { - (*object)[key].append(*iter); - } -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcjson.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcjson.h deleted file mode 100644 index 837543efa..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcjson.h +++ /dev/null @@ -1,62 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#ifndef TALK_APP_WEBRTC_WEBRTCJSON_H_ -#define TALK_APP_WEBRTC_WEBRTCJSON_H_ - -#include -#include - -#ifdef WEBRTC_RELATIVE_PATH -#include "json/json.h" -#else -#include "third_party/jsoncpp/json.h" -#endif -#include "talk/app/webrtc_dev/peerconnectionmessage.h" -#include "talk/p2p/base/candidate.h" - -namespace cricket { -class SessionDescription; -} - -namespace webrtc { - -std::string JsonSerialize( - const webrtc::PeerConnectionMessage::PeerConnectionMessageType type, - int error_code, - const cricket::SessionDescription* sdp, - const std::vector& candidates); - -bool JsonDeserialize( - webrtc::PeerConnectionMessage::PeerConnectionMessageType* type, - webrtc::PeerConnectionMessage::ErrorCode* error_code, - cricket::SessionDescription* sdp, - std::vector* candidates, - const std::string& signaling_message); -} - -#endif // TALK_APP_WEBRTC_WEBRTCJSON_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsession.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsession.cc deleted file mode 100644 index dd4a8c98c..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsession.cc +++ /dev/null @@ -1,335 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include "talk/app/webrtc_dev/webrtcsession.h" - -#include "talk/app/webrtc_dev/mediastream.h" -#include "talk/app/webrtc_dev/peerconnection.h" -#include "talk/app/webrtc_dev/peerconnectionsignaling.h" -#include "talk/base/helpers.h" -#include "talk/base/logging.h" -#include "talk/session/phone/channel.h" -#include "talk/session/phone/channelmanager.h" -#include "talk/session/phone/mediasession.h" - -using cricket::MediaContentDescription; - -namespace webrtc { - -enum { - MSG_CANDIDATE_TIMEOUT = 101, -}; - -// We allow 30 seconds to establish a connection, otherwise it's an error. -static const int kCallSetupTimeout = 30 * 1000; -// Session will accept one candidate per transport channel and dropping other -// candidates generated for that channel. During the session initialization -// one cricket::VoiceChannel and one cricket::VideoChannel will be created with -// rtcp enabled. -static const int kAllowedCandidates = 4; -// TODO(mallinath) - These are magic string used by cricket::VideoChannel. -// These should be moved to a common place. -static const std::string kRtpVideoChannelStr = "video_rtp"; -static const std::string kRtcpVideoChannelStr = "video_rtcp"; - -WebRtcSession::WebRtcSession(cricket::ChannelManager* channel_manager, - talk_base::Thread* signaling_thread, - talk_base::Thread* worker_thread, - cricket::PortAllocator* port_allocator) - : cricket::BaseSession(signaling_thread, worker_thread, port_allocator, - talk_base::ToString(talk_base::CreateRandomId()), - cricket::NS_JINGLE_RTP, true), - channel_manager_(channel_manager), - observer_(NULL), - session_desc_factory_(channel_manager) { -} - -WebRtcSession::~WebRtcSession() { - Terminate(); -} - -bool WebRtcSession::Initialize() { - return CreateChannels(); -} - -void WebRtcSession::Terminate() { - if (voice_channel_.get()) { - channel_manager_->DestroyVoiceChannel(voice_channel_.release()); - } - if (video_channel_.get()) { - channel_manager_->DestroyVideoChannel(video_channel_.release()); - } -} - -bool WebRtcSession::CreateChannels() { - voice_channel_.reset(channel_manager_->CreateVoiceChannel( - this, cricket::CN_AUDIO, true)); - if (!voice_channel_.get()) { - LOG(LS_ERROR) << "Failed to create voice channel"; - return false; - } - - video_channel_.reset(channel_manager_->CreateVideoChannel( - this, cricket::CN_VIDEO, true, voice_channel_.get())); - if (!video_channel_.get()) { - LOG(LS_ERROR) << "Failed to create video channel"; - return false; - } - - // TransportProxies and TransportChannels will be created when - // CreateVoiceChannel and CreateVideoChannel are called. - // Try connecting all transport channels. This is necessary to generate - // ICE candidates. - SpeculativelyConnectAllTransportChannels(); - return true; -} - -void WebRtcSession::SetRemoteCandidates( - const cricket::Candidates& candidates) { - // First partition the candidates for the proxies. During creation of channels - // we created CN_AUDIO (audio) and CN_VIDEO (video) proxies. - cricket::Candidates audio_candidates; - cricket::Candidates video_candidates; - for (cricket::Candidates::const_iterator citer = candidates.begin(); - citer != candidates.end(); ++citer) { - if (((*citer).name().compare(kRtpVideoChannelStr) == 0) || - ((*citer).name().compare(kRtcpVideoChannelStr)) == 0) { - // Candidate names for video rtp and rtcp channel - video_candidates.push_back(*citer); - } else { - // Candidates for audio rtp and rtcp channel - // Channel name will be "rtp" and "rtcp" - audio_candidates.push_back(*citer); - } - } - - if (!audio_candidates.empty()) { - cricket::TransportProxy* audio_proxy = GetTransportProxy(cricket::CN_AUDIO); - if (audio_proxy) { - // CompleteNegotiation will set actual impl's in Proxy. - if (!audio_proxy->negotiated()) - audio_proxy->CompleteNegotiation(); - // TODO(mallinath) - Add a interface to TransportProxy to accept - // remote candidate list. - audio_proxy->impl()->OnRemoteCandidates(audio_candidates); - } else { - LOG(LS_INFO) << "No audio TransportProxy exists"; - } - } - - if (!video_candidates.empty()) { - cricket::TransportProxy* video_proxy = GetTransportProxy(cricket::CN_VIDEO); - if (video_proxy) { - // CompleteNegotiation will set actual impl's in Proxy. - if (!video_proxy->negotiated()) - video_proxy->CompleteNegotiation(); - // TODO(mallinath) - Add a interface to TransportProxy to accept - // remote candidate list. - video_proxy->impl()->OnRemoteCandidates(video_candidates); - } else { - LOG(LS_INFO) << "No video TransportProxy exists"; - } - } -} - -void WebRtcSession::OnTransportRequestSignaling( - cricket::Transport* transport) { - ASSERT(signaling_thread()->IsCurrent()); - transport->OnSignalingReady(); -} - -void WebRtcSession::OnTransportConnecting(cricket::Transport* transport) { - ASSERT(signaling_thread()->IsCurrent()); - // start monitoring for the write state of the transport. - OnTransportWritable(transport); -} - -void WebRtcSession::OnTransportWritable(cricket::Transport* transport) { - ASSERT(signaling_thread()->IsCurrent()); - // If the transport is not in writable state, start a timer to monitor - // the state. If the transport doesn't become writable state in 30 seconds - // then we are assuming call can't be continued. - signaling_thread()->Clear(this, MSG_CANDIDATE_TIMEOUT); - if (transport->HasChannels() && !transport->writable()) { - signaling_thread()->PostDelayed( - kCallSetupTimeout, this, MSG_CANDIDATE_TIMEOUT); - } -} - -void WebRtcSession::OnTransportCandidatesReady( - cricket::Transport* transport, const cricket::Candidates& candidates) { - ASSERT(signaling_thread()->IsCurrent()); - // Drop additional candidates for the same channel; - // local_candidates_ will have one candidate per channel. - if (local_candidates_.size() == kAllowedCandidates) - return; - InsertTransportCandidates(candidates); - if (local_candidates_.size() == kAllowedCandidates && observer_) { - observer_->OnCandidatesReady(local_candidates_); - } -} - -void WebRtcSession::OnTransportChannelGone(cricket::Transport* transport) { - ASSERT(signaling_thread()->IsCurrent()); -} - -void WebRtcSession::OnMessage(talk_base::Message* msg) { - switch (msg->message_id) { - case MSG_CANDIDATE_TIMEOUT: - LOG(LS_ERROR) << "Transport is not in writable state."; - SignalError(); - break; - default: - break; - } -} - -void WebRtcSession::InsertTransportCandidates( - const cricket::Candidates& candidates) { - for (cricket::Candidates::const_iterator citer = candidates.begin(); - citer != candidates.end(); ++citer) { - // Find candidates by name, if this channel name not exists in local - // candidate list, store it. - if (!CheckCandidate((*citer).name())) { - local_candidates_.push_back(*citer); - } - } -} - -// Check transport candidate already available for transport channel as only -// one cricket::Candidate allower per channel. -bool WebRtcSession::CheckCandidate(const std::string& name) { - bool ret = false; - for (cricket::Candidates::iterator iter = local_candidates_.begin(); - iter != local_candidates_.end(); ++iter) { - if ((*iter).name().compare(name) == 0) { - ret = true; - break; - } - } - return ret; -} - -void WebRtcSession::SetCaptureDevice(const std::string& name, - VideoCaptureModule* camera) { - // should be called from a signaling thread - ASSERT(signaling_thread()->IsCurrent()); - - // TODO(mallinath): Refactor this when there is support for multiple cameras. - - // Register the the VideoCapture Module. - // TODO(mallinath): Fix SetCaptureDevice. - video_channel_->SetCaptureDevice(0, camera); - - // Actually associate the video capture module with the ViE channel. - channel_manager_->SetVideoOptions(""); -} - -void WebRtcSession::SetLocalRenderer(const std::string& name, - cricket::VideoRenderer* renderer) { - ASSERT(signaling_thread()->IsCurrent()); - // TODO(mallinath): Fix SetLocalRenderer. - video_channel_->SetLocalRenderer(0, renderer); -} - -void WebRtcSession::SetRemoteRenderer(const std::string& name, - cricket::VideoRenderer* renderer) { - ASSERT(signaling_thread()->IsCurrent()); - - // TODO(mallinath): Only the ssrc = 0 is supported at the moment. - // Only one channel. - video_channel_->SetRenderer(0, renderer); -} - -const cricket::SessionDescription* WebRtcSession::ProvideOffer( - const cricket::MediaSessionOptions& options) { - // TODO(mallinath) - Sanity check for options. - cricket::SessionDescription* offer( - session_desc_factory_.CreateOffer(options, local_description())); - set_local_description(offer); - return offer; -} - -const cricket::SessionDescription* WebRtcSession::SetRemoteSessionDescription( - const cricket::SessionDescription* remote_offer, - const std::vector& remote_candidates) { - set_remote_description( - const_cast(remote_offer)); - SetRemoteCandidates(remote_candidates); - return remote_offer; -} - -const cricket::SessionDescription* WebRtcSession::ProvideAnswer( - const cricket::MediaSessionOptions& options) { - cricket::SessionDescription* answer( - session_desc_factory_.CreateAnswer(remote_description(), options, - local_description())); - set_local_description(answer); - return answer; -} - -void WebRtcSession::NegotiationDone() { - // SetState of session is called after session receives both local and - // remote descriptions. State transition will happen only when session - // is in INIT state. - if (state() == STATE_INIT) { - SetState(STATE_SENTINITIATE); - SetState(STATE_RECEIVEDACCEPT); - - // Enabling voice and video channel. - voice_channel_->Enable(true); - video_channel_->Enable(true); - } - - const cricket::ContentInfo* audio_info = - cricket::GetFirstAudioContent(local_description()); - if (audio_info) { - const cricket::MediaContentDescription* audio_content = - static_cast( - audio_info->description); - // Since channels are currently not supporting multiple send streams, - // we can remove stream from a session by muting it. - // TODO(mallinath) - Change needed when multiple send streams support - // is available. - voice_channel_->Mute(audio_content->streams().size() == 0); - } - - const cricket::ContentInfo* video_info = - cricket::GetFirstVideoContent(local_description()); - if (video_info) { - const cricket::MediaContentDescription* video_content = - static_cast( - video_info->description); - // Since channels are currently not supporting multiple send streams, - // we can remove stream from a session by muting it. - // TODO(mallinath) - Change needed when multiple send streams support - // is available. - video_channel_->Mute(video_content->streams().size() == 0); - } -} - -} // namespace webrtc diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsession.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsession.h deleted file mode 100644 index 223bd90ee..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsession.h +++ /dev/null @@ -1,126 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#ifndef TALK_APP_WEBRTC_WEBRTCSESSION_H_ -#define TALK_APP_WEBRTC_WEBRTCSESSION_H_ - -#include -#include - -#include "talk/app/webrtc_dev/mediastreamprovider.h" -#include "talk/app/webrtc_dev/sessiondescriptionprovider.h" -#include "talk/app/webrtc_dev/webrtcsessionobserver.h" -#include "talk/base/sigslot.h" -#include "talk/base/thread.h" -#include "talk/p2p/base/session.h" -#include "talk/session/phone/mediasession.h" - -namespace cricket { -class ChannelManager; -class Transport; -class VideoChannel; -class VoiceChannel; -} - -namespace webrtc { - -class WebRtcSession : public cricket::BaseSession, - public MediaProviderInterface, - public SessionDescriptionProvider { - public: - WebRtcSession(cricket::ChannelManager* channel_manager, - talk_base::Thread* signaling_thread, - talk_base::Thread* worker_thread, - cricket::PortAllocator* port_allocator); - ~WebRtcSession(); - - bool Initialize(); - - void RegisterObserver(WebRtcSessionObserver* observer) { - observer_ = observer; - } - - const cricket::VoiceChannel* voice_channel() const { - return voice_channel_.get(); - } - const cricket::VideoChannel* video_channel() const { - return video_channel_.get(); - } - - // Generic error message callback from WebRtcSession. - // TODO(mallinath) - It may be necessary to supply error code as well. - sigslot::signal0<> SignalError; - - private: - // Implements SessionDescriptionProvider - virtual const cricket::SessionDescription* ProvideOffer( - const cricket::MediaSessionOptions& options); - virtual const cricket::SessionDescription* SetRemoteSessionDescription( - const cricket::SessionDescription* remote_offer, - const std::vector& remote_candidates); - virtual const cricket::SessionDescription* ProvideAnswer( - const cricket::MediaSessionOptions& options); - virtual void NegotiationDone(); - - // Implements MediaProviderInterface. - virtual void SetCaptureDevice(const std::string& name, - VideoCaptureModule* camera); - virtual void SetLocalRenderer(const std::string& name, - cricket::VideoRenderer* renderer); - virtual void SetRemoteRenderer(const std::string& name, - cricket::VideoRenderer* renderer); - - // Transport related callbacks, override from cricket::BaseSession. - virtual void OnTransportRequestSignaling(cricket::Transport* transport); - virtual void OnTransportConnecting(cricket::Transport* transport); - virtual void OnTransportWritable(cricket::Transport* transport); - virtual void OnTransportCandidatesReady( - cricket::Transport* transport, - const cricket::Candidates& candidates); - virtual void OnTransportChannelGone(cricket::Transport* transport); - - // Creates channels for voice and video. - bool CreateChannels(); - virtual void OnMessage(talk_base::Message* msg); - void InsertTransportCandidates(const cricket::Candidates& candidates); - void Terminate(); - // Get candidate from the local candidates list by the name. - bool CheckCandidate(const std::string& name); - void SetRemoteCandidates(const cricket::Candidates& candidates); - - private: - talk_base::scoped_ptr voice_channel_; - talk_base::scoped_ptr video_channel_; - cricket::ChannelManager* channel_manager_; - cricket::Candidates local_candidates_; - WebRtcSessionObserver* observer_; - cricket::MediaSessionDescriptionFactory session_desc_factory_; -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_ diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsession_unittest.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsession_unittest.cc deleted file mode 100644 index e4f0823e7..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsession_unittest.cc +++ /dev/null @@ -1,135 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include "gtest/gtest.h" -#include "talk/app/webrtc_dev/webrtcsession.h" -#include "talk/app/webrtc_dev/peerconnectionsignaling.h" -#include "talk/base/thread.h" -#include "talk/session/phone/channelmanager.h" -#include "talk/p2p/client/fakeportallocator.h" - -class MockWebRtcSessionObserver : public webrtc::WebRtcSessionObserver { - public: - virtual void OnCandidatesReady( - const std::vector& candidates) { - for (cricket::Candidates::const_iterator iter = candidates.begin(); - iter != candidates.end(); ++iter) { - candidates_.push_back(*iter); - } - } - std::vector candidates_; -}; - -class WebRtcSessionTest : public testing::Test { - protected: - virtual void SetUp() { - signaling_thread_ = talk_base::Thread::Current(); - worker_thread_ = talk_base::Thread::Current(); - channel_manager_.reset(new cricket::ChannelManager(worker_thread_)); - port_allocator_.reset( - new cricket::FakePortAllocator(worker_thread_, NULL)); - desc_factory_.reset( - new cricket::MediaSessionDescriptionFactory(channel_manager_.get())); - } - - bool InitializeSession() { - return session_.get()->Initialize(); - } - - bool CheckChannels() { - return (session_->voice_channel() != NULL && - session_->video_channel() != NULL); - } - - bool CheckTransportChannels() { - EXPECT_TRUE(session_->GetChannel(cricket::CN_AUDIO, "rtp") != NULL); - EXPECT_TRUE(session_->GetChannel(cricket::CN_AUDIO, "rtcp") != NULL); - EXPECT_TRUE(session_->GetChannel(cricket::CN_VIDEO, "video_rtp") != NULL); - EXPECT_TRUE(session_->GetChannel(cricket::CN_VIDEO, "video_rtcp") != NULL); - } - - void Init() { - ASSERT_TRUE(channel_manager_.get() != NULL); - ASSERT_TRUE(session_.get() == NULL); - EXPECT_TRUE(channel_manager_.get()->Init()); - session_.reset(new webrtc::WebRtcSession( - channel_manager_.get(), worker_thread_, signaling_thread_, - port_allocator_.get())); - session_->RegisterObserver(&observer_); - desc_provider_ = session_.get(); - EXPECT_TRUE(InitializeSession()); - } - - - void CreateOffer(uint32 ssrc) { - cricket::MediaSessionOptions options; - // TODO(mallinath) - Adding test cases for session. - local_desc_ = desc_provider_->ProvideOffer(options); - ASSERT_TRUE(local_desc_ != NULL); - } - void CreateAnswer(uint32 ssrc) { - cricket::MediaSessionOptions options; - // TODO(mallinath) - Adding test cases for session. - remote_desc_ = desc_factory_->CreateAnswer(local_desc_, options); - ASSERT_TRUE(remote_desc_ != NULL); - } - void SetRemoteContents() { - desc_provider_->SetRemoteSessionDescription( - remote_desc_, observer_.candidates_); - } - void NegotiationDone() { - desc_provider_->NegotiationDone(); - } - - const cricket::SessionDescription* local_desc_; - const cricket::SessionDescription* remote_desc_; - talk_base::Thread* signaling_thread_; - talk_base::Thread* worker_thread_; - talk_base::scoped_ptr port_allocator_; - talk_base::scoped_ptr session_; - webrtc::SessionDescriptionProvider* desc_provider_; - talk_base::scoped_ptr channel_manager_; - talk_base::scoped_ptr desc_factory_; - MockWebRtcSessionObserver observer_; -}; - -TEST_F(WebRtcSessionTest, TestInitialize) { - WebRtcSessionTest::Init(); - EXPECT_TRUE(CheckChannels()); - CheckTransportChannels(); - talk_base::Thread::Current()->ProcessMessages(1000); - EXPECT_EQ(4u, observer_.candidates_.size()); -} - -// TODO(mallinath) - Adding test cases for session. -TEST_F(WebRtcSessionTest, DISABLE_TestOfferAnswer) { - WebRtcSessionTest::Init(); - EXPECT_TRUE(CheckChannels()); - CheckTransportChannels(); - talk_base::Thread::Current()->ProcessMessages(1); -} - diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsessionobserver.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsessionobserver.h deleted file mode 100644 index c1653c93e..000000000 --- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsessionobserver.h +++ /dev/null @@ -1,47 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#ifndef TALK_APP_WEBRTC_WEBRTCSESSIONOBSERVER_H_ -#define TALK_APP_WEBRTC_WEBRTCSESSIONOBSERVER_H_ - -#include - -#include "talk/p2p/base/candidate.h" - -namespace webrtc { - -class WebRtcSessionObserver { - public: - virtual void OnCandidatesReady( - const std::vector& candiddates) = 0; - protected: - virtual ~WebRtcSessionObserver() {} -}; - -} // namespace webrtc - -#endif // TALK_APP_WEBRTC_WEBRTCSESSIONOBSERVER_H_ diff --git a/third_party_mods/libjingle/source/talk/base/refcount.h b/third_party_mods/libjingle/source/talk/base/refcount.h deleted file mode 100644 index be4ac8a94..000000000 --- a/third_party_mods/libjingle/source/talk/base/refcount.h +++ /dev/null @@ -1,90 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#ifndef TALK_BASE_REF_COUNT_H_ -#define TALK_BASE_REF_COUNT_H_ - -#include - -#include "talk/base/criticalsection.h" - -namespace talk_base { - -// Reference count interface. -class RefCountInterface { - public: - virtual int AddRef() = 0; - virtual int Release() = 0; -}; - -template -class RefCountedObject : public T { - public: - RefCountedObject() : ref_count_(0) { - } - - template - explicit RefCountedObject(P p) : ref_count_(0), T(p) { - } - - template - RefCountedObject(P1 p1, P2 p2) : ref_count_(0), T(p1, p2) { - } - - template - RefCountedObject(P1 p1, P2 p2, P3 p3) : ref_count_(0), T(p1, p2, p3) { - } - - template - RefCountedObject(P1 p1, P2 p2, P3 p3, P4 p4) - : ref_count_(0), T(p1, p2, p3, p4) { - } - - template - RefCountedObject(P1 p1, P2 p2, P3 p3, P4 p4, P5 p5) - : ref_count_(0), T(p1, p2, p3, p4, p5) { - } - - virtual int AddRef() { - return talk_base::AtomicOps::Increment(&ref_count_); - } - - virtual int Release() { - int count = talk_base::AtomicOps::Decrement(&ref_count_); - if (!count) { - delete this; - } - return count; - } - - protected: - int ref_count_; -}; - -} // namespace talk_base - -#endif // TALK_BASE_REF_COUNT_H_ diff --git a/third_party_mods/libjingle/source/talk/base/scoped_refptr.h b/third_party_mods/libjingle/source/talk/base/scoped_refptr.h deleted file mode 100644 index 7709962c9..000000000 --- a/third_party_mods/libjingle/source/talk/base/scoped_refptr.h +++ /dev/null @@ -1,132 +0,0 @@ -#ifndef TALK_BASE_SCOPED_REFPTR_H_ -#define TALK_BASE_SCOPED_REFPTR_H_ -// Originally these classes are copied from Chromium. - -// -// A smart pointer class for reference counted objects. Use this class instead -// of calling AddRef and Release manually on a reference counted object to -// avoid common memory leaks caused by forgetting to Release an object -// reference. Sample usage: -// -// class MyFoo : public RefCounted { -// ... -// }; -// -// void some_function() { -// scoped_refptr foo = new MyFoo(); -// foo->Method(param); -// // |foo| is released when this function returns -// } -// -// void some_other_function() { -// scoped_refptr foo = new MyFoo(); -// ... -// foo = NULL; // explicitly releases |foo| -// ... -// if (foo) -// foo->Method(param); -// } -// -// The above examples show how scoped_refptr acts like a pointer to T. -// Given two scoped_refptr classes, it is also possible to exchange -// references between the two objects, like so: -// -// { -// scoped_refptr a = new MyFoo(); -// scoped_refptr b; -// -// b.swap(a); -// // now, |b| references the MyFoo object, and |a| references NULL. -// } -// -// To make both |a| and |b| in the above example reference the same MyFoo -// object, simply use the assignment operator: -// -// { -// scoped_refptr a = new MyFoo(); -// scoped_refptr b; -// -// b = a; -// // now, |a| and |b| each own a reference to the same MyFoo object. -// } -// -namespace talk_base { - -template -class scoped_refptr { - public: - scoped_refptr() : ptr_(NULL) { - } - - scoped_refptr(T* p) : ptr_(p) { - if (ptr_) - ptr_->AddRef(); - } - - scoped_refptr(const scoped_refptr& r) : ptr_(r.ptr_) { - if (ptr_) - ptr_->AddRef(); - } - - template - scoped_refptr(const scoped_refptr& r) : ptr_(r.get()) { - if (ptr_) - ptr_->AddRef(); - } - - ~scoped_refptr() { - if (ptr_) - ptr_->Release(); - } - - T* get() const { return ptr_; } - operator T*() const { return ptr_; } - T* operator->() const { return ptr_; } - - // Release a pointer. - // The return value is the current pointer held by this object. - // If this object holds a NULL pointer, the return value is NULL. - // After this operation, this object will hold a NULL pointer, - // and will not own the object any more. - T* release() { - T* retVal = ptr_; - ptr_ = NULL; - return retVal; - } - - scoped_refptr& operator=(T* p) { - // AddRef first so that self assignment should work - if (p) - p->AddRef(); - if (ptr_ ) - ptr_ ->Release(); - ptr_ = p; - return *this; - } - - scoped_refptr& operator=(const scoped_refptr& r) { - return *this = r.ptr_; - } - - template - scoped_refptr& operator=(const scoped_refptr& r) { - return *this = r.get(); - } - - void swap(T** pp) { - T* p = ptr_; - ptr_ = *pp; - *pp = p; - } - - void swap(scoped_refptr& r) { - swap(&r.ptr_); - } - - protected: - T* ptr_; -}; - -} // namespace talk_base - -#endif // TALK_BASE_SCOPED_REFPTR_H_ diff --git a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/conductor.cc b/third_party_mods/libjingle/source/talk/examples/peerconnection_client/conductor.cc deleted file mode 100644 index 692b1fa85..000000000 --- a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/conductor.cc +++ /dev/null @@ -1,378 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "talk/examples/peerconnection_client/conductor.h" - -#include - -#include "modules/video_capture/main/interface/video_capture_factory.h" -#include "talk/app/webrtc_dev/peerconnection.h" -#include "talk/examples/peerconnection_client/defaults.h" -#include "talk/base/common.h" -#include "talk/base/logging.h" -#include "talk/p2p/client/basicportallocator.h" -#include "talk/session/phone/videorendererfactory.h" - -Conductor::Conductor(PeerConnectionClient* client, MainWindow* main_wnd) - : peer_id_(-1), - client_(client), - main_wnd_(main_wnd) { - client_->RegisterObserver(this); - main_wnd->RegisterObserver(this); -} - -Conductor::~Conductor() { - ASSERT(peer_connection_.get() == NULL); -} - -bool Conductor::connection_active() const { - return peer_connection_.get() != NULL; -} - -void Conductor::Close() { - client_->SignOut(); - DeletePeerConnection(); -} - -bool Conductor::InitializePeerConnection() { - ASSERT(peer_connection_factory_.get() == NULL); - ASSERT(peer_connection_.get() == NULL); - - peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(); - - if (!peer_connection_factory_.get()) { - main_wnd_->MessageBox("Error", - "Failed to initialize PeerConnectionFactory", true); - DeletePeerConnection(); - return false; - } - - peer_connection_ = peer_connection_factory_->CreatePeerConnection( - GetPeerConnectionString(), this); - - if (!peer_connection_.get()) { - main_wnd_->MessageBox("Error", - "CreatePeerConnection failed", true); - DeletePeerConnection(); - } - return peer_connection_.get() != NULL; -} - -void Conductor::DeletePeerConnection() { - peer_connection_ = NULL; - active_streams_.clear(); - peer_connection_factory_ = NULL; - peer_id_ = -1; -} - -void Conductor::EnsureStreamingUI() { - ASSERT(peer_connection_.get() != NULL); - if (main_wnd_->IsWindow()) { - if (main_wnd_->current_ui() != MainWindow::STREAMING) - main_wnd_->SwitchToStreamingUI(); - } -} - -// -// PeerConnectionObserver implementation. -// - -void Conductor::OnError() { - LOG(LS_ERROR) << __FUNCTION__; - main_wnd_->QueueUIThreadCallback(PEER_CONNECTION_ERROR, NULL); -} - -void Conductor::OnSignalingMessage(const std::string& msg) { - LOG(INFO) << __FUNCTION__; - - std::string* msg_copy = new std::string(msg); - main_wnd_->QueueUIThreadCallback(SEND_MESSAGE_TO_PEER, msg_copy); -} - -// Called when a remote stream is added -void Conductor::OnAddStream(webrtc::MediaStreamInterface* stream) { - LOG(INFO) << __FUNCTION__ << " " << stream->label(); - - stream->AddRef(); - main_wnd_->QueueUIThreadCallback(NEW_STREAM_ADDED, - stream); -} - -void Conductor::OnRemoveStream(webrtc::MediaStreamInterface* stream) { - LOG(INFO) << __FUNCTION__ << " " << stream->label(); - stream->AddRef(); - main_wnd_->QueueUIThreadCallback(STREAM_REMOVED, - stream); -} - -// -// PeerConnectionClientObserver implementation. -// - -void Conductor::OnSignedIn() { - LOG(INFO) << __FUNCTION__; - main_wnd_->SwitchToPeerList(client_->peers()); -} - -void Conductor::OnDisconnected() { - LOG(INFO) << __FUNCTION__; - - DeletePeerConnection(); - - if (main_wnd_->IsWindow()) - main_wnd_->SwitchToConnectUI(); -} - -void Conductor::OnPeerConnected(int id, const std::string& name) { - LOG(INFO) << __FUNCTION__; - // Refresh the list if we're showing it. - if (main_wnd_->current_ui() == MainWindow::LIST_PEERS) - main_wnd_->SwitchToPeerList(client_->peers()); -} - -void Conductor::OnPeerDisconnected(int id) { - LOG(INFO) << __FUNCTION__; - if (id == peer_id_) { - LOG(INFO) << "Our peer disconnected"; - main_wnd_->QueueUIThreadCallback(PEER_CONNECTION_CLOSED, NULL); - } else { - // Refresh the list if we're showing it. - if (main_wnd_->current_ui() == MainWindow::LIST_PEERS) - main_wnd_->SwitchToPeerList(client_->peers()); - } -} - -void Conductor::OnMessageFromPeer(int peer_id, const std::string& message) { - ASSERT(peer_id_ == peer_id || peer_id_ == -1); - ASSERT(!message.empty()); - - if (!peer_connection_.get()) { - ASSERT(peer_id_ == -1); - peer_id_ = peer_id; - - // Got an offer. Give it to the PeerConnection instance. - // Once processed, we will get a callback to OnSignalingMessage with - // our 'answer' which we'll send to the peer. - LOG(INFO) << "Got an offer from our peer: " << peer_id; - if (!InitializePeerConnection()) { - LOG(LS_ERROR) << "Failed to initialize our PeerConnection instance"; - client_->SignOut(); - return; - } - } else if (peer_id != peer_id_) { - ASSERT(peer_id_ != -1); - LOG(WARNING) << "Received an offer from a peer while already in a " - "conversation with a different peer."; - return; - } - - peer_connection_->ProcessSignalingMessage(message); -} - -void Conductor::OnMessageSent(int err) { - // Process the next pending message if any. - main_wnd_->QueueUIThreadCallback(SEND_MESSAGE_TO_PEER, NULL); -} - -// -// MainWndCallback implementation. -// - -bool Conductor::StartLogin(const std::string& server, int port) { - if (client_->is_connected()) - return false; - - if (!client_->Connect(server, port, GetPeerName())) { - main_wnd_->MessageBox("Error", ("Failed to connect to " + server).c_str(), - true); - return false; - } - - return true; -} - -void Conductor::DisconnectFromServer() { - if (client_->is_connected()) - client_->SignOut(); -} - -void Conductor::ConnectToPeer(int peer_id) { - ASSERT(peer_id_ == -1); - ASSERT(peer_id != -1); - - if (peer_connection_.get()) { - main_wnd_->MessageBox("Error", - "We only support connecting to one peer at a time", true); - return; - } - - if (InitializePeerConnection()) { - peer_id_ = peer_id; - AddStreams(); - } else { - main_wnd_->MessageBox("Error", "Failed to initialize PeerConnection", true); - } -} - -talk_base::scoped_refptr -Conductor::OpenVideoCaptureDevice() { - webrtc::VideoCaptureModule::DeviceInfo* device_info( - webrtc::VideoCaptureFactory::CreateDeviceInfo(0)); - talk_base::scoped_refptr video_device; - - const size_t kMaxDeviceNameLength = 128; - const size_t kMaxUniqueIdLength = 256; - uint8 device_name[kMaxDeviceNameLength]; - uint8 unique_id[kMaxUniqueIdLength]; - - const size_t device_count = device_info->NumberOfDevices(); - for (size_t i = 0; i < device_count; ++i) { - // Get the name of the video capture device. - device_info->GetDeviceName(i, device_name, kMaxDeviceNameLength, unique_id, - kMaxUniqueIdLength); - // Try to open this device. - video_device = - webrtc::VideoCaptureFactory::Create(0, unique_id); - if (video_device.get()) - break; - } - delete device_info; - return video_device; -} - -void Conductor::AddStreams() { - if (active_streams_.find(kStreamLabel) != active_streams_.end()) - return; // Already added. - - talk_base::scoped_refptr audio_track( - peer_connection_factory_->CreateLocalAudioTrack(kAudioLabel, NULL)); - - talk_base::scoped_refptr video_track( - peer_connection_factory_->CreateLocalVideoTrack( - kVideoLabel, OpenVideoCaptureDevice())); - - video_track->SetRenderer(main_wnd_->local_renderer()); - - talk_base::scoped_refptr stream = - peer_connection_factory_->CreateLocalMediaStream(kStreamLabel); - - stream->AddTrack(audio_track); - stream->AddTrack(video_track); - peer_connection_->AddStream(stream); - peer_connection_->CommitStreamChanges(); - typedef std::pair > - MediaStreamPair; - active_streams_.insert(MediaStreamPair(stream->label(), stream)); - main_wnd_->SwitchToStreamingUI(); -} - -void Conductor::DisconnectFromCurrentPeer() { - LOG(INFO) << __FUNCTION__; - if (peer_connection_.get()) { - client_->SendHangUp(peer_id_); - DeletePeerConnection(); - } - - if (main_wnd_->IsWindow()) - main_wnd_->SwitchToPeerList(client_->peers()); -} - -void Conductor::UIThreadCallback(int msg_id, void* data) { - switch (msg_id) { - case PEER_CONNECTION_CLOSED: - LOG(INFO) << "PEER_CONNECTION_CLOSED"; - DeletePeerConnection(); - - ASSERT(active_streams_.empty()); - - if (main_wnd_->IsWindow()) { - if (client_->is_connected()) { - main_wnd_->SwitchToPeerList(client_->peers()); - } else { - main_wnd_->SwitchToConnectUI(); - } - } else { - DisconnectFromServer(); - } - break; - - case SEND_MESSAGE_TO_PEER: { - LOG(INFO) << "SEND_MESSAGE_TO_PEER"; - std::string* msg = reinterpret_cast(data); - if (msg) { - // For convenience, we always run the message through the queue. - // This way we can be sure that messages are sent to the server - // in the same order they were signaled without much hassle. - pending_messages_.push_back(msg); - } - - if (!pending_messages_.empty() && !client_->IsSendingMessage()) { - msg = pending_messages_.front(); - pending_messages_.pop_front(); - - if (!client_->SendToPeer(peer_id_, *msg) && peer_id_ != -1) { - LOG(LS_ERROR) << "SendToPeer failed"; - DisconnectFromServer(); - } - delete msg; - } - - if (!peer_connection_.get()) - peer_id_ = -1; - - break; - } - - case PEER_CONNECTION_ADDSTREAMS: - AddStreams(); - break; - - case PEER_CONNECTION_ERROR: - main_wnd_->MessageBox("Error", "an unknown error occurred", true); - break; - - case NEW_STREAM_ADDED: { - webrtc::MediaStreamInterface* stream = - reinterpret_cast( - data); - talk_base::scoped_refptr tracks = - stream->video_tracks(); - for (size_t i = 0; i < tracks->count(); ++i) { - webrtc::VideoTrackInterface* track = tracks->at(i); - LOG(INFO) << "Setting video renderer for track: " << track->label(); - track->SetRenderer(main_wnd_->remote_renderer()); - } - // If we haven't shared any streams with this peer (we're the receiver) - // then do so now. - if (active_streams_.empty()) - AddStreams(); - stream->Release(); - break; - } - - case STREAM_REMOVED: { - webrtc::MediaStreamInterface* stream = - reinterpret_cast( - data); - active_streams_.erase(stream->label()); - stream->Release(); - if (active_streams_.empty()) { - LOG(INFO) << "All streams have been closed."; - main_wnd_->QueueUIThreadCallback(PEER_CONNECTION_CLOSED, NULL); - } - break; - } - - default: - ASSERT(false); - break; - } -} diff --git a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/conductor.h b/third_party_mods/libjingle/source/talk/examples/peerconnection_client/conductor.h deleted file mode 100644 index c54ebfa86..000000000 --- a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/conductor.h +++ /dev/null @@ -1,116 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef PEERCONNECTION_SAMPLES_CLIENT_CONDUCTOR_H_ -#define PEERCONNECTION_SAMPLES_CLIENT_CONDUCTOR_H_ -#pragma once - -#include -#include -#include -#include - -#include "talk/examples/peerconnection_client/main_wnd.h" -#include "talk/examples/peerconnection_client/peer_connection_client.h" -#include "talk/app/webrtc_dev/mediastream.h" -#include "talk/app/webrtc_dev/peerconnection.h" -#include "talk/base/scoped_ptr.h" - -namespace webrtc { -class VideoCaptureModule; -} // namespace webrtc - -namespace cricket { -class VideoRenderer; -} // namespace cricket - -class Conductor - : public webrtc::PeerConnectionObserver, - public PeerConnectionClientObserver, - public MainWndCallback { - public: - enum CallbackID { - MEDIA_CHANNELS_INITIALIZED = 1, - PEER_CONNECTION_CLOSED, - SEND_MESSAGE_TO_PEER, - PEER_CONNECTION_ADDSTREAMS, - PEER_CONNECTION_ERROR, - NEW_STREAM_ADDED, - STREAM_REMOVED, - }; - - Conductor(PeerConnectionClient* client, MainWindow* main_wnd); - ~Conductor(); - - bool connection_active() const; - - virtual void Close(); - - protected: - bool InitializePeerConnection(); - void DeletePeerConnection(); - void EnsureStreamingUI(); - void AddStreams(); - talk_base::scoped_refptr OpenVideoCaptureDevice(); - - // - // PeerConnectionObserver implementation. - // - virtual void OnError(); - virtual void OnMessage(const std::string& msg) {} - virtual void OnSignalingMessage(const std::string& msg); - virtual void OnStateChange(Readiness state) {} - virtual void OnAddStream(webrtc::MediaStreamInterface* stream); - virtual void OnRemoveStream(webrtc::MediaStreamInterface* stream); - - - // - // PeerConnectionClientObserver implementation. - // - - virtual void OnSignedIn(); - - virtual void OnDisconnected(); - - virtual void OnPeerConnected(int id, const std::string& name); - - virtual void OnPeerDisconnected(int id); - - virtual void OnMessageFromPeer(int peer_id, const std::string& message); - - virtual void OnMessageSent(int err); - - // - // MainWndCallback implementation. - // - - virtual bool StartLogin(const std::string& server, int port); - - virtual void DisconnectFromServer(); - - virtual void ConnectToPeer(int peer_id); - - virtual void DisconnectFromCurrentPeer(); - - virtual void UIThreadCallback(int msg_id, void* data); - - protected: - int peer_id_; - talk_base::scoped_refptr peer_connection_; - talk_base::scoped_refptr - peer_connection_factory_; - PeerConnectionClient* client_; - MainWindow* main_wnd_; - std::deque pending_messages_; - std::map > - active_streams_; -}; - -#endif // PEERCONNECTION_SAMPLES_CLIENT_CONDUCTOR_H_ diff --git a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/defaults.cc b/third_party_mods/libjingle/source/talk/examples/peerconnection_client/defaults.cc deleted file mode 100644 index 38b6d9d4a..000000000 --- a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/defaults.cc +++ /dev/null @@ -1,58 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "talk/examples/peerconnection_client/defaults.h" - -#include -#include - -#ifdef WIN32 -#include -#else -#include -#endif - -#include "talk/base/common.h" - -const char kAudioLabel[] = "audio_label"; -const char kVideoLabel[] = "video_label"; -const char kStreamLabel[] = "stream_label"; -const uint16 kDefaultServerPort = 8888; - -std::string GetEnvVarOrDefault(const char* env_var_name, - const char* default_value) { - std::string value; - const char* env_var = getenv(env_var_name); - if (env_var) - value = env_var; - - if (value.empty()) - value = default_value; - - return value; -} - -std::string GetPeerConnectionString() { - return GetEnvVarOrDefault("WEBRTC_CONNECT", "STUN stun.l.google.com:19302"); -} - -std::string GetDefaultServerName() { - return GetEnvVarOrDefault("WEBRTC_SERVER", "localhost"); -} - -std::string GetPeerName() { - char computer_name[256]; - if (gethostname(computer_name, ARRAY_SIZE(computer_name)) != 0) - strcpy(computer_name, "host"); - std::string ret(GetEnvVarOrDefault("USERNAME", "user")); - ret += '@'; - ret += computer_name; - return ret; -} diff --git a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/defaults.h b/third_party_mods/libjingle/source/talk/examples/peerconnection_client/defaults.h deleted file mode 100644 index bd05980c6..000000000 --- a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/defaults.h +++ /dev/null @@ -1,30 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef PEERCONNECTION_SAMPLES_CLIENT_DEFAULTS_H_ -#define PEERCONNECTION_SAMPLES_CLIENT_DEFAULTS_H_ -#pragma once - -#include - -#include "talk/base/basictypes.h" - -extern const char kAudioLabel[]; -extern const char kVideoLabel[]; -extern const char kStreamLabel[]; -extern const uint16 kDefaultServerPort; - -std::string GetEnvVarOrDefault(const char* env_var_name, - const char* default_value); -std::string GetPeerConnectionString(); -std::string GetDefaultServerName(); -std::string GetPeerName(); - -#endif // PEERCONNECTION_SAMPLES_CLIENT_DEFAULTS_H_ diff --git a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/linux/main.cc b/third_party_mods/libjingle/source/talk/examples/peerconnection_client/linux/main.cc deleted file mode 100644 index 87590fbe5..000000000 --- a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/linux/main.cc +++ /dev/null @@ -1,85 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include - -#include "talk/examples/peerconnection_client/conductor.h" -#include "talk/examples/peerconnection_client/linux/main_wnd.h" -#include "talk/examples/peerconnection_client/peer_connection_client.h" - -#include "talk/base/thread.h" - -class CustomSocketServer : public talk_base::PhysicalSocketServer { - public: - CustomSocketServer(talk_base::Thread* thread, GtkMainWnd* wnd) - : thread_(thread), wnd_(wnd), conductor_(NULL), client_(NULL) {} - virtual ~CustomSocketServer() {} - - void set_client(PeerConnectionClient* client) { client_ = client; } - void set_conductor(Conductor* conductor) { conductor_ = conductor; } - - // Override so that we can also pump the GTK message loop. - virtual bool Wait(int cms, bool process_io) { - // Pump GTK events. - // TODO(tommi): We really should move either the socket server or UI to a - // different thread. Alternatively we could look at merging the two loops - // by implementing a dispatcher for the socket server and/or use - // g_main_context_set_poll_func. - while (gtk_events_pending()) - gtk_main_iteration(); - - if (!wnd_->IsWindow() && !conductor_->connection_active() && - client_ != NULL && !client_->is_connected()) { - thread_->Quit(); - } - return talk_base::PhysicalSocketServer::Wait(0/*cms == -1 ? 1 : cms*/, - process_io); - } - - protected: - talk_base::Thread* thread_; - GtkMainWnd* wnd_; - Conductor* conductor_; - PeerConnectionClient* client_; -}; - -int main(int argc, char* argv[]) { - gtk_init(&argc, &argv); - g_type_init(); - g_thread_init(NULL); - - GtkMainWnd wnd; - wnd.Create(); - - talk_base::AutoThread auto_thread; - talk_base::Thread* thread = talk_base::Thread::Current(); - CustomSocketServer socket_server(thread, &wnd); - thread->set_socketserver(&socket_server); - - // Must be constructed after we set the socketserver. - PeerConnectionClient client; - Conductor conductor(&client, &wnd); - socket_server.set_client(&client); - socket_server.set_conductor(&conductor); - - thread->Run(); - - // gtk_main(); - wnd.Destroy(); - - thread->set_socketserver(NULL); - // TODO(tommi): Run the Gtk main loop to tear down the connection. - //while (gtk_events_pending()) { - // gtk_main_iteration(); - //} - - return 0; -} - diff --git a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/linux/main_wnd.cc b/third_party_mods/libjingle/source/talk/examples/peerconnection_client/linux/main_wnd.cc deleted file mode 100644 index 61e6e842b..000000000 --- a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/linux/main_wnd.cc +++ /dev/null @@ -1,464 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - - -#include "talk/examples/peerconnection_client/linux/main_wnd.h" - -#include -#include -#include - -#include "talk/examples/peerconnection_client/defaults.h" -#include "talk/base/common.h" -#include "talk/base/logging.h" -#include "talk/base/stringutils.h" - -using talk_base::sprintfn; - -namespace { - -// -// Simple static functions that simply forward the callback to the -// GtkMainWnd instance. -// - -gboolean OnDestroyedCallback(GtkWidget* widget, GdkEvent* event, - gpointer data) { - reinterpret_cast(data)->OnDestroyed(widget, event); - return FALSE; -} - -void OnClickedCallback(GtkWidget* widget, gpointer data) { - reinterpret_cast(data)->OnClicked(widget); -} - -gboolean OnKeyPressCallback(GtkWidget* widget, GdkEventKey* key, - gpointer data) { - reinterpret_cast(data)->OnKeyPress(widget, key); - return false; -} - -void OnRowActivatedCallback(GtkTreeView* tree_view, GtkTreePath* path, - GtkTreeViewColumn* column, gpointer data) { - reinterpret_cast(data)->OnRowActivated(tree_view, path, column); -} - -// Creates a tree view, that we use to display the list of peers. -void InitializeList(GtkWidget* list) { - GtkCellRenderer* renderer = gtk_cell_renderer_text_new(); - GtkTreeViewColumn* column = gtk_tree_view_column_new_with_attributes( - "List Items", renderer, "text", 0, NULL); - gtk_tree_view_append_column(GTK_TREE_VIEW(list), column); - GtkListStore* store = gtk_list_store_new(2, G_TYPE_STRING, G_TYPE_INT); - gtk_tree_view_set_model(GTK_TREE_VIEW(list), GTK_TREE_MODEL(store)); - g_object_unref(store); -} - -// Adds an entry to a tree view. -void AddToList(GtkWidget* list, const gchar* str, int value) { - GtkListStore* store = GTK_LIST_STORE( - gtk_tree_view_get_model(GTK_TREE_VIEW(list))); - - GtkTreeIter iter; - gtk_list_store_append(store, &iter); - gtk_list_store_set(store, &iter, 0, str, 1, value, -1); -} - -struct UIThreadCallbackData { - explicit UIThreadCallbackData(MainWndCallback* cb, int id, void* d) - : callback(cb), msg_id(id), data(d) {} - MainWndCallback* callback; - int msg_id; - void* data; -}; - -gboolean HandleUIThreadCallback(gpointer data) { - UIThreadCallbackData* cb_data = reinterpret_cast(data); - cb_data->callback->UIThreadCallback(cb_data->msg_id, cb_data->data); - delete cb_data; - return false; -} - -gboolean Redraw(gpointer data) { - GtkMainWnd* wnd = reinterpret_cast(data); - wnd->OnRedraw(); - return false; -} -} // end anonymous - -// -// GtkMainWnd implementation. -// - -GtkMainWnd::GtkMainWnd() - : window_(NULL), draw_area_(NULL), vbox_(NULL), server_edit_(NULL), - port_edit_(NULL), peer_list_(NULL), callback_(NULL), - server_("localhost") { - char buffer[10]; - sprintfn(buffer, sizeof(buffer), "%i", kDefaultServerPort); - port_ = buffer; -} - -GtkMainWnd::~GtkMainWnd() { - ASSERT(!IsWindow()); -} - -void GtkMainWnd::RegisterObserver(MainWndCallback* callback) { - callback_ = callback; -} - -bool GtkMainWnd::IsWindow() { - return window_ != NULL && GTK_IS_WINDOW(window_); -} - -void GtkMainWnd::MessageBox(const char* caption, const char* text, - bool is_error) { - GtkWidget* dialog = gtk_message_dialog_new(GTK_WINDOW(window_), - GTK_DIALOG_DESTROY_WITH_PARENT, - is_error ? GTK_MESSAGE_ERROR : GTK_MESSAGE_INFO, - GTK_BUTTONS_CLOSE, "%s", text); - gtk_window_set_title(GTK_WINDOW(dialog), caption); - gtk_dialog_run(GTK_DIALOG(dialog)); - gtk_widget_destroy(dialog); -} - -MainWindow::UI GtkMainWnd::current_ui() { - if (vbox_) - return CONNECT_TO_SERVER; - - if (peer_list_) - return LIST_PEERS; - - return STREAMING; -} - -webrtc::VideoRendererWrapperInterface* GtkMainWnd::local_renderer() { - if (!local_renderer_wrapper_.get()) - local_renderer_wrapper_ = - webrtc::CreateVideoRenderer(new VideoRenderer(this)); - return local_renderer_wrapper_.get(); -} - -webrtc::VideoRendererWrapperInterface* GtkMainWnd::remote_renderer() { - if (!remote_renderer_wrapper_.get()) - remote_renderer_wrapper_ = - webrtc::CreateVideoRenderer(new VideoRenderer(this)); - return remote_renderer_wrapper_.get(); -} - -void GtkMainWnd::QueueUIThreadCallback(int msg_id, void* data) { - g_idle_add(HandleUIThreadCallback, - new UIThreadCallbackData(callback_, msg_id, data)); -} - -bool GtkMainWnd::Create() { - ASSERT(window_ == NULL); - - window_ = gtk_window_new(GTK_WINDOW_TOPLEVEL); - if (window_) { - gtk_window_set_position(GTK_WINDOW(window_), GTK_WIN_POS_CENTER); - gtk_window_set_default_size(GTK_WINDOW(window_), 640, 480); - gtk_window_set_title(GTK_WINDOW(window_), "PeerConnection client"); - g_signal_connect(G_OBJECT(window_), "delete-event", - G_CALLBACK(&OnDestroyedCallback), this); - g_signal_connect(window_, "key-press-event", G_CALLBACK(OnKeyPressCallback), - this); - - SwitchToConnectUI(); - } - - return window_ != NULL; -} - -bool GtkMainWnd::Destroy() { - if (!IsWindow()) - return false; - - gtk_widget_destroy(window_); - window_ = NULL; - - return true; -} - -void GtkMainWnd::SwitchToConnectUI() { - LOG(INFO) << __FUNCTION__; - - ASSERT(IsWindow()); - ASSERT(vbox_ == NULL); - - gtk_container_set_border_width(GTK_CONTAINER(window_), 10); - - if (peer_list_) { - gtk_widget_destroy(peer_list_); - peer_list_ = NULL; - } - - vbox_ = gtk_vbox_new(FALSE, 5); - GtkWidget* valign = gtk_alignment_new(0, 1, 0, 0); - gtk_container_add(GTK_CONTAINER(vbox_), valign); - gtk_container_add(GTK_CONTAINER(window_), vbox_); - - GtkWidget* hbox = gtk_hbox_new(FALSE, 5); - - GtkWidget* label = gtk_label_new("Server"); - gtk_container_add(GTK_CONTAINER(hbox), label); - - server_edit_ = gtk_entry_new(); - gtk_entry_set_text(GTK_ENTRY(server_edit_), server_.c_str()); - gtk_widget_set_size_request(server_edit_, 400, 30); - gtk_container_add(GTK_CONTAINER(hbox), server_edit_); - - port_edit_ = gtk_entry_new(); - gtk_entry_set_text(GTK_ENTRY(port_edit_), port_.c_str()); - gtk_widget_set_size_request(port_edit_, 70, 30); - gtk_container_add(GTK_CONTAINER(hbox), port_edit_); - - GtkWidget* button = gtk_button_new_with_label("Connect"); - gtk_widget_set_size_request(button, 70, 30); - g_signal_connect(button, "clicked", G_CALLBACK(OnClickedCallback), this); - gtk_container_add(GTK_CONTAINER(hbox), button); - - GtkWidget* halign = gtk_alignment_new(1, 0, 0, 0); - gtk_container_add(GTK_CONTAINER(halign), hbox); - gtk_box_pack_start(GTK_BOX(vbox_), halign, FALSE, FALSE, 0); - - gtk_widget_show_all(window_); -} - -void GtkMainWnd::SwitchToPeerList(const Peers& peers) { - LOG(INFO) << __FUNCTION__; - - // Clean up buffers from a potential previous session. - local_renderer_wrapper_ = NULL; - remote_renderer_wrapper_ = NULL; - - if (!peer_list_) { - gtk_container_set_border_width(GTK_CONTAINER(window_), 0); - if (vbox_) { - gtk_widget_destroy(vbox_); - vbox_ = NULL; - server_edit_ = NULL; - port_edit_ = NULL; - } else if (draw_area_) { - gtk_widget_destroy(draw_area_); - draw_area_ = NULL; - draw_buffer_.reset(); - } - - peer_list_ = gtk_tree_view_new(); - g_signal_connect(peer_list_, "row-activated", - G_CALLBACK(OnRowActivatedCallback), this); - gtk_tree_view_set_headers_visible(GTK_TREE_VIEW(peer_list_), FALSE); - InitializeList(peer_list_); - gtk_container_add(GTK_CONTAINER(window_), peer_list_); - gtk_widget_show_all(window_); - } else { - GtkListStore* store = - GTK_LIST_STORE(gtk_tree_view_get_model(GTK_TREE_VIEW(peer_list_))); - gtk_list_store_clear(store); - } - - AddToList(peer_list_, "List of currently connected peers:", -1); - for (Peers::const_iterator i = peers.begin(); i != peers.end(); ++i) - AddToList(peer_list_, i->second.c_str(), i->first); -} - -void GtkMainWnd::SwitchToStreamingUI() { - LOG(INFO) << __FUNCTION__; - - ASSERT(draw_area_ == NULL); - - gtk_container_set_border_width(GTK_CONTAINER(window_), 0); - if (peer_list_) { - gtk_widget_destroy(peer_list_); - peer_list_ = NULL; - } - - draw_area_ = gtk_drawing_area_new(); - gtk_container_add(GTK_CONTAINER(window_), draw_area_); - - gtk_widget_show_all(window_); -} - -void GtkMainWnd::OnDestroyed(GtkWidget* widget, GdkEvent* event) { - callback_->Close(); - window_ = NULL; - draw_area_ = NULL; - vbox_ = NULL; - server_edit_ = NULL; - port_edit_ = NULL; - peer_list_ = NULL; -} - -void GtkMainWnd::OnClicked(GtkWidget* widget) { - server_ = gtk_entry_get_text(GTK_ENTRY(server_edit_)); - port_ = gtk_entry_get_text(GTK_ENTRY(port_edit_)); - int port = port_.length() ? atoi(port_.c_str()) : 0; - callback_->StartLogin(server_, port); -} - -void GtkMainWnd::OnKeyPress(GtkWidget* widget, GdkEventKey* key) { - if (key->type == GDK_KEY_PRESS) { - switch (key->keyval) { - case GDK_Escape: - if (draw_area_) { - callback_->DisconnectFromCurrentPeer(); - } else if (peer_list_) { - callback_->DisconnectFromServer(); - } - break; - - case GDK_KP_Enter: - case GDK_Return: - if (vbox_) { - OnClicked(NULL); - } else if (peer_list_) { - // OnRowActivated will be called automatically when the user - // presses enter. - } - break; - - default: - break; - } - } -} - -void GtkMainWnd::OnRowActivated(GtkTreeView* tree_view, GtkTreePath* path, - GtkTreeViewColumn* column) { - ASSERT(peer_list_ != NULL); - GtkTreeIter iter; - GtkTreeModel* model; - GtkTreeSelection* selection = - gtk_tree_view_get_selection(GTK_TREE_VIEW(tree_view)); - if (gtk_tree_selection_get_selected(selection, &model, &iter)) { - char* text; - int id = -1; - gtk_tree_model_get(model, &iter, 0, &text, 1, &id, -1); - if (id != -1) - callback_->ConnectToPeer(id); - g_free(text); - } -} - -void GtkMainWnd::OnRedraw() { - gdk_threads_enter(); - - VideoRenderer* remote_renderer = - static_cast(remote_renderer_wrapper_->renderer()); - if (remote_renderer && remote_renderer->image() != NULL && - draw_area_ != NULL) { - int width = remote_renderer->width(); - int height = remote_renderer->height(); - - if (!draw_buffer_.get()) { - draw_buffer_size_ = (width * height * 4) * 4; - draw_buffer_.reset(new uint8[draw_buffer_size_]); - gtk_widget_set_size_request(draw_area_, width * 2, height * 2); - } - - const uint32* image = reinterpret_cast( - remote_renderer->image()); - uint32* scaled = reinterpret_cast(draw_buffer_.get()); - for (int r = 0; r < height; ++r) { - for (int c = 0; c < width; ++c) { - int x = c * 2; - scaled[x] = scaled[x + 1] = image[c]; - } - - uint32* prev_line = scaled; - scaled += width * 2; - memcpy(scaled, prev_line, (width * 2) * 4); - - image += width; - scaled += width * 2; - } - - VideoRenderer* local_renderer = - static_cast(local_renderer_wrapper_->renderer()); - if (local_renderer && local_renderer->image()) { - image = reinterpret_cast(local_renderer->image()); - scaled = reinterpret_cast(draw_buffer_.get()); - // Position the local preview on the right side. - scaled += (width * 2) - (local_renderer->width() / 2); - // right margin... - scaled -= 10; - // ... towards the bottom. - scaled += (height * width * 4) - - ((local_renderer->height() / 2) * - (local_renderer->width() / 2) * 4); - // bottom margin... - scaled -= (width * 2) * 5; - for (int r = 0; r < local_renderer->height(); r += 2) { - for (int c = 0; c < local_renderer->width(); c += 2) { - scaled[c / 2] = image[c + r * local_renderer->width()]; - } - scaled += width * 2; - } - } - - gdk_draw_rgb_32_image(draw_area_->window, - draw_area_->style->fg_gc[GTK_STATE_NORMAL], - 0, - 0, - width * 2, - height * 2, - GDK_RGB_DITHER_MAX, - draw_buffer_.get(), - (width * 2) * 4); - } - - gdk_threads_leave(); -} - -GtkMainWnd::VideoRenderer::VideoRenderer(GtkMainWnd* main_wnd) - : width_(0), height_(0), main_wnd_(main_wnd) { -} - -GtkMainWnd::VideoRenderer::~VideoRenderer() { -} - -bool GtkMainWnd::VideoRenderer::SetSize(int width, int height, int reserved) { - gdk_threads_enter(); - width_ = width; - height_ = height; - image_.reset(new uint8[width * height * 4]); - gdk_threads_leave(); - return true; -} - -bool GtkMainWnd::VideoRenderer::RenderFrame(const cricket::VideoFrame* frame) { - gdk_threads_enter(); - - int size = width_ * height_ * 4; - frame->ConvertToRgbBuffer(cricket::FOURCC_ARGB, - image_.get(), - size, - width_ * 4); - // Convert the B,G,R,A frame to R,G,B,A, which is accepted by GTK. - // The 'A' is just padding for GTK, so we can use it as temp. - uint8* pix = image_.get(); - uint8* end = image_.get() + size; - while (pix < end) { - pix[3] = pix[0]; // Save B to A. - pix[0] = pix[2]; // Set Red. - pix[2] = pix[3]; // Set Blue. - pix[3] = 0xFF; // Fixed Alpha. - pix += 4; - } - - gdk_threads_leave(); - - g_idle_add(Redraw, main_wnd_); - - return true; -} - - diff --git a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/linux/main_wnd.h b/third_party_mods/libjingle/source/talk/examples/peerconnection_client/linux/main_wnd.h deleted file mode 100644 index 1a3fb459f..000000000 --- a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/linux/main_wnd.h +++ /dev/null @@ -1,118 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - - -#ifndef PEERCONNECTION_SAMPLES_CLIENT_LINUX_MAIN_WND_H_ -#define PEERCONNECTION_SAMPLES_CLIENT_LINUX_MAIN_WND_H_ - -#include "talk/examples/peerconnection_client/main_wnd.h" -#include "talk/examples/peerconnection_client/peer_connection_client.h" -#include "talk/base/scoped_refptr.h" - -// Forward declarations. -typedef struct _GtkWidget GtkWidget; -typedef union _GdkEvent GdkEvent; -typedef struct _GdkEventKey GdkEventKey; -typedef struct _GtkTreeView GtkTreeView; -typedef struct _GtkTreePath GtkTreePath; -typedef struct _GtkTreeViewColumn GtkTreeViewColumn; - -// Implements the main UI of the peer connection client. -// This is functionally equivalent to the MainWnd class in the Windows -// implementation. -class GtkMainWnd : public MainWindow { - public: - GtkMainWnd(); - ~GtkMainWnd(); - - virtual void RegisterObserver(MainWndCallback* callback); - virtual bool IsWindow(); - virtual void SwitchToConnectUI(); - virtual void SwitchToPeerList(const Peers& peers); - virtual void SwitchToStreamingUI(); - virtual void MessageBox(const char* caption, const char* text, - bool is_error); - virtual MainWindow::UI current_ui(); - virtual webrtc::VideoRendererWrapperInterface* local_renderer(); - virtual webrtc::VideoRendererWrapperInterface* remote_renderer(); - - virtual void QueueUIThreadCallback(int msg_id, void* data); - - // Creates and shows the main window with the |Connect UI| enabled. - bool Create(); - - // Destroys the window. When the window is destroyed, it ends the - // main message loop. - bool Destroy(); - - // Callback for when the main window is destroyed. - void OnDestroyed(GtkWidget* widget, GdkEvent* event); - - // Callback for when the user clicks the "Connect" button. - void OnClicked(GtkWidget* widget); - - // Callback for keystrokes. Used to capture Esc and Return. - void OnKeyPress(GtkWidget* widget, GdkEventKey* key); - - // Callback when the user double clicks a peer in order to initiate a - // connection. - void OnRowActivated(GtkTreeView* tree_view, GtkTreePath* path, - GtkTreeViewColumn* column); - - void OnRedraw(); - - protected: - class VideoRenderer : public cricket::VideoRenderer { - public: - explicit VideoRenderer(GtkMainWnd* main_wnd); - virtual ~VideoRenderer(); - - virtual bool SetSize(int width, int height, int reserved); - - virtual bool RenderFrame(const cricket::VideoFrame* frame); - - const uint8* image() const { - return image_.get(); - } - - int width() const { - return width_; - } - - int height() const { - return height_; - } - - protected: - talk_base::scoped_array image_; - int width_; - int height_; - GtkMainWnd* main_wnd_; - }; - - protected: - GtkWidget* window_; // Our main window. - GtkWidget* draw_area_; // The drawing surface for rendering video streams. - GtkWidget* vbox_; // Container for the Connect UI. - GtkWidget* server_edit_; - GtkWidget* port_edit_; - GtkWidget* peer_list_; // The list of peers. - MainWndCallback* callback_; - std::string server_; - std::string port_; - talk_base::scoped_refptr - local_renderer_wrapper_; - talk_base::scoped_refptr - remote_renderer_wrapper_; - talk_base::scoped_ptr draw_buffer_; - int draw_buffer_size_; -}; - -#endif // PEERCONNECTION_SAMPLES_CLIENT_LINUX_MAIN_WND_H_ diff --git a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/main.cc b/third_party_mods/libjingle/source/talk/examples/peerconnection_client/main.cc deleted file mode 100644 index 27d81e0bf..000000000 --- a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/main.cc +++ /dev/null @@ -1,58 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include - -#include "talk/examples/peerconnection_client/conductor.h" -#include "talk/examples/peerconnection_client/main_wnd.h" -#include "talk/examples/peerconnection_client/peer_connection_client.h" -#include "system_wrappers/source/trace_impl.h" -#include "talk/base/win32socketinit.h" - - -int PASCAL wWinMain(HINSTANCE instance, HINSTANCE prev_instance, - wchar_t* cmd_line, int cmd_show) { - talk_base::EnsureWinsockInit(); - - webrtc::Trace::CreateTrace(); - webrtc::Trace::SetTraceFile("peerconnection_client.log"); - webrtc::Trace::SetLevelFilter(webrtc::kTraceWarning); - - MainWnd wnd; - if (!wnd.Create()) { - ASSERT(false); - return -1; - } - - PeerConnectionClient client; - Conductor conductor(&client, &wnd); - - // Main loop. - MSG msg; - BOOL gm; - while ((gm = ::GetMessage(&msg, NULL, 0, 0)) && gm != -1) { - if (!wnd.PreTranslateMessage(&msg)) { - ::TranslateMessage(&msg); - ::DispatchMessage(&msg); - } - } - - if (conductor.connection_active() || client.is_connected()) { - while ((conductor.connection_active() || client.is_connected()) && - (gm = ::GetMessage(&msg, NULL, 0, 0)) && gm != -1) { - if (!wnd.PreTranslateMessage(&msg)) { - ::TranslateMessage(&msg); - ::DispatchMessage(&msg); - } - } - } - - return 0; -} diff --git a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/main_wnd.cc b/third_party_mods/libjingle/source/talk/examples/peerconnection_client/main_wnd.cc deleted file mode 100644 index c237331e2..000000000 --- a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/main_wnd.cc +++ /dev/null @@ -1,582 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "talk/examples/peerconnection_client/main_wnd.h" - -#include - -#include "talk/base/common.h" -#include "talk/base/logging.h" - -ATOM MainWnd::wnd_class_ = 0; -const wchar_t MainWnd::kClassName[] = L"WebRTC_MainWnd"; - -namespace { - -const char kConnecting[] = "Connecting... "; -const char kNoVideoStreams[] = "(no video streams either way)"; -const char kNoIncomingStream[] = "(no incoming video)"; - -void CalculateWindowSizeForText(HWND wnd, const wchar_t* text, - size_t* width, size_t* height) { - HDC dc = ::GetDC(wnd); - RECT text_rc = {0}; - ::DrawText(dc, text, -1, &text_rc, DT_CALCRECT | DT_SINGLELINE); - ::ReleaseDC(wnd, dc); - RECT client, window; - ::GetClientRect(wnd, &client); - ::GetWindowRect(wnd, &window); - - *width = text_rc.right - text_rc.left; - *width += (window.right - window.left) - - (client.right - client.left); - *height = text_rc.bottom - text_rc.top; - *height += (window.bottom - window.top) - - (client.bottom - client.top); -} - -HFONT GetDefaultFont() { - static HFONT font = reinterpret_cast(GetStockObject(DEFAULT_GUI_FONT)); - return font; -} - -std::string GetWindowText(HWND wnd) { - char text[MAX_PATH] = {0}; - ::GetWindowTextA(wnd, &text[0], ARRAYSIZE(text)); - return text; -} - -void AddListBoxItem(HWND listbox, const std::string& str, LPARAM item_data) { - LRESULT index = ::SendMessageA(listbox, LB_ADDSTRING, 0, - reinterpret_cast(str.c_str())); - ::SendMessageA(listbox, LB_SETITEMDATA, index, item_data); -} - -} // namespace - -MainWnd::MainWnd() - : ui_(CONNECT_TO_SERVER), wnd_(NULL), edit1_(NULL), edit2_(NULL), - label1_(NULL), label2_(NULL), button_(NULL), listbox_(NULL), - destroyed_(false), callback_(NULL), nested_msg_(NULL) { -} - -MainWnd::~MainWnd() { - ASSERT(!IsWindow()); -} - -bool MainWnd::Create() { - ASSERT(wnd_ == NULL); - if (!RegisterWindowClass()) - return false; - - ui_thread_id_ = ::GetCurrentThreadId(); - wnd_ = ::CreateWindowExW(WS_EX_OVERLAPPEDWINDOW, kClassName, L"WebRTC", - WS_OVERLAPPEDWINDOW | WS_VISIBLE | WS_CLIPCHILDREN, - CW_USEDEFAULT, CW_USEDEFAULT, CW_USEDEFAULT, CW_USEDEFAULT, - NULL, NULL, GetModuleHandle(NULL), this); - - ::SendMessage(wnd_, WM_SETFONT, reinterpret_cast(GetDefaultFont()), - TRUE); - - CreateChildWindows(); - SwitchToConnectUI(); - - return wnd_ != NULL; -} - -bool MainWnd::Destroy() { - BOOL ret = FALSE; - if (IsWindow()) { - ret = ::DestroyWindow(wnd_); - } - - return ret != FALSE; -} - -void MainWnd::RegisterObserver(MainWndCallback* callback) { - callback_ = callback; -} - -bool MainWnd::IsWindow() { - return wnd_ && ::IsWindow(wnd_) != FALSE; -} - -bool MainWnd::PreTranslateMessage(MSG* msg) { - bool ret = false; - if (msg->message == WM_CHAR) { - if (msg->wParam == VK_TAB) { - HandleTabbing(); - ret = true; - } else if (msg->wParam == VK_RETURN) { - OnDefaultAction(); - ret = true; - } else if (msg->wParam == VK_ESCAPE) { - if (callback_) { - if (ui_ == STREAMING) { - callback_->DisconnectFromCurrentPeer(); - } else { - callback_->DisconnectFromServer(); - } - } - } - } else if (msg->hwnd == NULL && msg->message == UI_THREAD_CALLBACK) { - callback_->UIThreadCallback(static_cast(msg->wParam), - reinterpret_cast(msg->lParam)); - ret = true; - } - return ret; -} - -void MainWnd::SwitchToConnectUI() { - ASSERT(IsWindow()); - LayoutPeerListUI(false); - ui_ = CONNECT_TO_SERVER; - LayoutConnectUI(true); - ::SetFocus(edit1_); -} - -void MainWnd::SwitchToPeerList(const Peers& peers) { - remote_video_.reset(); - local_video_.reset(); - - LayoutConnectUI(false); - - ::SendMessage(listbox_, LB_RESETCONTENT, 0, 0); - - AddListBoxItem(listbox_, "List of currently connected peers:", -1); - Peers::const_iterator i = peers.begin(); - for (; i != peers.end(); ++i) - AddListBoxItem(listbox_, i->second.c_str(), i->first); - - ui_ = LIST_PEERS; - LayoutPeerListUI(true); - ::SetFocus(listbox_); -} - -void MainWnd::SwitchToStreamingUI() { - LayoutConnectUI(false); - LayoutPeerListUI(false); - ui_ = STREAMING; -} - -void MainWnd::MessageBox(const char* caption, const char* text, bool is_error) { - DWORD flags = MB_OK; - if (is_error) - flags |= MB_ICONERROR; - - ::MessageBoxA(handle(), text, caption, flags); -} - -cricket::VideoRenderer* MainWnd::local_renderer() { - if (!local_video_.get()) - local_video_.reset(new VideoRenderer(handle(), 1, 1)); - return local_video_.get(); -} - -cricket::VideoRenderer* MainWnd::remote_renderer() { - if (!remote_video_.get()) - remote_video_.reset(new VideoRenderer(handle(), 1, 1)); - return remote_video_.get(); -} - -void MainWnd::QueueUIThreadCallback(int msg_id, void* data) { - ::PostThreadMessage(ui_thread_id_, UI_THREAD_CALLBACK, - static_cast(msg_id), reinterpret_cast(data)); -} - -void MainWnd::OnPaint() { - PAINTSTRUCT ps; - ::BeginPaint(handle(), &ps); - - RECT rc; - ::GetClientRect(handle(), &rc); - - if (ui_ == STREAMING && remote_video_.get() && local_video_.get()) { - AutoLock local_lock(local_video_.get()); - AutoLock remote_lock(remote_video_.get()); - - const BITMAPINFO& bmi = remote_video_->bmi(); - int height = abs(bmi.bmiHeader.biHeight); - int width = bmi.bmiHeader.biWidth; - - const uint8* image = remote_video_->image(); - if (image != NULL) { - HDC dc_mem = ::CreateCompatibleDC(ps.hdc); - ::SetStretchBltMode(dc_mem, HALFTONE); - - // Set the map mode so that the ratio will be maintained for us. - HDC all_dc[] = { ps.hdc, dc_mem }; - for (int i = 0; i < ARRAY_SIZE(all_dc); ++i) { - SetMapMode(all_dc[i], MM_ISOTROPIC); - SetWindowExtEx(all_dc[i], width, height, NULL); - SetViewportExtEx(all_dc[i], rc.right, rc.bottom, NULL); - } - - HBITMAP bmp_mem = ::CreateCompatibleBitmap(ps.hdc, rc.right, rc.bottom); - HGDIOBJ bmp_old = ::SelectObject(dc_mem, bmp_mem); - - POINT logical_area = { rc.right, rc.bottom }; - DPtoLP(ps.hdc, &logical_area, 1); - - HBRUSH brush = ::CreateSolidBrush(RGB(0, 0, 0)); - RECT logical_rect = {0, 0, logical_area.x, logical_area.y }; - ::FillRect(dc_mem, &logical_rect, brush); - ::DeleteObject(brush); - - int max_unit = std::max(width, height); - int x = (logical_area.x / 2) - (width / 2); - int y = (logical_area.y / 2) - (height / 2); - - StretchDIBits(dc_mem, x, y, width, height, - 0, 0, width, height, image, &bmi, DIB_RGB_COLORS, SRCCOPY); - - if ((rc.right - rc.left) > 200 && (rc.bottom - rc.top) > 200) { - const BITMAPINFO& bmi = local_video_->bmi(); - image = local_video_->image(); - int thumb_width = bmi.bmiHeader.biWidth / 4; - int thumb_height = abs(bmi.bmiHeader.biHeight) / 4; - StretchDIBits(dc_mem, - logical_area.x - thumb_width - 10, - logical_area.y - thumb_height - 10, - thumb_width, thumb_height, - 0, 0, bmi.bmiHeader.biWidth, -bmi.bmiHeader.biHeight, - image, &bmi, DIB_RGB_COLORS, SRCCOPY); - } - - BitBlt(ps.hdc, 0, 0, logical_area.x, logical_area.y, - dc_mem, 0, 0, SRCCOPY); - - // Cleanup. - ::SelectObject(dc_mem, bmp_old); - ::DeleteObject(bmp_mem); - ::DeleteDC(dc_mem); - } else { - // We're still waiting for the video stream to be initialized. - HBRUSH brush = ::CreateSolidBrush(RGB(0, 0, 0)); - ::FillRect(ps.hdc, &rc, brush); - ::DeleteObject(brush); - - HGDIOBJ old_font = ::SelectObject(ps.hdc, GetDefaultFont()); - ::SetTextColor(ps.hdc, RGB(0xff, 0xff, 0xff)); - ::SetBkMode(ps.hdc, TRANSPARENT); - - std::string text(kConnecting); - if (!local_video_->image()) { - text += kNoVideoStreams; - } else { - text += kNoIncomingStream; - } - ::DrawTextA(ps.hdc, text.c_str(), -1, &rc, - DT_SINGLELINE | DT_CENTER | DT_VCENTER); - ::SelectObject(ps.hdc, old_font); - } - } else { - HBRUSH brush = ::CreateSolidBrush(::GetSysColor(COLOR_WINDOW)); - ::FillRect(ps.hdc, &rc, brush); - ::DeleteObject(brush); - } - - ::EndPaint(handle(), &ps); -} - -void MainWnd::OnDestroyed() { - PostQuitMessage(0); -} - -void MainWnd::OnDefaultAction() { - if (!callback_) - return; - if (ui_ == CONNECT_TO_SERVER) { - std::string server(GetWindowText(edit1_)); - std::string port_str(GetWindowText(edit2_)); - int port = port_str.length() ? atoi(port_str.c_str()) : 0; - callback_->StartLogin(server, port); - } else if (ui_ == LIST_PEERS) { - LRESULT sel = ::SendMessage(listbox_, LB_GETCURSEL, 0, 0); - if (sel != LB_ERR) { - LRESULT peer_id = ::SendMessage(listbox_, LB_GETITEMDATA, sel, 0); - if (peer_id != -1 && callback_) { - callback_->ConnectToPeer(peer_id); - } - } - } else { - MessageBoxA(wnd_, "OK!", "Yeah", MB_OK); - } -} - -bool MainWnd::OnMessage(UINT msg, WPARAM wp, LPARAM lp, LRESULT* result) { - switch (msg) { - case WM_ERASEBKGND: - *result = TRUE; - return true; - - case WM_PAINT: - OnPaint(); - return true; - - case WM_SETFOCUS: - if (ui_ == CONNECT_TO_SERVER) { - SetFocus(edit1_); - } else if (ui_ == LIST_PEERS) { - SetFocus(listbox_); - } - return true; - - case WM_SIZE: - if (ui_ == CONNECT_TO_SERVER) { - LayoutConnectUI(true); - } else if (ui_ == LIST_PEERS) { - LayoutPeerListUI(true); - } - break; - - case WM_CTLCOLORSTATIC: - *result = reinterpret_cast(GetSysColorBrush(COLOR_WINDOW)); - return true; - - case WM_COMMAND: - if (button_ == reinterpret_cast(lp)) { - if (BN_CLICKED == HIWORD(wp)) - OnDefaultAction(); - } else if (listbox_ == reinterpret_cast(lp)) { - if (LBN_DBLCLK == HIWORD(wp)) { - OnDefaultAction(); - } - } - return true; - - case WM_CLOSE: - if (callback_) - callback_->Close(); - break; - } - return false; -} - -// static -LRESULT CALLBACK MainWnd::WndProc(HWND hwnd, UINT msg, WPARAM wp, LPARAM lp) { - MainWnd* me = reinterpret_cast( - ::GetWindowLongPtr(hwnd, GWL_USERDATA)); - if (!me && WM_CREATE == msg) { - CREATESTRUCT* cs = reinterpret_cast(lp); - me = reinterpret_cast(cs->lpCreateParams); - me->wnd_ = hwnd; - ::SetWindowLongPtr(hwnd, GWL_USERDATA, reinterpret_cast(me)); - } - - LRESULT result = 0; - if (me) { - void* prev_nested_msg = me->nested_msg_; - me->nested_msg_ = &msg; - - bool handled = me->OnMessage(msg, wp, lp, &result); - if (WM_NCDESTROY == msg) { - me->destroyed_ = true; - } else if (!handled) { - result = ::DefWindowProc(hwnd, msg, wp, lp); - } - - if (me->destroyed_ && prev_nested_msg == NULL) { - me->OnDestroyed(); - me->wnd_ = NULL; - me->destroyed_ = false; - } - - me->nested_msg_ = prev_nested_msg; - } else { - result = ::DefWindowProc(hwnd, msg, wp, lp); - } - - return result; -} - -// static -bool MainWnd::RegisterWindowClass() { - if (wnd_class_) - return true; - - WNDCLASSEX wcex = { sizeof(WNDCLASSEX) }; - wcex.style = CS_DBLCLKS; - wcex.hInstance = GetModuleHandle(NULL); - wcex.hbrBackground = reinterpret_cast(COLOR_WINDOW + 1); - wcex.hCursor = ::LoadCursor(NULL, IDC_ARROW); - wcex.lpfnWndProc = &WndProc; - wcex.lpszClassName = kClassName; - wnd_class_ = ::RegisterClassEx(&wcex); - ASSERT(wnd_class_ != 0); - return wnd_class_ != 0; -} - -void MainWnd::CreateChildWindow(HWND* wnd, MainWnd::ChildWindowID id, - const wchar_t* class_name, DWORD control_style, - DWORD ex_style) { - if (::IsWindow(*wnd)) - return; - - // Child windows are invisible at first, and shown after being resized. - DWORD style = WS_CHILD | control_style; - *wnd = ::CreateWindowEx(ex_style, class_name, L"", style, - 100, 100, 100, 100, wnd_, - reinterpret_cast(id), - GetModuleHandle(NULL), NULL); - ASSERT(::IsWindow(*wnd) != FALSE); - ::SendMessage(*wnd, WM_SETFONT, reinterpret_cast(GetDefaultFont()), - TRUE); -} - -void MainWnd::CreateChildWindows() { - // Create the child windows in tab order. - CreateChildWindow(&label1_, LABEL1_ID, L"Static", ES_CENTER | ES_READONLY, 0); - CreateChildWindow(&edit1_, EDIT_ID, L"Edit", - ES_LEFT | ES_NOHIDESEL | WS_TABSTOP, WS_EX_CLIENTEDGE); - CreateChildWindow(&label2_, LABEL2_ID, L"Static", ES_CENTER | ES_READONLY, 0); - CreateChildWindow(&edit2_, EDIT_ID, L"Edit", - ES_LEFT | ES_NOHIDESEL | WS_TABSTOP, WS_EX_CLIENTEDGE); - CreateChildWindow(&button_, BUTTON_ID, L"Button", BS_CENTER | WS_TABSTOP, 0); - - CreateChildWindow(&listbox_, LISTBOX_ID, L"ListBox", - LBS_HASSTRINGS | LBS_NOTIFY, WS_EX_CLIENTEDGE); - - ::SetWindowTextA(edit1_, GetDefaultServerName().c_str()); - ::SetWindowTextA(edit2_, "8888"); -} - -void MainWnd::LayoutConnectUI(bool show) { - struct Windows { - HWND wnd; - const wchar_t* text; - size_t width; - size_t height; - } windows[] = { - { label1_, L"Server" }, - { edit1_, L"XXXyyyYYYgggXXXyyyYYYggg" }, - { label2_, L":" }, - { edit2_, L"XyXyX" }, - { button_, L"Connect" }, - }; - - if (show) { - const size_t kSeparator = 5; - size_t total_width = (ARRAYSIZE(windows) - 1) * kSeparator; - - for (size_t i = 0; i < ARRAYSIZE(windows); ++i) { - CalculateWindowSizeForText(windows[i].wnd, windows[i].text, - &windows[i].width, &windows[i].height); - total_width += windows[i].width; - } - - RECT rc; - ::GetClientRect(wnd_, &rc); - size_t x = (rc.right / 2) - (total_width / 2); - size_t y = rc.bottom / 2; - for (size_t i = 0; i < ARRAYSIZE(windows); ++i) { - size_t top = y - (windows[i].height / 2); - ::MoveWindow(windows[i].wnd, x, top, windows[i].width, windows[i].height, - TRUE); - x += kSeparator + windows[i].width; - if (windows[i].text[0] != 'X') - ::SetWindowText(windows[i].wnd, windows[i].text); - ::ShowWindow(windows[i].wnd, SW_SHOWNA); - } - } else { - for (size_t i = 0; i < ARRAYSIZE(windows); ++i) { - ::ShowWindow(windows[i].wnd, SW_HIDE); - } - } -} - -void MainWnd::LayoutPeerListUI(bool show) { - if (show) { - RECT rc; - ::GetClientRect(wnd_, &rc); - ::MoveWindow(listbox_, 0, 0, rc.right, rc.bottom, TRUE); - ::ShowWindow(listbox_, SW_SHOWNA); - } else { - ::ShowWindow(listbox_, SW_HIDE); - InvalidateRect(wnd_, NULL, TRUE); - } -} - -void MainWnd::HandleTabbing() { - bool shift = ((::GetAsyncKeyState(VK_SHIFT) & 0x8000) != 0); - UINT next_cmd = shift ? GW_HWNDPREV : GW_HWNDNEXT; - UINT loop_around_cmd = shift ? GW_HWNDLAST : GW_HWNDFIRST; - HWND focus = GetFocus(), next; - do { - next = ::GetWindow(focus, next_cmd); - if (IsWindowVisible(next) && - (GetWindowLong(next, GWL_STYLE) & WS_TABSTOP)) { - break; - } - - if (!next) { - next = ::GetWindow(focus, loop_around_cmd); - if (IsWindowVisible(next) && - (GetWindowLong(next, GWL_STYLE) & WS_TABSTOP)) { - break; - } - } - focus = next; - } while (true); - ::SetFocus(next); -} - -// -// MainWnd::VideoRenderer -// - -MainWnd::VideoRenderer::VideoRenderer(HWND wnd, int width, int height) - : wnd_(wnd) { - ::InitializeCriticalSection(&buffer_lock_); - ZeroMemory(&bmi_, sizeof(bmi_)); - bmi_.bmiHeader.biSize = sizeof(BITMAPINFOHEADER); - bmi_.bmiHeader.biPlanes = 1; - bmi_.bmiHeader.biBitCount = 32; - bmi_.bmiHeader.biCompression = BI_RGB; - bmi_.bmiHeader.biWidth = width; - bmi_.bmiHeader.biHeight = -height; - bmi_.bmiHeader.biSizeImage = width * height * - (bmi_.bmiHeader.biBitCount >> 3); -} - -MainWnd::VideoRenderer::~VideoRenderer() { - ::DeleteCriticalSection(&buffer_lock_); -} - -bool MainWnd::VideoRenderer::SetSize(int width, int height, int reserved) { - AutoLock lock(this); - - bmi_.bmiHeader.biWidth = width; - bmi_.bmiHeader.biHeight = -height; - bmi_.bmiHeader.biSizeImage = width * height * - (bmi_.bmiHeader.biBitCount >> 3); - image_.reset(new uint8[bmi_.bmiHeader.biSizeImage]); - - return true; -} - -bool MainWnd::VideoRenderer::RenderFrame(const cricket::VideoFrame* frame) { - if (!frame) - return false; - - { - AutoLock lock(this); - - ASSERT(image_.get() != NULL); - frame->ConvertToRgbBuffer(cricket::FOURCC_ARGB, image_.get(), - bmi_.bmiHeader.biSizeImage, - bmi_.bmiHeader.biWidth * - (bmi_.bmiHeader.biBitCount >> 3)); - } - - InvalidateRect(wnd_, NULL, TRUE); - - return true; -} diff --git a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/main_wnd.h b/third_party_mods/libjingle/source/talk/examples/peerconnection_client/main_wnd.h deleted file mode 100644 index 5b514c08c..000000000 --- a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/main_wnd.h +++ /dev/null @@ -1,191 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef PEERCONNECTION_SAMPLES_CLIENT_MAIN_WND_H_ -#define PEERCONNECTION_SAMPLES_CLIENT_MAIN_WND_H_ -#pragma once - -#include -#include - -#include "talk/app/webrtc_dev/mediastream.h" -#include "talk/examples/peerconnection_client/peer_connection_client.h" -#include "talk/base/win32.h" -#include "talk/session/phone/mediachannel.h" -#include "talk/session/phone/videocommon.h" -#include "talk/session/phone/videoframe.h" -#include "talk/session/phone/videorenderer.h" - -class MainWndCallback { - public: - virtual bool StartLogin(const std::string& server, int port) = 0; - virtual void DisconnectFromServer() = 0; - virtual void ConnectToPeer(int peer_id) = 0; - virtual void DisconnectFromCurrentPeer() = 0; - virtual void UIThreadCallback(int msg_id, void* data) = 0; - virtual void Close() = 0; - protected: - virtual ~MainWndCallback() {} -}; - -// Pure virtual interface for the main window. -class MainWindow { - public: - virtual ~MainWindow() {} - - enum UI { - CONNECT_TO_SERVER, - LIST_PEERS, - STREAMING, - }; - - virtual void RegisterObserver(MainWndCallback* callback) = 0; - - virtual bool IsWindow() = 0; - virtual void MessageBox(const char* caption, const char* text, - bool is_error) = 0; - - virtual UI current_ui() = 0; - - virtual void SwitchToConnectUI() = 0; - virtual void SwitchToPeerList(const Peers& peers) = 0; - virtual void SwitchToStreamingUI() = 0; - - virtual webrtc::VideoRendererWrapperInterface* local_renderer() = 0; - virtual webrtc::VideoRendererWrapperInterface* remote_renderer() = 0; - - virtual void QueueUIThreadCallback(int msg_id, void* data) = 0; -}; - -#ifdef WIN32 - -class MainWnd : public MainWindow { - public: - static const wchar_t kClassName[]; - - enum WindowMessages { - UI_THREAD_CALLBACK = WM_APP + 1, - }; - - MainWnd(); - ~MainWnd(); - - bool Create(); - bool Destroy(); - bool PreTranslateMessage(MSG* msg); - - virtual void RegisterObserver(MainWndCallback* callback); - virtual bool IsWindow(); - virtual void SwitchToConnectUI(); - virtual void SwitchToPeerList(const Peers& peers); - virtual void SwitchToStreamingUI(); - virtual void MessageBox(const char* caption, const char* text, - bool is_error); - virtual UI current_ui() { return ui_; } - - virtual cricket::VideoRenderer* local_renderer(); - virtual cricket::VideoRenderer* remote_renderer(); - - virtual void QueueUIThreadCallback(int msg_id, void* data); - - HWND handle() const { return wnd_; } - - class VideoRenderer : public cricket::VideoRenderer { - public: - VideoRenderer(HWND wnd, int width, int height); - virtual ~VideoRenderer(); - - void Lock() { - ::EnterCriticalSection(&buffer_lock_); - } - - void Unlock() { - ::LeaveCriticalSection(&buffer_lock_); - } - - virtual bool SetSize(int width, int height, int reserved); - - // Called when a new frame is available for display. - virtual bool RenderFrame(const cricket::VideoFrame* frame); - - const BITMAPINFO& bmi() const { return bmi_; } - const uint8* image() const { return image_.get(); } - - protected: - enum { - SET_SIZE, - RENDER_FRAME, - }; - - HWND wnd_; - BITMAPINFO bmi_; - talk_base::scoped_array image_; - CRITICAL_SECTION buffer_lock_; - }; - - // A little helper class to make sure we always to proper locking and - // unlocking when working with VideoRenderer buffers. - template - class AutoLock { - public: - explicit AutoLock(T* obj) : obj_(obj) { obj_->Lock(); } - ~AutoLock() { obj_->Unlock(); } - protected: - T* obj_; - }; - - protected: - enum ChildWindowID { - EDIT_ID = 1, - BUTTON_ID, - LABEL1_ID, - LABEL2_ID, - LISTBOX_ID, - }; - - void OnPaint(); - void OnDestroyed(); - - void OnDefaultAction(); - - bool OnMessage(UINT msg, WPARAM wp, LPARAM lp, LRESULT* result); - - static LRESULT CALLBACK WndProc(HWND hwnd, UINT msg, WPARAM wp, LPARAM lp); - static bool RegisterWindowClass(); - - void CreateChildWindow(HWND* wnd, ChildWindowID id, const wchar_t* class_name, - DWORD control_style, DWORD ex_style); - void CreateChildWindows(); - - void LayoutConnectUI(bool show); - void LayoutPeerListUI(bool show); - - void HandleTabbing(); - - private: - talk_base::scoped_ptr remote_video_; - talk_base::scoped_ptr local_video_; - UI ui_; - HWND wnd_; - DWORD ui_thread_id_; - HWND edit1_; - HWND edit2_; - HWND label1_; - HWND label2_; - HWND button_; - HWND listbox_; - bool destroyed_; - void* nested_msg_; - MainWndCallback* callback_; - static ATOM wnd_class_; -}; -#endif // WIN32 - -#endif // PEERCONNECTION_SAMPLES_CLIENT_MAIN_WND_H_ diff --git a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/peer_connection_client.cc b/third_party_mods/libjingle/source/talk/examples/peerconnection_client/peer_connection_client.cc deleted file mode 100644 index 3de4c311e..000000000 --- a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/peer_connection_client.cc +++ /dev/null @@ -1,482 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "talk/examples/peerconnection_client/peer_connection_client.h" - -#include "talk/examples/peerconnection_client/defaults.h" -#include "talk/base/common.h" -#include "talk/base/nethelpers.h" -#include "talk/base/logging.h" -#include "talk/base/stringutils.h" - -#ifdef WIN32 -#include "talk/base/win32socketserver.h" -#endif - -using talk_base::sprintfn; - -namespace { - -// This is our magical hangup signal. -const char kByeMessage[] = "BYE"; - -talk_base::AsyncSocket* CreateClientSocket() { -#ifdef WIN32 - return new talk_base::Win32Socket(); -#elif defined(POSIX) - talk_base::Thread* thread = talk_base::Thread::Current(); - ASSERT(thread != NULL); - return thread->socketserver()->CreateAsyncSocket(SOCK_STREAM); -#else -#error Platform not supported. -#endif -} - -} - -PeerConnectionClient::PeerConnectionClient() - : callback_(NULL), - control_socket_(CreateClientSocket()), - hanging_get_(CreateClientSocket()), - state_(NOT_CONNECTED), - my_id_(-1) { - control_socket_->SignalCloseEvent.connect(this, - &PeerConnectionClient::OnClose); - hanging_get_->SignalCloseEvent.connect(this, - &PeerConnectionClient::OnClose); - control_socket_->SignalConnectEvent.connect(this, - &PeerConnectionClient::OnConnect); - hanging_get_->SignalConnectEvent.connect(this, - &PeerConnectionClient::OnHangingGetConnect); - control_socket_->SignalReadEvent.connect(this, - &PeerConnectionClient::OnRead); - hanging_get_->SignalReadEvent.connect(this, - &PeerConnectionClient::OnHangingGetRead); -} - -PeerConnectionClient::~PeerConnectionClient() { -} - -int PeerConnectionClient::id() const { - return my_id_; -} - -bool PeerConnectionClient::is_connected() const { - return my_id_ != -1; -} - -const Peers& PeerConnectionClient::peers() const { - return peers_; -} - -void PeerConnectionClient::RegisterObserver( - PeerConnectionClientObserver* callback) { - ASSERT(!callback_); - callback_ = callback; -} - -bool PeerConnectionClient::Connect(const std::string& server, int port, - const std::string& client_name) { - ASSERT(!server.empty()); - ASSERT(!client_name.empty()); - - if (state_ != NOT_CONNECTED) { - LOG(WARNING) - << "The client must not be connected before you can call Connect()"; - return false; - } - - if (server.empty() || client_name.empty()) - return false; - - if (port <= 0) - port = kDefaultServerPort; - - server_address_.SetIP(server); - server_address_.SetPort(port); - - if (server_address_.IsUnresolved()) { - int errcode = 0; - hostent* h = talk_base::SafeGetHostByName( - server_address_.IPAsString().c_str(), &errcode); - if (!h) { - LOG(LS_ERROR) << "Failed to resolve host name: " - << server_address_.IPAsString(); - return false; - } else { - server_address_.SetResolvedIP( - ntohl(*reinterpret_cast(h->h_addr_list[0]))); - talk_base::FreeHostEnt(h); - } - } - - char buffer[1024]; - sprintfn(buffer, sizeof(buffer), - "GET /sign_in?%s HTTP/1.0\r\n\r\n", client_name.c_str()); - onconnect_data_ = buffer; - - bool ret = ConnectControlSocket(); - if (ret) - state_ = SIGNING_IN; - - return ret; -} - -bool PeerConnectionClient::SendToPeer(int peer_id, const std::string& message) { - if (state_ != CONNECTED) - return false; - - ASSERT(is_connected()); - ASSERT(control_socket_->GetState() == talk_base::Socket::CS_CLOSED); - if (!is_connected() || peer_id == -1) - return false; - - char headers[1024]; - sprintfn(headers, sizeof(headers), - "POST /message?peer_id=%i&to=%i HTTP/1.0\r\n" - "Content-Length: %i\r\n" - "Content-Type: text/plain\r\n" - "\r\n", - my_id_, peer_id, message.length()); - onconnect_data_ = headers; - onconnect_data_ += message; - return ConnectControlSocket(); -} - -bool PeerConnectionClient::SendHangUp(int peer_id) { - return SendToPeer(peer_id, kByeMessage); -} - -bool PeerConnectionClient::IsSendingMessage() { - return state_ == CONNECTED && - control_socket_->GetState() != talk_base::Socket::CS_CLOSED; -} - -bool PeerConnectionClient::SignOut() { - if (state_ == NOT_CONNECTED || state_ == SIGNING_OUT) - return true; - - if (hanging_get_->GetState() != talk_base::Socket::CS_CLOSED) - hanging_get_->Close(); - - if (control_socket_->GetState() == talk_base::Socket::CS_CLOSED) { - state_ = SIGNING_OUT; - - if (my_id_ != -1) { - char buffer[1024]; - sprintfn(buffer, sizeof(buffer), - "GET /sign_out?peer_id=%i HTTP/1.0\r\n\r\n", my_id_); - onconnect_data_ = buffer; - return ConnectControlSocket(); - } else { - // Can occur if the app is closed before we finish connecting. - return true; - } - } else { - state_ = SIGNING_OUT_WAITING; - } - - return true; -} - -void PeerConnectionClient::Close() { - control_socket_->Close(); - hanging_get_->Close(); - onconnect_data_.clear(); - peers_.clear(); - my_id_ = -1; - state_ = NOT_CONNECTED; -} - -bool PeerConnectionClient::ConnectControlSocket() { - ASSERT(control_socket_->GetState() == talk_base::Socket::CS_CLOSED); - int err = control_socket_->Connect(server_address_); - if (err == SOCKET_ERROR) { - Close(); - return false; - } - return true; -} - -void PeerConnectionClient::OnConnect(talk_base::AsyncSocket* socket) { - ASSERT(!onconnect_data_.empty()); - size_t sent = socket->Send(onconnect_data_.c_str(), onconnect_data_.length()); - ASSERT(sent == onconnect_data_.length()); - UNUSED(sent); - onconnect_data_.clear(); -} - -void PeerConnectionClient::OnHangingGetConnect(talk_base::AsyncSocket* socket) { - char buffer[1024]; - sprintfn(buffer, sizeof(buffer), - "GET /wait?peer_id=%i HTTP/1.0\r\n\r\n", my_id_); - int len = strlen(buffer); - int sent = socket->Send(buffer, len); - ASSERT(sent == len); - UNUSED2(sent, len); -} - -void PeerConnectionClient::OnMessageFromPeer(int peer_id, - const std::string& message) { - if (message.length() == (sizeof(kByeMessage) - 1) && - message.compare(kByeMessage) == 0) { - callback_->OnPeerDisconnected(peer_id); - } else { - callback_->OnMessageFromPeer(peer_id, message); - } -} - -bool PeerConnectionClient::GetHeaderValue(const std::string& data, - size_t eoh, - const char* header_pattern, - size_t* value) { - ASSERT(value != NULL); - size_t found = data.find(header_pattern); - if (found != std::string::npos && found < eoh) { - *value = atoi(&data[found + strlen(header_pattern)]); - return true; - } - return false; -} - -bool PeerConnectionClient::GetHeaderValue(const std::string& data, size_t eoh, - const char* header_pattern, - std::string* value) { - ASSERT(value != NULL); - size_t found = data.find(header_pattern); - if (found != std::string::npos && found < eoh) { - size_t begin = found + strlen(header_pattern); - size_t end = data.find("\r\n", begin); - if (end == std::string::npos) - end = eoh; - value->assign(data.substr(begin, end - begin)); - return true; - } - return false; -} - -bool PeerConnectionClient::ReadIntoBuffer(talk_base::AsyncSocket* socket, - std::string* data, - size_t* content_length) { - LOG(INFO) << __FUNCTION__; - - char buffer[0xffff]; - do { - int bytes = socket->Recv(buffer, sizeof(buffer)); - if (bytes <= 0) - break; - data->append(buffer, bytes); - } while (true); - - bool ret = false; - size_t i = data->find("\r\n\r\n"); - if (i != std::string::npos) { - LOG(INFO) << "Headers received"; - if (GetHeaderValue(*data, i, "\r\nContent-Length: ", content_length)) { - LOG(INFO) << "Expecting " << *content_length << " bytes."; - size_t total_response_size = (i + 4) + *content_length; - if (data->length() >= total_response_size) { - ret = true; - std::string should_close; - const char kConnection[] = "\r\nConnection: "; - if (GetHeaderValue(*data, i, kConnection, &should_close) && - should_close.compare("close") == 0) { - socket->Close(); - // Since we closed the socket, there was no notification delivered - // to us. Compensate by letting ourselves know. - OnClose(socket, 0); - } - } else { - // We haven't received everything. Just continue to accept data. - } - } else { - LOG(LS_ERROR) << "No content length field specified by the server."; - } - } - return ret; -} - -void PeerConnectionClient::OnRead(talk_base::AsyncSocket* socket) { - LOG(INFO) << __FUNCTION__; - size_t content_length = 0; - if (ReadIntoBuffer(socket, &control_data_, &content_length)) { - size_t peer_id = 0, eoh = 0; - bool ok = ParseServerResponse(control_data_, content_length, &peer_id, - &eoh); - if (ok) { - if (my_id_ == -1) { - // First response. Let's store our server assigned ID. - ASSERT(state_ == SIGNING_IN); - my_id_ = peer_id; - ASSERT(my_id_ != -1); - - // The body of the response will be a list of already connected peers. - if (content_length) { - size_t pos = eoh + 4; - while (pos < control_data_.size()) { - size_t eol = control_data_.find('\n', pos); - if (eol == std::string::npos) - break; - int id = 0; - std::string name; - bool connected; - if (ParseEntry(control_data_.substr(pos, eol - pos), &name, &id, - &connected) && id != my_id_) { - peers_[id] = name; - callback_->OnPeerConnected(id, name); - } - pos = eol + 1; - } - } - ASSERT(is_connected()); - callback_->OnSignedIn(); - } else if (state_ == SIGNING_OUT) { - Close(); - callback_->OnDisconnected(); - } else if (state_ == SIGNING_OUT_WAITING) { - SignOut(); - } - } - - control_data_.clear(); - - if (state_ == SIGNING_IN) { - ASSERT(hanging_get_->GetState() == talk_base::Socket::CS_CLOSED); - state_ = CONNECTED; - hanging_get_->Connect(server_address_); - } - } -} - -void PeerConnectionClient::OnHangingGetRead(talk_base::AsyncSocket* socket) { - LOG(INFO) << __FUNCTION__; - size_t content_length = 0; - if (ReadIntoBuffer(socket, ¬ification_data_, &content_length)) { - size_t peer_id = 0, eoh = 0; - bool ok = ParseServerResponse(notification_data_, content_length, - &peer_id, &eoh); - - if (ok) { - // Store the position where the body begins. - size_t pos = eoh + 4; - - if (my_id_ == static_cast(peer_id)) { - // A notification about a new member or a member that just - // disconnected. - int id = 0; - std::string name; - bool connected = false; - if (ParseEntry(notification_data_.substr(pos), &name, &id, - &connected)) { - if (connected) { - peers_[id] = name; - callback_->OnPeerConnected(id, name); - } else { - peers_.erase(id); - callback_->OnPeerDisconnected(id); - } - } - } else { - OnMessageFromPeer(peer_id, notification_data_.substr(pos)); - } - } - - notification_data_.clear(); - } - - if (hanging_get_->GetState() == talk_base::Socket::CS_CLOSED && - state_ == CONNECTED) { - hanging_get_->Connect(server_address_); - } -} - -bool PeerConnectionClient::ParseEntry(const std::string& entry, - std::string* name, - int* id, - bool* connected) { - ASSERT(name != NULL); - ASSERT(id != NULL); - ASSERT(connected != NULL); - ASSERT(!entry.empty()); - - *connected = false; - size_t separator = entry.find(','); - if (separator != std::string::npos) { - *id = atoi(&entry[separator + 1]); - name->assign(entry.substr(0, separator)); - separator = entry.find(',', separator + 1); - if (separator != std::string::npos) { - *connected = atoi(&entry[separator + 1]) ? true : false; - } - } - return !name->empty(); -} - -int PeerConnectionClient::GetResponseStatus(const std::string& response) { - int status = -1; - size_t pos = response.find(' '); - if (pos != std::string::npos) - status = atoi(&response[pos + 1]); - return status; -} - -bool PeerConnectionClient::ParseServerResponse(const std::string& response, - size_t content_length, - size_t* peer_id, - size_t* eoh) { - LOG(INFO) << response; - - int status = GetResponseStatus(response.c_str()); - if (status != 200) { - LOG(LS_ERROR) << "Received error from server"; - Close(); - callback_->OnDisconnected(); - return false; - } - - *eoh = response.find("\r\n\r\n"); - ASSERT(*eoh != std::string::npos); - if (*eoh == std::string::npos) - return false; - - *peer_id = -1; - - // See comment in peer_channel.cc for why we use the Pragma header and - // not e.g. "X-Peer-Id". - GetHeaderValue(response, *eoh, "\r\nPragma: ", peer_id); - - return true; -} - -void PeerConnectionClient::OnClose(talk_base::AsyncSocket* socket, int err) { - LOG(INFO) << __FUNCTION__; - - socket->Close(); - -#ifdef WIN32 - if (err != WSAECONNREFUSED) { -#else - if (err != ECONNREFUSED) { -#endif - if (socket == hanging_get_.get()) { - if (state_ == CONNECTED) { - LOG(INFO) << "Issuing a new hanging get"; - hanging_get_->Close(); - hanging_get_->Connect(server_address_); - } - } else { - callback_->OnMessageSent(err); - } - } else { - LOG(WARNING) << "Failed to connect to the server"; - Close(); - callback_->OnDisconnected(); - } -} diff --git a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/peer_connection_client.h b/third_party_mods/libjingle/source/talk/examples/peerconnection_client/peer_connection_client.h deleted file mode 100644 index bc02bdd82..000000000 --- a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/peer_connection_client.h +++ /dev/null @@ -1,109 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef PEERCONNECTION_SAMPLES_CLIENT_PEER_CONNECTION_CLIENT_H_ -#define PEERCONNECTION_SAMPLES_CLIENT_PEER_CONNECTION_CLIENT_H_ -#pragma once - -#include -#include - -#include "talk/base/sigslot.h" -#include "talk/base/physicalsocketserver.h" -#include "talk/base/scoped_ptr.h" - -typedef std::map Peers; - -struct PeerConnectionClientObserver { - virtual void OnSignedIn() = 0; // Called when we're logged on. - virtual void OnDisconnected() = 0; - virtual void OnPeerConnected(int id, const std::string& name) = 0; - virtual void OnPeerDisconnected(int peer_id) = 0; - virtual void OnMessageFromPeer(int peer_id, const std::string& message) = 0; - virtual void OnMessageSent(int err) = 0; - - protected: - virtual ~PeerConnectionClientObserver() {} -}; - -class PeerConnectionClient : public sigslot::has_slots<> { - public: - enum State { - NOT_CONNECTED, - SIGNING_IN, - CONNECTED, - SIGNING_OUT_WAITING, - SIGNING_OUT, - }; - - PeerConnectionClient(); - ~PeerConnectionClient(); - - int id() const; - bool is_connected() const; - const Peers& peers() const; - - void RegisterObserver(PeerConnectionClientObserver* callback); - - bool Connect(const std::string& server, int port, - const std::string& client_name); - - bool SendToPeer(int peer_id, const std::string& message); - bool SendHangUp(int peer_id); - bool IsSendingMessage(); - - bool SignOut(); - - protected: - void Close(); - bool ConnectControlSocket(); - void OnConnect(talk_base::AsyncSocket* socket); - void OnHangingGetConnect(talk_base::AsyncSocket* socket); - void OnMessageFromPeer(int peer_id, const std::string& message); - - // Quick and dirty support for parsing HTTP header values. - bool GetHeaderValue(const std::string& data, size_t eoh, - const char* header_pattern, size_t* value); - - bool GetHeaderValue(const std::string& data, size_t eoh, - const char* header_pattern, std::string* value); - - // Returns true if the whole response has been read. - bool ReadIntoBuffer(talk_base::AsyncSocket* socket, std::string* data, - size_t* content_length); - - void OnRead(talk_base::AsyncSocket* socket); - - void OnHangingGetRead(talk_base::AsyncSocket* socket); - - // Parses a single line entry in the form ",," - bool ParseEntry(const std::string& entry, std::string* name, int* id, - bool* connected); - - int GetResponseStatus(const std::string& response); - - bool ParseServerResponse(const std::string& response, size_t content_length, - size_t* peer_id, size_t* eoh); - - void OnClose(talk_base::AsyncSocket* socket, int err); - - PeerConnectionClientObserver* callback_; - talk_base::SocketAddress server_address_; - talk_base::scoped_ptr control_socket_; - talk_base::scoped_ptr hanging_get_; - std::string onconnect_data_; - std::string control_data_; - std::string notification_data_; - Peers peers_; - State state_; - int my_id_; -}; - -#endif // PEERCONNECTION_SAMPLES_CLIENT_PEER_CONNECTION_CLIENT_H_ diff --git a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/peerconnection_client.gyp b/third_party_mods/libjingle/source/talk/examples/peerconnection_client/peerconnection_client.gyp deleted file mode 100644 index 4592646ef..000000000 --- a/third_party_mods/libjingle/source/talk/examples/peerconnection_client/peerconnection_client.gyp +++ /dev/null @@ -1,59 +0,0 @@ -# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. -# -# Use of this source code is governed by a BSD-style license -# that can be found in the LICENSE file in the root of the source -# tree. An additional intellectual property rights grant can be found -# in the file PATENTS. All contributing project authors may -# be found in the AUTHORS file in the root of the source tree. - -{ - 'conditions': [ - ['OS=="linux"', { - 'targets': [ - { - 'target_name': 'peerconnection_client_dev', - 'type': 'executable', - 'sources': [ - 'conductor.cc', - 'conductor.h', - 'defaults.cc', - 'defaults.h', - 'linux/main.cc', - 'linux/main_wnd.cc', - 'linux/main_wnd.h', - 'peer_connection_client.cc', - 'peer_connection_client.h', - ], - 'dependencies': [ - '../../../../libjingle.gyp:libjingle_app', - '../../../../../../src/modules/modules.gyp:video_capture_module', - '../../../../../../src/system_wrappers/source/' - 'system_wrappers.gyp:system_wrappers', - # TODO(tommi): Switch to this and remove specific gtk dependency - # sections below for cflags and link_settings. - # '<(DEPTH)/build/linux/system.gyp:gtk', - ], - 'include_dirs': [ - '../../../', - '../../../../../../src', # webrtc modules - #TODO(perkj): Remove when this project is in the correct folder. - '../../../../../../third_party/libjingle/source/', - ], - 'cflags': [ - ' - -PeerConnection server test page - - - - - -Server:
- Loopback (just send -received messages right back)
-Your name: - - -
-
-Target peer id: -Message: - -
- - -
-
-
- - diff --git a/third_party_mods/libjingle/source/talk/p2p/client/fakeportallocator.h b/third_party_mods/libjingle/source/talk/p2p/client/fakeportallocator.h deleted file mode 100644 index 9df5ccaa8..000000000 --- a/third_party_mods/libjingle/source/talk/p2p/client/fakeportallocator.h +++ /dev/null @@ -1,92 +0,0 @@ -// Copyright 2010 Google Inc. All Rights Reserved, -// -// Author: Justin Uberti (juberti@google.com) - -#ifndef TALK_P2P_CLIENT_FAKEPORTALLOCATOR_H_ -#define TALK_P2P_CLIENT_FAKEPORTALLOCATOR_H_ - -#include -#include "talk/base/basicpacketsocketfactory.h" -#include "talk/base/scoped_ptr.h" -#include "talk/p2p/base/portallocator.h" -#include "talk/p2p/base/udpport.h" - -namespace talk_base { -class SocketFactory; -class Thread; -} - -namespace cricket { - -class FakePortAllocatorSession : public PortAllocatorSession { - public: - FakePortAllocatorSession(talk_base::Thread* worker_thread, - talk_base::PacketSocketFactory* factory, - const std::string& name, - const std::string& session_type) - : PortAllocatorSession(0), worker_thread_(worker_thread), - factory_(factory), name_(name), - network_("network", "unittest", 0x7F000001, 0), - port_(NULL), running_(false) { - } - - virtual void GetInitialPorts() { - if (!port_.get()) { - port_.reset(cricket::UDPPort::Create(worker_thread_, factory_, - &network_, network_.ip(), 0, 0)); - AddPort(port_.get()); - } - } - virtual void StartGetAllPorts() { running_ = true; } - virtual void StopGetAllPorts() { running_ = false; } - virtual bool IsGettingAllPorts() { return running_; } - - void AddPort(cricket::Port* port) { - port->set_name(name_); - port->set_preference(1.0); - port->set_generation(0); - port->SignalAddressReady.connect( - this, &FakePortAllocatorSession::OnAddressReady); - port->PrepareAddress(); - SignalPortReady(this, port); - } - void OnAddressReady(cricket::Port* port) { - SignalCandidatesReady(this, port->candidates()); - } - - private: - talk_base::Thread* worker_thread_; - talk_base::PacketSocketFactory* factory_; - std::string name_; - talk_base::Network network_; - talk_base::scoped_ptr port_; - bool running_; -}; - -class FakePortAllocator : public cricket::PortAllocator { - public: - FakePortAllocator(talk_base::Thread* worker_thread, - talk_base::PacketSocketFactory* factory) - : worker_thread_(worker_thread), factory_(factory) { - if (factory_ == NULL) { - owned_factory_.reset(new talk_base::BasicPacketSocketFactory( - worker_thread_)); - factory_ = owned_factory_.get(); - } - } - - virtual cricket::PortAllocatorSession* CreateSession( - const std::string &name, const std::string &session_type) { - return new FakePortAllocatorSession(worker_thread_, factory_, name, - session_type); - } - - private: - talk_base::Thread* worker_thread_; - talk_base::PacketSocketFactory* factory_; - talk_base::scoped_ptr owned_factory_; -}; - -} // namespace cricket - -#endif // TALK_P2P_CLIENT_FAKEPORTALLOCATOR_H_ diff --git a/third_party_mods/libjingle/source/talk/session/phone/channel.cc b/third_party_mods/libjingle/source/talk/session/phone/channel.cc deleted file mode 100644 index 812fb4e98..000000000 --- a/third_party_mods/libjingle/source/talk/session/phone/channel.cc +++ /dev/null @@ -1,1317 +0,0 @@ -/* - * libjingle - * Copyright 2004--2007, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include "talk/session/phone/channel.h" - -#include "talk/base/buffer.h" -#include "talk/base/byteorder.h" -#include "talk/base/common.h" -#include "talk/base/logging.h" -#include "talk/p2p/base/transportchannel.h" -#include "talk/session/phone/channelmanager.h" -#include "talk/session/phone/mediasessionclient.h" -#include "talk/session/phone/rtcpmuxfilter.h" -#include "talk/session/phone/rtputils.h" -#include "talk/session/phone/webrtcmediaengine.h" - -namespace cricket { - -struct PacketMessageData : public talk_base::MessageData { - talk_base::Buffer packet; -}; - -struct VoiceChannelErrorMessageData : public talk_base::MessageData { - VoiceChannelErrorMessageData(uint32 in_ssrc, - VoiceMediaChannel::Error in_error) - : ssrc(in_ssrc), - error(in_error) {} - uint32 ssrc; - VoiceMediaChannel::Error error; -}; - -struct VideoChannelErrorMessageData : public talk_base::MessageData { - VideoChannelErrorMessageData(uint32 in_ssrc, - VideoMediaChannel::Error in_error) - : ssrc(in_ssrc), - error(in_error) {} - uint32 ssrc; - VideoMediaChannel::Error error; -}; - -static const char* PacketType(bool rtcp) { - return (!rtcp) ? "RTP" : "RTCP"; -} - -static bool ValidPacket(bool rtcp, const talk_base::Buffer* packet) { - // Check the packet size. We could check the header too if needed. - return (packet && - packet->length() >= (!rtcp ? kMinRtpPacketLen : kMinRtcpPacketLen) && - packet->length() <= kMaxRtpPacketLen); -} - -BaseChannel::BaseChannel(talk_base::Thread* thread, - MediaEngineInterface* media_engine, - MediaChannel* media_channel, BaseSession* session, - const std::string& content_name, bool rtcp) - : worker_thread_(thread), - media_engine_(media_engine), - session_(session), - media_channel_(media_channel), - content_name_(content_name), - rtcp_(rtcp), - transport_channel_(NULL), - rtcp_transport_channel_(NULL), - enabled_(false), - writable_(false), - was_ever_writable_(false), - has_local_content_(false), - has_remote_content_(false), - muted_(false) { - ASSERT(worker_thread_ == talk_base::Thread::Current()); - LOG(LS_INFO) << "Created channel"; -} - -BaseChannel::~BaseChannel() { - ASSERT(worker_thread_ == talk_base::Thread::Current()); - StopConnectionMonitor(); - FlushRtcpMessages(); // Send any outstanding RTCP packets. - Clear(); // eats any outstanding messages or packets - // We must destroy the media channel before the transport channel, otherwise - // the media channel may try to send on the dead transport channel. NULLing - // is not an effective strategy since the sends will come on another thread. - delete media_channel_; - set_rtcp_transport_channel(NULL); - if (transport_channel_ != NULL) - session_->DestroyChannel(content_name_, transport_channel_->name()); - LOG(LS_INFO) << "Destroyed channel"; -} - -bool BaseChannel::Init(TransportChannel* transport_channel, - TransportChannel* rtcp_transport_channel) { - if (transport_channel == NULL) { - return false; - } - if (rtcp() && rtcp_transport_channel == NULL) { - return false; - } - transport_channel_ = transport_channel; - media_channel_->SetInterface(this); - transport_channel_->SignalWritableState.connect( - this, &BaseChannel::OnWritableState); - transport_channel_->SignalReadPacket.connect( - this, &BaseChannel::OnChannelRead); - - session_->SignalState.connect(this, &BaseChannel::OnSessionState); - session_->SignalRemoteDescriptionUpdate.connect(this, - &BaseChannel::OnRemoteDescriptionUpdate); - - OnSessionState(session(), session()->state()); - set_rtcp_transport_channel(rtcp_transport_channel); - return true; -} - -bool BaseChannel::Enable(bool enable) { - // Can be called from thread other than worker thread - Send(enable ? MSG_ENABLE : MSG_DISABLE); - return true; -} - -bool BaseChannel::Mute(bool mute) { - // Can be called from thread other than worker thread - Send(mute ? MSG_MUTE : MSG_UNMUTE); - return true; -} - -bool BaseChannel::RemoveStream(uint32 ssrc) { - StreamMessageData data(ssrc, 0); - Send(MSG_REMOVESTREAM, &data); - return true; -} - -bool BaseChannel::SetRtcpCName(const std::string& cname) { - SetRtcpCNameData data(cname); - Send(MSG_SETRTCPCNAME, &data); - return data.result; -} - -bool BaseChannel::SetLocalContent(const MediaContentDescription* content, - ContentAction action) { - SetContentData data(content, action); - Send(MSG_SETLOCALCONTENT, &data); - return data.result; -} - -bool BaseChannel::SetRemoteContent(const MediaContentDescription* content, - ContentAction action) { - SetContentData data(content, action); - Send(MSG_SETREMOTECONTENT, &data); - return data.result; -} - -bool BaseChannel::SetMaxSendBandwidth(int max_bandwidth) { - SetBandwidthData data(max_bandwidth); - Send(MSG_SETMAXSENDBANDWIDTH, &data); - return data.result; -} - -void BaseChannel::StartConnectionMonitor(int cms) { - socket_monitor_.reset(new SocketMonitor(transport_channel_, - worker_thread(), - talk_base::Thread::Current())); - socket_monitor_->SignalUpdate.connect( - this, &BaseChannel::OnConnectionMonitorUpdate); - socket_monitor_->Start(cms); -} - -void BaseChannel::StopConnectionMonitor() { - if (socket_monitor_.get()) { - socket_monitor_->Stop(); - socket_monitor_.reset(); - } -} - -void BaseChannel::set_rtcp_transport_channel(TransportChannel* channel) { - if (rtcp_transport_channel_ != channel) { - if (rtcp_transport_channel_) { - session_->DestroyChannel(content_name_, rtcp_transport_channel_->name()); - } - rtcp_transport_channel_ = channel; - if (rtcp_transport_channel_) { - rtcp_transport_channel_->SignalWritableState.connect( - this, &BaseChannel::OnWritableState); - rtcp_transport_channel_->SignalReadPacket.connect( - this, &BaseChannel::OnChannelRead); - } - } -} - -bool BaseChannel::SendPacket(talk_base::Buffer* packet) { - return SendPacket(false, packet); -} - -bool BaseChannel::SendRtcp(talk_base::Buffer* packet) { - return SendPacket(true, packet); -} - -int BaseChannel::SetOption(SocketType type, talk_base::Socket::Option opt, - int value) { - switch (type) { - case ST_RTP: return transport_channel_->SetOption(opt, value); - case ST_RTCP: return rtcp_transport_channel_->SetOption(opt, value); - default: return -1; - } -} - -void BaseChannel::OnWritableState(TransportChannel* channel) { - ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_); - if (transport_channel_->writable() - && (!rtcp_transport_channel_ || rtcp_transport_channel_->writable())) { - ChannelWritable_w(); - } else { - ChannelNotWritable_w(); - } -} - -void BaseChannel::OnChannelRead(TransportChannel* channel, - const char* data, size_t len) { - // OnChannelRead gets called from P2PSocket; now pass data to MediaEngine - ASSERT(worker_thread_ == talk_base::Thread::Current()); - - // When using RTCP multiplexing we might get RTCP packets on the RTP - // transport. We feed RTP traffic into the demuxer to determine if it is RTCP. - bool rtcp = PacketIsRtcp(channel, data, len); - talk_base::Buffer packet(data, len); - HandlePacket(rtcp, &packet); -} - -bool BaseChannel::PacketIsRtcp(const TransportChannel* channel, - const char* data, size_t len) { - return (channel == rtcp_transport_channel_ || - rtcp_mux_filter_.DemuxRtcp(data, len)); -} - -bool BaseChannel::SendPacket(bool rtcp, talk_base::Buffer* packet) { - // Ensure we have a path capable of sending packets. - if (!writable_) { - return false; - } - - // SendPacket gets called from MediaEngine, typically on an encoder thread. - // If the thread is not our worker thread, we will post to our worker - // so that the real work happens on our worker. This avoids us having to - // synchronize access to all the pieces of the send path, including - // SRTP and the inner workings of the transport channels. - // The only downside is that we can't return a proper failure code if - // needed. Since UDP is unreliable anyway, this should be a non-issue. - if (talk_base::Thread::Current() != worker_thread_) { - // Avoid a copy by transferring the ownership of the packet data. - int message_id = (!rtcp) ? MSG_RTPPACKET : MSG_RTCPPACKET; - PacketMessageData* data = new PacketMessageData; - packet->TransferTo(&data->packet); - worker_thread_->Post(this, message_id, data); - return true; - } - - // Now that we are on the correct thread, ensure we have a place to send this - // packet before doing anything. (We might get RTCP packets that we don't - // intend to send.) If we've negotiated RTCP mux, send RTCP over the RTP - // transport. - TransportChannel* channel = (!rtcp || rtcp_mux_filter_.IsActive()) ? - transport_channel_ : rtcp_transport_channel_; - if (!channel || !channel->writable()) { - return false; - } - - // Protect ourselves against crazy data. - if (!ValidPacket(rtcp, packet)) { - LOG(LS_ERROR) << "Dropping outgoing " << content_name_ << " " - << PacketType(rtcp) << " packet: wrong size=" - << packet->length(); - return false; - } - - // Protect if needed. - if (srtp_filter_.IsActive()) { - bool res; - char* data = packet->data(); - int len = packet->length(); - if (!rtcp) { - res = srtp_filter_.ProtectRtp(data, len, packet->capacity(), &len); - if (!res) { - int seq_num = -1; - uint32 ssrc = 0; - GetRtpSeqNum(data, len, &seq_num); - GetRtpSsrc(data, len, &ssrc); - LOG(LS_ERROR) << "Failed to protect " << content_name_ - << " RTP packet: size=" << len - << ", seqnum=" << seq_num << ", SSRC=" << ssrc; - return false; - } - } else { - res = srtp_filter_.ProtectRtcp(data, len, packet->capacity(), &len); - if (!res) { - int type = -1; - GetRtcpType(data, len, &type); - LOG(LS_ERROR) << "Failed to protect " << content_name_ - << " RTCP packet: size=" << len << ", type=" << type; - return false; - } - } - - // Update the length of the packet now that we've added the auth tag. - packet->SetLength(len); - } - - // Signal to the media sink after protecting the packet. TODO: - // Separate APIs to record unprotected media and protected header. - { - talk_base::CritScope cs(&signal_send_packet_cs_); - SignalSendPacket(packet->data(), packet->length(), rtcp); - } - - // Bon voyage. - return (channel->SendPacket(packet->data(), packet->length()) - == static_cast(packet->length())); -} - -void BaseChannel::HandlePacket(bool rtcp, talk_base::Buffer* packet) { - // Protect ourselvs against crazy data. - if (!ValidPacket(rtcp, packet)) { - LOG(LS_ERROR) << "Dropping incoming " << content_name_ << " " - << PacketType(rtcp) << " packet: wrong size=" - << packet->length(); - return; - } - - // Signal to the media sink before unprotecting the packet. TODO: - // Separate APIs to record unprotected media and protected header. - { - talk_base::CritScope cs(&signal_recv_packet_cs_); - SignalRecvPacket(packet->data(), packet->length(), rtcp); - } - - // Unprotect the packet, if needed. - if (srtp_filter_.IsActive()) { - char* data = packet->data(); - int len = packet->length(); - bool res; - if (!rtcp) { - res = srtp_filter_.UnprotectRtp(data, len, &len); - if (!res) { - int seq_num = -1; - uint32 ssrc = 0; - GetRtpSeqNum(data, len, &seq_num); - GetRtpSsrc(data, len, &ssrc); - LOG(LS_ERROR) << "Failed to unprotect " << content_name_ - << " RTP packet: size=" << len - << ", seqnum=" << seq_num << ", SSRC=" << ssrc; - return; - } - } else { - res = srtp_filter_.UnprotectRtcp(data, len, &len); - if (!res) { - int type = -1; - GetRtcpType(data, len, &type); - LOG(LS_ERROR) << "Failed to unprotect " << content_name_ - << " RTCP packet: size=" << len << ", type=" << type; - return; - } - } - - packet->SetLength(len); - } - - // Push it down to the media channel. - if (!rtcp) { - media_channel_->OnPacketReceived(packet); - } else { - media_channel_->OnRtcpReceived(packet); - } -} - -void BaseChannel::OnSessionState(BaseSession* session, - BaseSession::State state) { - const MediaContentDescription* content = NULL; - switch (state) { - case Session::STATE_SENTINITIATE: - content = GetFirstContent(session->local_description()); - if (content && !SetLocalContent(content, CA_OFFER)) { - LOG(LS_ERROR) << "Failure in SetLocalContent with CA_OFFER"; - session->SetError(BaseSession::ERROR_CONTENT); - } - break; - case Session::STATE_SENTACCEPT: - content = GetFirstContent(session->local_description()); - if (content && !SetLocalContent(content, CA_ANSWER)) { - LOG(LS_ERROR) << "Failure in SetLocalContent with CA_ANSWER"; - session->SetError(BaseSession::ERROR_CONTENT); - } - break; - case Session::STATE_RECEIVEDINITIATE: - content = GetFirstContent(session->remote_description()); - if (content && !SetRemoteContent(content, CA_OFFER)) { - LOG(LS_ERROR) << "Failure in SetRemoteContent with CA_OFFER"; - session->SetError(BaseSession::ERROR_CONTENT); - } - break; - case Session::STATE_RECEIVEDACCEPT: - content = GetFirstContent(session->remote_description()); - if (content && !SetRemoteContent(content, CA_ANSWER)) { - LOG(LS_ERROR) << "Failure in SetRemoteContent with CA_ANSWER"; - session->SetError(BaseSession::ERROR_CONTENT); - } - break; - default: - break; - } -} - -void BaseChannel::OnRemoteDescriptionUpdate(BaseSession* session) { - const MediaContentDescription* content = - GetFirstContent(session->remote_description()); - - if (content && !SetRemoteContent(content, CA_UPDATE)) { - LOG(LS_ERROR) << "Failure in SetRemoteContent with CA_UPDATE"; - session->SetError(BaseSession::ERROR_CONTENT); - } -} - -void BaseChannel::EnableMedia_w() { - ASSERT(worker_thread_ == talk_base::Thread::Current()); - if (enabled_) - return; - - LOG(LS_INFO) << "Channel enabled"; - enabled_ = true; - ChangeState(); -} - -void BaseChannel::DisableMedia_w() { - ASSERT(worker_thread_ == talk_base::Thread::Current()); - if (!enabled_) - return; - - LOG(LS_INFO) << "Channel disabled"; - enabled_ = false; - ChangeState(); -} - -void BaseChannel::MuteMedia_w() { - ASSERT(worker_thread_ == talk_base::Thread::Current()); - if (muted_) - return; - - if (media_channel()->Mute(true)) { - LOG(LS_INFO) << "Channel muted"; - muted_ = true; - } -} - -void BaseChannel::UnmuteMedia_w() { - ASSERT(worker_thread_ == talk_base::Thread::Current()); - if (!muted_) - return; - - if (media_channel()->Mute(false)) { - LOG(LS_INFO) << "Channel unmuted"; - muted_ = false; - } -} - -void BaseChannel::ChannelWritable_w() { - ASSERT(worker_thread_ == talk_base::Thread::Current()); - if (writable_) - return; - LOG(LS_INFO) << "Channel socket writable (" - << transport_channel_->name().c_str() << ")" - << (was_ever_writable_ ? "" : " for the first time"); - was_ever_writable_ = true; - writable_ = true; - ChangeState(); -} - -void BaseChannel::ChannelNotWritable_w() { - ASSERT(worker_thread_ == talk_base::Thread::Current()); - if (!writable_) - return; - - LOG(LS_INFO) << "Channel socket not writable (" - << transport_channel_->name().c_str() << ")"; - writable_ = false; - ChangeState(); -} - -// Sets the maximum video bandwidth for automatic bandwidth adjustment. -bool BaseChannel::SetMaxSendBandwidth_w(int max_bandwidth) { - return media_channel()->SetSendBandwidth(true, max_bandwidth); -} - -bool BaseChannel::SetRtcpCName_w(const std::string& cname) { - return media_channel()->SetRtcpCName(cname); -} - -bool BaseChannel::SetSrtp_w(const std::vector& cryptos, - ContentAction action, ContentSource src) { - bool ret; - if (action == CA_OFFER) { - ret = srtp_filter_.SetOffer(cryptos, src); - } else if (action == CA_ANSWER) { - ret = srtp_filter_.SetAnswer(cryptos, src); - } else { - // CA_UPDATE, no crypto params. - ret = true; - } - return ret; -} - -bool BaseChannel::SetRtcpMux_w(bool enable, ContentAction action, - ContentSource src) { - bool ret; - if (action == CA_OFFER) { - ret = rtcp_mux_filter_.SetOffer(enable, src); - } else if (action == CA_ANSWER) { - ret = rtcp_mux_filter_.SetAnswer(enable, src); - if (ret && rtcp_mux_filter_.IsActive()) { - // We activated RTCP mux, close down the RTCP transport. - set_rtcp_transport_channel(NULL); - // If the RTP transport is already writable, then so are we. - if (transport_channel_->writable()) { - ChannelWritable_w(); - } - } - } else { - // CA_UPDATE, no RTCP mux info. - ret = true; - } - return ret; -} - -void BaseChannel::OnMessage(talk_base::Message *pmsg) { - switch (pmsg->message_id) { - case MSG_ENABLE: - EnableMedia_w(); - break; - case MSG_DISABLE: - DisableMedia_w(); - break; - - case MSG_MUTE: - MuteMedia_w(); - break; - case MSG_UNMUTE: - UnmuteMedia_w(); - break; - - case MSG_SETRTCPCNAME: { - SetRtcpCNameData* data = static_cast(pmsg->pdata); - data->result = SetRtcpCName_w(data->cname); - break; - } - - case MSG_SETLOCALCONTENT: { - SetContentData* data = static_cast(pmsg->pdata); - data->result = SetLocalContent_w(data->content, data->action); - break; - } - case MSG_SETREMOTECONTENT: { - SetContentData* data = static_cast(pmsg->pdata); - data->result = SetRemoteContent_w(data->content, data->action); - break; - } - - case MSG_REMOVESTREAM: { - StreamMessageData* data = static_cast(pmsg->pdata); - RemoveStream_w(data->ssrc1); - break; - } - - case MSG_SETMAXSENDBANDWIDTH: { - SetBandwidthData* data = static_cast(pmsg->pdata); - data->result = SetMaxSendBandwidth_w(data->value); - break; - } - - case MSG_RTPPACKET: - case MSG_RTCPPACKET: { - PacketMessageData* data = static_cast(pmsg->pdata); - SendPacket(pmsg->message_id == MSG_RTCPPACKET, &data->packet); - delete data; // because it is Posted - break; - } - } -} - -void BaseChannel::Send(uint32 id, talk_base::MessageData *pdata) { - worker_thread_->Send(this, id, pdata); -} - -void BaseChannel::Post(uint32 id, talk_base::MessageData *pdata) { - worker_thread_->Post(this, id, pdata); -} - -void BaseChannel::PostDelayed(int cmsDelay, uint32 id, - talk_base::MessageData *pdata) { - worker_thread_->PostDelayed(cmsDelay, this, id, pdata); -} - -void BaseChannel::Clear(uint32 id, talk_base::MessageList* removed) { - worker_thread_->Clear(this, id, removed); -} - -void BaseChannel::FlushRtcpMessages() { - // Flush all remaining RTCP messages. This should only be called in - // destructor. - ASSERT(talk_base::Thread::Current() == worker_thread_); - talk_base::MessageList rtcp_messages; - Clear(MSG_RTCPPACKET, &rtcp_messages); - for (talk_base::MessageList::iterator it = rtcp_messages.begin(); - it != rtcp_messages.end(); ++it) { - Send(MSG_RTCPPACKET, it->pdata); - } -} - -VoiceChannel::VoiceChannel(talk_base::Thread* thread, - MediaEngineInterface* media_engine, - VoiceMediaChannel* media_channel, - BaseSession* session, - const std::string& content_name, - bool rtcp) - : BaseChannel(thread, media_engine, media_channel, session, content_name, - rtcp), - received_media_(false) { -} - -VoiceChannel::~VoiceChannel() { - StopAudioMonitor(); - StopMediaMonitor(); - // this can't be done in the base class, since it calls a virtual - DisableMedia_w(); -} - -bool VoiceChannel::Init() { - TransportChannel* rtcp_channel = rtcp() ? - session()->CreateChannel(content_name(), "rtcp") : NULL; - if (!BaseChannel::Init(session()->CreateChannel(content_name(), "rtp"), - rtcp_channel)) { - return false; - } - media_channel()->SignalMediaError.connect( - this, &VoiceChannel::OnVoiceChannelError); - srtp_filter()->SignalSrtpError.connect( - this, &VoiceChannel::OnSrtpError); - return true; -} - -bool VoiceChannel::AddStream(uint32 ssrc) { - StreamMessageData data(ssrc, 0); - Send(MSG_ADDSTREAM, &data); - return true; -} - -bool VoiceChannel::SetRingbackTone(const void* buf, int len) { - SetRingbackToneMessageData data(buf, len); - Send(MSG_SETRINGBACKTONE, &data); - return data.result; -} - -// TODO: Handle early media the right way. We should get an explicit -// ringing message telling us to start playing local ringback, which we cancel -// if any early media actually arrives. For now, we do the opposite, which is -// to wait 1 second for early media, and start playing local ringback if none -// arrives. -void VoiceChannel::SetEarlyMedia(bool enable) { - if (enable) { - // Start the early media timeout - PostDelayed(kEarlyMediaTimeout, MSG_EARLYMEDIATIMEOUT); - } else { - // Stop the timeout if currently going. - Clear(MSG_EARLYMEDIATIMEOUT); - } -} - -bool VoiceChannel::PlayRingbackTone(uint32 ssrc, bool play, bool loop) { - PlayRingbackToneMessageData data(ssrc, play, loop); - Send(MSG_PLAYRINGBACKTONE, &data); - return data.result; -} - -bool VoiceChannel::PressDTMF(int digit, bool playout) { - DtmfMessageData data(digit, playout); - Send(MSG_PRESSDTMF, &data); - return data.result; -} - -bool VoiceChannel::SetOutputScaling(uint32 ssrc, double left, double right) { - ScaleVolumeMessageData data(ssrc, left, right); - Send(MSG_SCALEVOLUME, &data); - return data.result; -} - -void VoiceChannel::StartMediaMonitor(int cms) { - media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(), - talk_base::Thread::Current())); - media_monitor_->SignalUpdate.connect( - this, &VoiceChannel::OnMediaMonitorUpdate); - media_monitor_->Start(cms); -} - -void VoiceChannel::StopMediaMonitor() { - if (media_monitor_.get()) { - media_monitor_->Stop(); - media_monitor_->SignalUpdate.disconnect(this); - media_monitor_.reset(); - } -} - -void VoiceChannel::StartAudioMonitor(int cms) { - audio_monitor_.reset(new AudioMonitor(this, talk_base::Thread::Current())); - audio_monitor_ - ->SignalUpdate.connect(this, &VoiceChannel::OnAudioMonitorUpdate); - audio_monitor_->Start(cms); -} - -void VoiceChannel::StopAudioMonitor() { - if (audio_monitor_.get()) { - audio_monitor_->Stop(); - audio_monitor_.reset(); - } -} - -bool VoiceChannel::IsAudioMonitorRunning() const { - return (audio_monitor_.get() != NULL); -} - -int VoiceChannel::GetInputLevel_w() { - return media_engine()->GetInputLevel(); -} - -int VoiceChannel::GetOutputLevel_w() { - return media_channel()->GetOutputLevel(); -} - -void VoiceChannel::GetActiveStreams_w(AudioInfo::StreamList* actives) { - media_channel()->GetActiveStreams(actives); -} - -void VoiceChannel::OnChannelRead(TransportChannel* channel, - const char* data, size_t len) { - BaseChannel::OnChannelRead(channel, data, len); - - // Set a flag when we've received an RTP packet. If we're waiting for early - // media, this will disable the timeout. - if (!received_media_ && !PacketIsRtcp(channel, data, len)) { - received_media_ = true; - } -} - -void VoiceChannel::ChangeState() { - // Render incoming data if we're the active call, and we have the local - // content. We receive data on the default channel and multiplexed streams. - bool recv = enabled() && has_local_content(); - if (!media_channel()->SetPlayout(recv)) { - SendLastMediaError(); - } - - // Send outgoing data if we're the active call, we have the remote content, - // and we have had some form of connectivity. - bool send = enabled() && has_remote_content() && was_ever_writable(); - SendFlags send_flag = send ? SEND_MICROPHONE : SEND_NOTHING; - if (!media_channel()->SetSend(send_flag)) { - LOG(LS_ERROR) << "Failed to SetSend " << send_flag << " on voice channel"; - SendLastMediaError(); - } - - LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send; -} - -const MediaContentDescription* VoiceChannel::GetFirstContent( - const SessionDescription* sdesc) { - const ContentInfo* cinfo = GetFirstAudioContent(sdesc); - if (cinfo == NULL) - return NULL; - - return static_cast(cinfo->description); -} - -bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content, - ContentAction action) { - ASSERT(worker_thread() == talk_base::Thread::Current()); - LOG(LS_INFO) << "Setting local voice description"; - - const AudioContentDescription* audio = - static_cast(content); - ASSERT(audio != NULL); - - bool ret; - if (audio->ssrc_set()) { - media_channel()->SetSendSsrc(audio->ssrc()); - LOG(LS_INFO) << "Set send ssrc for audio: " << audio->ssrc(); - } - // set SRTP - ret = SetSrtp_w(audio->cryptos(), action, CS_LOCAL); - // set RTCP mux - if (ret) { - ret = SetRtcpMux_w(audio->rtcp_mux(), action, CS_LOCAL); - } - // set payload type and config for voice codecs - if (ret) { - ret = media_channel()->SetRecvCodecs(audio->codecs()); - } - // set header extensions - if (ret && audio->rtp_header_extensions_set()) { - ret = media_channel()->SetRecvRtpHeaderExtensions( - audio->rtp_header_extensions()); - } - if (ret) { - set_has_local_content(true); - ChangeState(); - } else { - LOG(LS_WARNING) << "Failed to set local voice description"; - } - return ret; -} - -bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content, - ContentAction action) { - ASSERT(worker_thread() == talk_base::Thread::Current()); - LOG(LS_INFO) << "Setting remote voice description"; - - const AudioContentDescription* audio = - static_cast(content); - ASSERT(audio != NULL); - - bool ret; - // set SRTP - ret = SetSrtp_w(audio->cryptos(), action, CS_REMOTE); - // set RTCP mux - if (ret) { - ret = SetRtcpMux_w(audio->rtcp_mux(), action, CS_REMOTE); - } - // set codecs and payload types - if (ret) { - ret = media_channel()->SetSendCodecs(audio->codecs()); - } - // set header extensions - if (ret && audio->rtp_header_extensions_set()) { - ret = media_channel()->SetSendRtpHeaderExtensions( - audio->rtp_header_extensions()); - } - - int audio_options = 0; - if (audio->conference_mode()) { - audio_options |= OPT_CONFERENCE; - } - if (!media_channel()->SetOptions(audio_options)) { - // Log an error on failure, but don't abort the call. - LOG(LS_ERROR) << "Failed to set voice channel options"; - } - - // update state - if (ret) { - set_has_remote_content(true); - ChangeState(); - } else { - LOG(LS_WARNING) << "Failed to set remote voice description"; - } - return ret; -} - -void VoiceChannel::AddStream_w(uint32 ssrc) { - ASSERT(worker_thread() == talk_base::Thread::Current()); - media_channel()->AddStream(ssrc); -} - -void VoiceChannel::RemoveStream_w(uint32 ssrc) { - media_channel()->RemoveStream(ssrc); -} - -bool VoiceChannel::SetRingbackTone_w(const void* buf, int len) { - ASSERT(worker_thread() == talk_base::Thread::Current()); - return media_channel()->SetRingbackTone(static_cast(buf), len); -} - -bool VoiceChannel::PlayRingbackTone_w(uint32 ssrc, bool play, bool loop) { - ASSERT(worker_thread() == talk_base::Thread::Current()); - if (play) { - LOG(LS_INFO) << "Playing ringback tone, loop=" << loop; - } else { - LOG(LS_INFO) << "Stopping ringback tone"; - } - return media_channel()->PlayRingbackTone(ssrc, play, loop); -} - -void VoiceChannel::HandleEarlyMediaTimeout() { - // This occurs on the main thread, not the worker thread. - if (!received_media_) { - LOG(LS_INFO) << "No early media received before timeout"; - SignalEarlyMediaTimeout(this); - } -} - -bool VoiceChannel::PressDTMF_w(int digit, bool playout) { - if (!enabled() || !writable()) { - return false; - } - - return media_channel()->PressDTMF(digit, playout); -} - -bool VoiceChannel::SetOutputScaling_w(uint32 ssrc, double left, double right) { - return media_channel()->SetOutputScaling(ssrc, left, right); -} - -void VoiceChannel::OnMessage(talk_base::Message *pmsg) { - switch (pmsg->message_id) { - case MSG_ADDSTREAM: { - StreamMessageData* data = static_cast(pmsg->pdata); - AddStream_w(data->ssrc1); - break; - } - case MSG_SETRINGBACKTONE: { - SetRingbackToneMessageData* data = - static_cast(pmsg->pdata); - data->result = SetRingbackTone_w(data->buf, data->len); - break; - } - case MSG_PLAYRINGBACKTONE: { - PlayRingbackToneMessageData* data = - static_cast(pmsg->pdata); - data->result = PlayRingbackTone_w(data->ssrc, data->play, data->loop); - break; - } - case MSG_EARLYMEDIATIMEOUT: - HandleEarlyMediaTimeout(); - break; - case MSG_PRESSDTMF: { - DtmfMessageData* data = static_cast(pmsg->pdata); - data->result = PressDTMF_w(data->digit, data->playout); - break; - } - case MSG_SCALEVOLUME: { - ScaleVolumeMessageData* data = - static_cast(pmsg->pdata); - data->result = SetOutputScaling_w(data->ssrc, data->left, data->right); - break; - } - case MSG_CHANNEL_ERROR: { - VoiceChannelErrorMessageData* data = - static_cast(pmsg->pdata); - SignalMediaError(this, data->ssrc, data->error); - delete data; - break; - } - - default: - BaseChannel::OnMessage(pmsg); - break; - } -} - -void VoiceChannel::OnConnectionMonitorUpdate( - SocketMonitor* monitor, const std::vector& infos) { - SignalConnectionMonitor(this, infos); -} - -void VoiceChannel::OnMediaMonitorUpdate( - VoiceMediaChannel* media_channel, const VoiceMediaInfo& info) { - ASSERT(media_channel == this->media_channel()); - SignalMediaMonitor(this, info); -} - -void VoiceChannel::OnAudioMonitorUpdate(AudioMonitor* monitor, - const AudioInfo& info) { - SignalAudioMonitor(this, info); -} - -void VoiceChannel::OnVoiceChannelError( - uint32 ssrc, VoiceMediaChannel::Error error) { - VoiceChannelErrorMessageData *data = new VoiceChannelErrorMessageData( - ssrc, error); - signaling_thread()->Post(this, MSG_CHANNEL_ERROR, data); -} - -void VoiceChannel::OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, - SrtpFilter::Error error) { - switch (error) { - case SrtpFilter::ERROR_FAIL: - OnVoiceChannelError(ssrc, (mode == SrtpFilter::PROTECT) ? - VoiceMediaChannel::ERROR_REC_SRTP_ERROR : - VoiceMediaChannel::ERROR_PLAY_SRTP_ERROR); - break; - case SrtpFilter::ERROR_AUTH: - OnVoiceChannelError(ssrc, (mode == SrtpFilter::PROTECT) ? - VoiceMediaChannel::ERROR_REC_SRTP_AUTH_FAILED : - VoiceMediaChannel::ERROR_PLAY_SRTP_AUTH_FAILED); - break; - case SrtpFilter::ERROR_REPLAY: - // Only receving channel should have this error. - ASSERT(mode == SrtpFilter::UNPROTECT); - OnVoiceChannelError(ssrc, VoiceMediaChannel::ERROR_PLAY_SRTP_REPLAY); - break; - default: - break; - } -} - -VideoChannel::VideoChannel(talk_base::Thread* thread, - MediaEngineInterface* media_engine, - VideoMediaChannel* media_channel, - BaseSession* session, - const std::string& content_name, - bool rtcp, - VoiceChannel* voice_channel) - : BaseChannel(thread, media_engine, media_channel, session, content_name, - rtcp), - voice_channel_(voice_channel), renderer_(NULL) { -} - -bool VideoChannel::Init() { - TransportChannel* rtcp_channel = rtcp() ? - session()->CreateChannel(content_name(), "video_rtcp") : NULL; - if (!BaseChannel::Init( - session()->CreateChannel(content_name(), "video_rtp"), - rtcp_channel)) { - return false; - } - media_channel()->SignalMediaError.connect( - this, &VideoChannel::OnVideoChannelError); - srtp_filter()->SignalSrtpError.connect( - this, &VideoChannel::OnSrtpError); - return true; -} - -void VoiceChannel::SendLastMediaError() { - uint32 ssrc; - VoiceMediaChannel::Error error; - media_channel()->GetLastMediaError(&ssrc, &error); - SignalMediaError(this, ssrc, error); -} - -VideoChannel::~VideoChannel() { - StopMediaMonitor(); - // this can't be done in the base class, since it calls a virtual - DisableMedia_w(); -} - -bool VideoChannel::AddStream(uint32 ssrc, uint32 voice_ssrc) { - StreamMessageData data(ssrc, voice_ssrc); - Send(MSG_ADDSTREAM, &data); - return true; -} - -bool VideoChannel::SetRenderer(uint32 ssrc, VideoRenderer* renderer) { - RenderMessageData data(ssrc, renderer); - Send(MSG_SETRENDERER, &data); - return true; -} - - - -bool VideoChannel::SendIntraFrame() { - Send(MSG_SENDINTRAFRAME); - return true; -} - -bool VideoChannel::RequestIntraFrame() { - Send(MSG_REQUESTINTRAFRAME); - return true; -} - -void VideoChannel::EnableCpuAdaptation(bool enable) { - Send(enable ? MSG_ENABLECPUADAPTATION : MSG_DISABLECPUADAPTATION); -} - -void VideoChannel::ChangeState() { - // Render incoming data if we're the active call, and we have the local - // content. We receive data on the default channel and multiplexed streams. - bool recv = enabled() && has_local_content(); - if (!media_channel()->SetRender(recv)) { - LOG(LS_ERROR) << "Failed to SetRender on video channel"; - // TODO: Report error back to server. - } - - // Send outgoing data if we're the active call, we have the remote content, - // and we have had some form of connectivity. - bool send = enabled() && has_remote_content() && was_ever_writable(); - if (!media_channel()->SetSend(send)) { - LOG(LS_ERROR) << "Failed to SetSend on video channel"; - // TODO: Report error back to server. - } - - LOG(LS_INFO) << "Changing video state, recv=" << recv << " send=" << send; -} - -void VideoChannel::StartMediaMonitor(int cms) { - media_monitor_.reset(new VideoMediaMonitor(media_channel(), worker_thread(), - talk_base::Thread::Current())); - media_monitor_->SignalUpdate.connect( - this, &VideoChannel::OnMediaMonitorUpdate); - media_monitor_->Start(cms); -} - -void VideoChannel::StopMediaMonitor() { - if (media_monitor_.get()) { - media_monitor_->Stop(); - media_monitor_.reset(); - } -} - -const MediaContentDescription* VideoChannel::GetFirstContent( - const SessionDescription* sdesc) { - const ContentInfo* cinfo = GetFirstVideoContent(sdesc); - if (cinfo == NULL) - return NULL; - - return static_cast(cinfo->description); -} - -bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content, - ContentAction action) { - ASSERT(worker_thread() == talk_base::Thread::Current()); - LOG(LS_INFO) << "Setting local video description"; - - const VideoContentDescription* video = - static_cast(content); - ASSERT(video != NULL); - - bool ret; - if (video->ssrc_set()) { - media_channel()->SetSendSsrc(video->ssrc()); - LOG(LS_INFO) << "Set send ssrc for video: " << video->ssrc(); - } - // set SRTP - ret = SetSrtp_w(video->cryptos(), action, CS_LOCAL); - // set RTCP mux - if (ret) { - ret = SetRtcpMux_w(video->rtcp_mux(), action, CS_LOCAL); - } - // set payload types and config for receiving video - if (ret) { - ret = media_channel()->SetRecvCodecs(video->codecs()); - } - if (ret && video->rtp_header_extensions_set()) { - ret = media_channel()->SetRecvRtpHeaderExtensions( - video->rtp_header_extensions()); - } - if (ret) { - set_has_local_content(true); - ChangeState(); - } else { - LOG(LS_WARNING) << "Failed to set local video description"; - } - return ret; -} - -bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content, - ContentAction action) { - ASSERT(worker_thread() == talk_base::Thread::Current()); - LOG(LS_INFO) << "Setting remote video description"; - - const VideoContentDescription* video = - static_cast(content); - ASSERT(video != NULL); - - bool ret; - // set SRTP - ret = SetSrtp_w(video->cryptos(), action, CS_REMOTE); - // set RTCP mux - if (ret) { - ret = SetRtcpMux_w(video->rtcp_mux(), action, CS_REMOTE); - } - // Set video bandwidth parameters. - if (ret) { - int bandwidth_bps = video->bandwidth(); - bool auto_bandwidth = (bandwidth_bps == kAutoBandwidth); - ret = media_channel()->SetSendBandwidth(auto_bandwidth, bandwidth_bps); - } - if (ret) { - ret = media_channel()->SetSendCodecs(video->codecs()); - } - // set header extensions - if (ret && video->rtp_header_extensions_set()) { - ret = media_channel()->SetSendRtpHeaderExtensions( - video->rtp_header_extensions()); - } - if (ret) { - set_has_remote_content(true); - ChangeState(); - } else { - LOG(LS_WARNING) << "Failed to set remote video description"; - } - return ret; -} - -void VideoChannel::AddStream_w(uint32 ssrc, uint32 voice_ssrc) { - media_channel()->AddStream(ssrc, voice_ssrc); -} - -void VideoChannel::RemoveStream_w(uint32 ssrc) { - media_channel()->RemoveStream(ssrc); -} - -void VideoChannel::SetRenderer_w(uint32 ssrc, VideoRenderer* renderer) { - media_channel()->SetRenderer(ssrc, renderer); -} - - -void VideoChannel::OnMessage(talk_base::Message *pmsg) { - switch (pmsg->message_id) { - case MSG_ADDSTREAM: { - StreamMessageData* data = static_cast(pmsg->pdata); - AddStream_w(data->ssrc1, data->ssrc2); - break; - } - case MSG_SETRENDERER: { - RenderMessageData* data = static_cast(pmsg->pdata); - SetRenderer_w(data->ssrc, data->renderer); - break; - } - case MSG_SENDINTRAFRAME: - SendIntraFrame_w(); - break; - case MSG_REQUESTINTRAFRAME: - RequestIntraFrame_w(); - break; - case MSG_ENABLECPUADAPTATION: - EnableCpuAdaptation_w(true); - break; - case MSG_DISABLECPUADAPTATION: - EnableCpuAdaptation_w(false); - break; - case MSG_CHANNEL_ERROR: { - const VideoChannelErrorMessageData* data = - static_cast(pmsg->pdata); - SignalMediaError(this, data->ssrc, data->error); - delete data; - break; - } - default: - BaseChannel::OnMessage(pmsg); - break; - } -} - -void VideoChannel::OnConnectionMonitorUpdate( - SocketMonitor *monitor, const std::vector &infos) { - SignalConnectionMonitor(this, infos); -} - -void VideoChannel::OnMediaMonitorUpdate( - VideoMediaChannel* media_channel, const VideoMediaInfo &info) { - ASSERT(media_channel == this->media_channel()); - SignalMediaMonitor(this, info); -} - - -void VideoChannel::OnVideoChannelError(uint32 ssrc, - VideoMediaChannel::Error error) { - VideoChannelErrorMessageData* data = new VideoChannelErrorMessageData( - ssrc, error); - signaling_thread()->Post(this, MSG_CHANNEL_ERROR, data); -} - -void VideoChannel::OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, - SrtpFilter::Error error) { - switch (error) { - case SrtpFilter::ERROR_FAIL: - OnVideoChannelError(ssrc, (mode == SrtpFilter::PROTECT) ? - VideoMediaChannel::ERROR_REC_SRTP_ERROR : - VideoMediaChannel::ERROR_PLAY_SRTP_ERROR); - break; - case SrtpFilter::ERROR_AUTH: - OnVideoChannelError(ssrc, (mode == SrtpFilter::PROTECT) ? - VideoMediaChannel::ERROR_REC_SRTP_AUTH_FAILED : - VideoMediaChannel::ERROR_PLAY_SRTP_AUTH_FAILED); - break; - case SrtpFilter::ERROR_REPLAY: - // Only receving channel should have this error. - ASSERT(mode == SrtpFilter::UNPROTECT); - // TODO: Turn on the signaling of replay error once we have - // switched to the new mechanism for doing video retransmissions. - // OnVideoChannelError(ssrc, VideoMediaChannel::ERROR_PLAY_SRTP_REPLAY); - break; - default: - break; - } -} - -// TODO(mallinath) - Post on worker thread? -void VideoChannel::SetCaptureDevice( - uint32 ssrc, webrtc::VideoCaptureModule* camera) { - // Ignore SSRC for now - WebRtcMediaEngine* me = - static_cast (media_engine()); - ASSERT(me != NULL); - me->SetVideoCaptureModule(camera); -} - -void VideoChannel::SetLocalRenderer(uint32 ssrc, VideoRenderer* renderer) { - // Ignore SSRC for now as mutliple send streams are not there yet. - media_engine()->SetLocalRenderer(renderer); -} - -} // namespace cricket diff --git a/third_party_mods/libjingle/source/talk/session/phone/channel.h b/third_party_mods/libjingle/source/talk/session/phone/channel.h deleted file mode 100644 index a9ac75391..000000000 --- a/third_party_mods/libjingle/source/talk/session/phone/channel.h +++ /dev/null @@ -1,515 +0,0 @@ -/* - * libjingle - * Copyright 2004--2007, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#ifndef TALK_SESSION_PHONE_CHANNEL_H_ -#define TALK_SESSION_PHONE_CHANNEL_H_ - -#include -#include - -#include "talk/base/asyncudpsocket.h" -#include "talk/base/criticalsection.h" -#include "talk/base/network.h" -#include "talk/base/sigslot.h" -#include "talk/p2p/client/socketmonitor.h" -#include "talk/p2p/base/session.h" -#include "talk/session/phone/audiomonitor.h" -#include "talk/session/phone/mediachannel.h" -#include "talk/session/phone/mediaengine.h" -#include "talk/session/phone/mediamonitor.h" -#include "talk/session/phone/rtcpmuxfilter.h" -#include "talk/session/phone/srtpfilter.h" - -namespace webrtc { -class VideoCaptureModule; -} - -namespace cricket { - -class MediaContentDescription; -struct CryptoParams; - -enum { - MSG_ENABLE = 1, - MSG_DISABLE = 2, - MSG_MUTE = 3, - MSG_UNMUTE = 4, - MSG_SETREMOTECONTENT = 5, - MSG_SETLOCALCONTENT = 6, - MSG_EARLYMEDIATIMEOUT = 8, - MSG_PRESSDTMF = 9, - MSG_SETRENDERER = 10, - MSG_ADDSTREAM = 11, - MSG_REMOVESTREAM = 12, - MSG_SETRINGBACKTONE = 13, - MSG_PLAYRINGBACKTONE = 14, - MSG_SETMAXSENDBANDWIDTH = 15, - MSG_SETRTCPCNAME = 18, - MSG_SENDINTRAFRAME = 19, - MSG_REQUESTINTRAFRAME = 20, - MSG_RTPPACKET = 22, - MSG_RTCPPACKET = 23, - MSG_CHANNEL_ERROR = 24, - MSG_ENABLECPUADAPTATION = 25, - MSG_DISABLECPUADAPTATION = 26, - MSG_SCALEVOLUME = 27 -}; - -// BaseChannel contains logic common to voice and video, including -// enable/mute, marshaling calls to a worker thread, and -// connection and media monitors. -class BaseChannel - : public talk_base::MessageHandler, public sigslot::has_slots<>, - public MediaChannel::NetworkInterface { - public: - BaseChannel(talk_base::Thread* thread, MediaEngineInterface* media_engine, - MediaChannel* channel, BaseSession* session, - const std::string& content_name, bool rtcp); - virtual ~BaseChannel(); - bool Init(TransportChannel* transport_channel, - TransportChannel* rtcp_transport_channel); - - talk_base::Thread* worker_thread() const { return worker_thread_; } - BaseSession* session() const { return session_; } - const std::string& content_name() { return content_name_; } - TransportChannel* transport_channel() const { - return transport_channel_; - } - TransportChannel* rtcp_transport_channel() const { - return rtcp_transport_channel_; - } - bool enabled() const { return enabled_; } - bool secure() const { return srtp_filter_.IsActive(); } - - // Channel control - bool SetRtcpCName(const std::string& cname); - bool SetLocalContent(const MediaContentDescription* content, - ContentAction action); - bool SetRemoteContent(const MediaContentDescription* content, - ContentAction action); - bool SetMaxSendBandwidth(int max_bandwidth); - - bool Enable(bool enable); - bool Mute(bool mute); - - // Multiplexing - bool RemoveStream(uint32 ssrc); - - // Monitoring - void StartConnectionMonitor(int cms); - void StopConnectionMonitor(); - - void set_srtp_signal_silent_time(uint32 silent_time) { - srtp_filter_.set_signal_silent_time(silent_time); - } - - template - void RegisterSendSink(T* sink, - void (T::*OnPacket)(const void*, size_t, bool)) { - talk_base::CritScope cs(&signal_send_packet_cs_); - SignalSendPacket.disconnect(sink); - SignalSendPacket.connect(sink, OnPacket); - } - - void UnregisterSendSink(sigslot::has_slots<>* sink) { - talk_base::CritScope cs(&signal_send_packet_cs_); - SignalSendPacket.disconnect(sink); - } - - bool HasSendSinks() { - talk_base::CritScope cs(&signal_send_packet_cs_); - return !SignalSendPacket.is_empty(); - } - - template - void RegisterRecvSink(T* sink, - void (T::*OnPacket)(const void*, size_t, bool)) { - talk_base::CritScope cs(&signal_recv_packet_cs_); - SignalRecvPacket.disconnect(sink); - SignalRecvPacket.connect(sink, OnPacket); - } - - void UnregisterRecvSink(sigslot::has_slots<>* sink) { - talk_base::CritScope cs(&signal_recv_packet_cs_); - SignalRecvPacket.disconnect(sink); - } - - bool HasRecvSinks() { - talk_base::CritScope cs(&signal_recv_packet_cs_); - return !SignalRecvPacket.is_empty(); - } - - protected: - MediaEngineInterface* media_engine() const { return media_engine_; } - virtual MediaChannel* media_channel() const { return media_channel_; } - void set_rtcp_transport_channel(TransportChannel* transport); - bool writable() const { return writable_; } - bool was_ever_writable() const { return was_ever_writable_; } - bool has_local_content() const { return has_local_content_; } - bool has_remote_content() const { return has_remote_content_; } - void set_has_local_content(bool has) { has_local_content_ = has; } - void set_has_remote_content(bool has) { has_remote_content_ = has; } - bool muted() const { return muted_; } - talk_base::Thread* signaling_thread() { return session_->signaling_thread(); } - SrtpFilter* srtp_filter() { return &srtp_filter_; } - bool rtcp() const { return rtcp_; } - - void Send(uint32 id, talk_base::MessageData *pdata = NULL); - void Post(uint32 id, talk_base::MessageData *pdata = NULL); - void PostDelayed(int cmsDelay, uint32 id = 0, - talk_base::MessageData *pdata = NULL); - void Clear(uint32 id = talk_base::MQID_ANY, - talk_base::MessageList* removed = NULL); - void FlushRtcpMessages(); - - // NetworkInterface implementation, called by MediaEngine - virtual bool SendPacket(talk_base::Buffer* packet); - virtual bool SendRtcp(talk_base::Buffer* packet); - virtual int SetOption(SocketType type, talk_base::Socket::Option o, int val); - - // From TransportChannel - void OnWritableState(TransportChannel* channel); - virtual void OnChannelRead(TransportChannel* channel, const char* data, - size_t len); - - bool PacketIsRtcp(const TransportChannel* channel, const char* data, - size_t len); - bool SendPacket(bool rtcp, talk_base::Buffer* packet); - void HandlePacket(bool rtcp, talk_base::Buffer* packet); - - // Setting the send codec based on the remote description. - void OnSessionState(BaseSession* session, BaseSession::State state); - void OnRemoteDescriptionUpdate(BaseSession* session); - - void EnableMedia_w(); - void DisableMedia_w(); - void MuteMedia_w(); - void UnmuteMedia_w(); - void ChannelWritable_w(); - void ChannelNotWritable_w(); - - struct StreamMessageData : public talk_base::MessageData { - StreamMessageData(uint32 s1, uint32 s2) : ssrc1(s1), ssrc2(s2) {} - uint32 ssrc1; - uint32 ssrc2; - }; - virtual void RemoveStream_w(uint32 ssrc) = 0; - - virtual void ChangeState() = 0; - - struct SetRtcpCNameData : public talk_base::MessageData { - explicit SetRtcpCNameData(const std::string& cname) - : cname(cname), result(false) {} - std::string cname; - bool result; - }; - bool SetRtcpCName_w(const std::string& cname); - - struct SetContentData : public talk_base::MessageData { - SetContentData(const MediaContentDescription* content, - ContentAction action) - : content(content), action(action), result(false) {} - const MediaContentDescription* content; - ContentAction action; - bool result; - }; - - // Gets the content appropriate to the channel (audio or video). - virtual const MediaContentDescription* GetFirstContent( - const SessionDescription* sdesc) = 0; - virtual bool SetLocalContent_w(const MediaContentDescription* content, - ContentAction action) = 0; - virtual bool SetRemoteContent_w(const MediaContentDescription* content, - ContentAction action) = 0; - - bool SetSrtp_w(const std::vector& params, ContentAction action, - ContentSource src); - bool SetRtcpMux_w(bool enable, ContentAction action, ContentSource src); - - struct SetBandwidthData : public talk_base::MessageData { - explicit SetBandwidthData(int value) : value(value), result(false) {} - int value; - bool result; - }; - bool SetMaxSendBandwidth_w(int max_bandwidth); - - // From MessageHandler - virtual void OnMessage(talk_base::Message *pmsg); - - // Handled in derived classes - virtual void OnConnectionMonitorUpdate(SocketMonitor *monitor, - const std::vector &infos) = 0; - - private: - sigslot::signal3 SignalSendPacket; - sigslot::signal3 SignalRecvPacket; - talk_base::CriticalSection signal_send_packet_cs_; - talk_base::CriticalSection signal_recv_packet_cs_; - - talk_base::Thread *worker_thread_; - MediaEngineInterface *media_engine_; - BaseSession *session_; - MediaChannel *media_channel_; - - std::string content_name_; - bool rtcp_; - TransportChannel *transport_channel_; - TransportChannel *rtcp_transport_channel_; - SrtpFilter srtp_filter_; - RtcpMuxFilter rtcp_mux_filter_; - talk_base::scoped_ptr socket_monitor_; - bool enabled_; - bool writable_; - bool was_ever_writable_; - bool has_local_content_; - bool has_remote_content_; - bool muted_; -}; - -// VoiceChannel is a specialization that adds support for early media, DTMF, -// and input/output level monitoring. -class VoiceChannel : public BaseChannel { - public: - VoiceChannel(talk_base::Thread *thread, MediaEngineInterface *media_engine, - VoiceMediaChannel *channel, BaseSession *session, - const std::string& content_name, bool rtcp); - ~VoiceChannel(); - bool Init(); - - // downcasts a MediaChannel - virtual VoiceMediaChannel* media_channel() const { - return static_cast(BaseChannel::media_channel()); - } - - // Add an incoming stream with the specified SSRC. - bool AddStream(uint32 ssrc); - - bool SetRingbackTone(const void* buf, int len); - void SetEarlyMedia(bool enable); - // This signal is emitted when we have gone a period of time without - // receiving early media. When received, a UI should start playing its - // own ringing sound - sigslot::signal1 SignalEarlyMediaTimeout; - - bool PlayRingbackTone(uint32 ssrc, bool play, bool loop); - bool PressDTMF(int digit, bool playout); - bool SetOutputScaling(uint32 ssrc, double left, double right); - - // Monitoring functions - sigslot::signal2 &> - SignalConnectionMonitor; - - void StartMediaMonitor(int cms); - void StopMediaMonitor(); - sigslot::signal2 SignalMediaMonitor; - - void StartAudioMonitor(int cms); - void StopAudioMonitor(); - bool IsAudioMonitorRunning() const; - sigslot::signal2 SignalAudioMonitor; - - int GetInputLevel_w(); - int GetOutputLevel_w(); - void GetActiveStreams_w(AudioInfo::StreamList* actives); - - // Signal errors from VoiceMediaChannel. Arguments are: - // ssrc(uint32), and error(VoiceMediaChannel::Error). - sigslot::signal3 - SignalMediaError; - - private: - struct SetRingbackToneMessageData : public talk_base::MessageData { - SetRingbackToneMessageData(const void* b, int l) - : buf(b), - len(l), - result(false) { - } - const void* buf; - int len; - bool result; - }; - struct PlayRingbackToneMessageData : public talk_base::MessageData { - PlayRingbackToneMessageData(uint32 s, bool p, bool l) - : ssrc(s), - play(p), - loop(l), - result(false) { - } - uint32 ssrc; - bool play; - bool loop; - bool result; - }; - struct DtmfMessageData : public talk_base::MessageData { - DtmfMessageData(int d, bool p) - : digit(d), - playout(p), - result(false) { - } - int digit; - bool playout; - bool result; - }; - struct ScaleVolumeMessageData : public talk_base::MessageData { - ScaleVolumeMessageData(uint32 s, double l, double r) - : ssrc(s), - left(l), - right(r), - result(false) { - } - uint32 ssrc; - double left; - double right; - bool result; - }; - - // overrides from BaseChannel - virtual void OnChannelRead(TransportChannel* channel, - const char *data, size_t len); - virtual void ChangeState(); - virtual const MediaContentDescription* GetFirstContent( - const SessionDescription* sdesc); - virtual bool SetLocalContent_w(const MediaContentDescription* content, - ContentAction action); - virtual bool SetRemoteContent_w(const MediaContentDescription* content, - ContentAction action); - - void AddStream_w(uint32 ssrc); - void RemoveStream_w(uint32 ssrc); - - bool SetRingbackTone_w(const void* buf, int len); - bool PlayRingbackTone_w(uint32 ssrc, bool play, bool loop); - void HandleEarlyMediaTimeout(); - bool PressDTMF_w(int digit, bool playout); - bool SetOutputScaling_w(uint32 ssrc, double left, double right); - - virtual void OnMessage(talk_base::Message *pmsg); - virtual void OnConnectionMonitorUpdate( - SocketMonitor *monitor, const std::vector &infos); - virtual void OnMediaMonitorUpdate( - VoiceMediaChannel *media_channel, const VoiceMediaInfo& info); - void OnAudioMonitorUpdate(AudioMonitor *monitor, const AudioInfo& info); - void OnVoiceChannelError(uint32 ssrc, VoiceMediaChannel::Error error); - void SendLastMediaError(); - void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error); - - static const int kEarlyMediaTimeout = 1000; - bool received_media_; - talk_base::scoped_ptr media_monitor_; - talk_base::scoped_ptr audio_monitor_; -}; - -// VideoChannel is a specialization for video. -class VideoChannel : public BaseChannel { - public: - VideoChannel(talk_base::Thread *thread, MediaEngineInterface *media_engine, - VideoMediaChannel *channel, BaseSession *session, - const std::string& content_name, bool rtcp, - VoiceChannel *voice_channel); - ~VideoChannel(); - bool Init(); - - // downcasts a MediaChannel - virtual VideoMediaChannel* media_channel() const { - return static_cast(BaseChannel::media_channel()); - } - - // Add an incoming stream with the specified SSRC. - bool AddStream(uint32 ssrc, uint32 voice_ssrc); - - bool SetRenderer(uint32 ssrc, VideoRenderer* renderer); - - - sigslot::signal2 &> - SignalConnectionMonitor; - - void StartMediaMonitor(int cms); - void StopMediaMonitor(); - sigslot::signal2 SignalMediaMonitor; - - bool SendIntraFrame(); - bool RequestIntraFrame(); - void EnableCpuAdaptation(bool enable); - - sigslot::signal3 - SignalMediaError; - - void SetCaptureDevice(uint32 ssrc, webrtc::VideoCaptureModule* camera); - void SetLocalRenderer(uint32 ssrc, VideoRenderer* renderer); - - private: - // overrides from BaseChannel - virtual void ChangeState(); - virtual const MediaContentDescription* GetFirstContent( - const SessionDescription* sdesc); - virtual bool SetLocalContent_w(const MediaContentDescription* content, - ContentAction action); - virtual bool SetRemoteContent_w(const MediaContentDescription* content, - ContentAction action); - - void AddStream_w(uint32 ssrc, uint32 voice_ssrc); - void RemoveStream_w(uint32 ssrc); - - void SendIntraFrame_w() { - media_channel()->SendIntraFrame(); - } - void RequestIntraFrame_w() { - media_channel()->RequestIntraFrame(); - } - void EnableCpuAdaptation_w(bool enable) { - // TODO: The following call will clear all other options, which is - // OK now since SetOptions is not used in video media channel. In the - // future, add GetOptions() method and change the options. - media_channel()->SetOptions(enable ? OPT_CPU_ADAPTATION : 0); - } - - struct RenderMessageData : public talk_base::MessageData { - RenderMessageData(uint32 s, VideoRenderer* r) : ssrc(s), renderer(r) {} - uint32 ssrc; - VideoRenderer* renderer; - }; - - - void SetRenderer_w(uint32 ssrc, VideoRenderer* renderer); - - - virtual void OnMessage(talk_base::Message *pmsg); - virtual void OnConnectionMonitorUpdate( - SocketMonitor *monitor, const std::vector &infos); - virtual void OnMediaMonitorUpdate( - VideoMediaChannel *media_channel, const VideoMediaInfo& info); - void OnVideoChannelError(uint32 ssrc, VideoMediaChannel::Error error); - void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error); - - VoiceChannel *voice_channel_; - VideoRenderer *renderer_; - talk_base::scoped_ptr media_monitor_; -}; - -} // namespace cricket - -#endif // TALK_SESSION_PHONE_CHANNEL_H_ diff --git a/third_party_mods/libjingle/source/talk/session/phone/mediasession.cc b/third_party_mods/libjingle/source/talk/session/phone/mediasession.cc deleted file mode 100644 index 2a441d4e5..000000000 --- a/third_party_mods/libjingle/source/talk/session/phone/mediasession.cc +++ /dev/null @@ -1,591 +0,0 @@ -/* - * libjingle - * Copyright 2004--2005, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include "talk/session/phone/mediasession.h" - -#include "talk/base/helpers.h" -#include "talk/base/logging.h" -#include "talk/base/scoped_ptr.h" -#include "talk/p2p/base/constants.h" -#include "talk/session/phone/channelmanager.h" -#include "talk/session/phone/cryptoparams.h" -#include "talk/session/phone/srtpfilter.h" -#include "talk/xmpp/constants.h" - -namespace { -const char kInline[] = "inline:"; -} - -namespace cricket { - -using talk_base::scoped_ptr; - -static bool CreateCryptoParams(int tag, const std::string& cipher, - CryptoParams *out) { - std::string key; - key.reserve(SRTP_MASTER_KEY_BASE64_LEN); - - if (!talk_base::CreateRandomString(SRTP_MASTER_KEY_BASE64_LEN, &key)) { - return false; - } - out->tag = tag; - out->cipher_suite = cipher; - out->key_params = kInline; - out->key_params += key; - return true; -} - -#ifdef HAVE_SRTP -static bool AddCryptoParams(const std::string& cipher_suite, - CryptoParamsVec *out) { - int size = out->size(); - - out->resize(size + 1); - return CreateCryptoParams(size, cipher_suite, &out->at(size)); -} -#endif - -// For audio, HMAC 32 is prefered because of the low overhead. -static bool GetSupportedAudioCryptos(CryptoParamsVec* cryptos) { -#ifdef HAVE_SRTP - return AddCryptoParams(CS_AES_CM_128_HMAC_SHA1_32, cryptos) && - AddCryptoParams(CS_AES_CM_128_HMAC_SHA1_80, cryptos); -#else - return false; -#endif -} - -static bool GetSupportedVideoCryptos(CryptoParamsVec* cryptos) { -#ifdef HAVE_SRTP - return AddCryptoParams(CS_AES_CM_128_HMAC_SHA1_80, cryptos); -#else - return false; -#endif -} - -// For video support only 80-bit SHA1 HMAC. For audio 32-bit HMAC is -// tolerated because it is low overhead. Pick the crypto in the list -// that is supported. -static bool SelectCrypto(const MediaContentDescription* offer, - CryptoParams *crypto) { - bool audio = offer->type() == MEDIA_TYPE_AUDIO; - const CryptoParamsVec& cryptos = offer->cryptos(); - - for (CryptoParamsVec::const_iterator i = cryptos.begin(); - i != cryptos.end(); ++i) { - if (CS_AES_CM_128_HMAC_SHA1_80 == i->cipher_suite || - (CS_AES_CM_128_HMAC_SHA1_32 == i->cipher_suite && audio)) { - return CreateCryptoParams(i->tag, i->cipher_suite, crypto); - } - } - return false; -} - -static const StreamParams* FindStreamParamsByName( - const StreamParamsVec& params_vec, - const std::string& name) { - for (StreamParamsVec::const_iterator it = params_vec.begin(); - it != params_vec.end(); ++it) { - if (it->name == name) - return &*it; - } - return NULL; -} - -static const StreamParams* FindFirstStreamParamsByCname( - const StreamParamsVec& params_vec, - const std::string& cname) { - for (StreamParamsVec::const_iterator it = params_vec.begin(); - it != params_vec.end(); ++it) { - if (cname == it->cname) - return &*it; - } - return NULL; -} - -static const StreamParams* FindStreamParamsBySsrc( - const StreamParamsVec& params_vec, - uint32 ssrc) { - for (StreamParamsVec::const_iterator stream_it = params_vec.begin(); - stream_it != params_vec.end(); ++stream_it) { - const std::vector& ssrcs = stream_it->ssrcs; - for (std::vector::const_iterator ssrc_it = ssrcs.begin(); - ssrc_it != ssrcs.end(); ++ssrc_it) { - if (ssrc == *ssrc_it) - return &*stream_it; - } - } - return NULL; -} - -// Generates a new CNAME or the CNAME of an already existing StreamParams -// if a StreamParams exist for another Stream in streams with sync_label -// sync_label. -static bool GenerateCname(const StreamParamsVec& params_vec, - const MediaSessionOptions::Streams& streams, - const std::string& synch_label, - std::string* cname) { - ASSERT(cname); - if (!cname) - return false; - - // Check if a CNAME exist for any of the other synched streams. - for (MediaSessionOptions::Streams::const_iterator stream_it = streams.begin(); - stream_it != streams.end() ; ++stream_it) { - if (synch_label != stream_it->sync_label) - continue; - const StreamParams* param = FindStreamParamsByName(params_vec, - stream_it->name); - if (param) { - *cname = param->cname; - return true; - } - } - // No other stream seems to exist that we should sync with. - // Generate a random string for the RTCP CNAME, as stated in RFC 6222. - // This string is only used for synchronization, and therefore is opaque. - do { - if (!talk_base::CreateRandomString(16, cname)) { - ASSERT(false); - return false; - } - } while (FindFirstStreamParamsByCname(params_vec, *cname)); - - return true; -} - -// Generate a new SSRC and make sure it does not exist in params_vec. -static uint32 GenerateSsrc(const StreamParamsVec& params_vec) { - uint32 ssrc = 0; - do { - ssrc = talk_base::CreateRandomNonZeroId(); - } while (FindStreamParamsBySsrc(params_vec, ssrc)); - return ssrc; -} - -// Finds all StreamParams of all media types and attach them to stream_params. -static void GetCurrentStreamParams(const SessionDescription* sdesc, - StreamParamsVec* stream_params) { - if (!sdesc) - return; - - const ContentInfos& contents = sdesc->contents(); - for (ContentInfos::const_iterator content = contents.begin(); - content != contents.end(); content++) { - if (!IsAudioContent(&*content) && !IsVideoContent(&*content)) - continue; - const MediaContentDescription* media = - static_cast( - content->description); - const StreamParamsVec& streams = media->streams(); - for (StreamParamsVec::const_iterator it = streams.begin(); - it != streams.end(); ++it) { - stream_params->push_back(*it); - } - } -} - -// Adds a StreamParams for each Stream in Streams with media type -// media_type to content_description. -// current_parms - All currently known StreamParams of any media type. -static bool AddStreamParams( - MediaType media_type, - const MediaSessionOptions::Streams& streams, - StreamParamsVec* current_params, - MediaContentDescription* content_description) { - for (MediaSessionOptions::Streams::const_iterator stream_it = streams.begin(); - stream_it != streams.end(); ++stream_it) { - if (stream_it->type != media_type) - continue; // Wrong media type. - const StreamParams* params = FindStreamParamsByName(*current_params, - stream_it->name); - if (!params) { - // This is a new stream. - // Get a CNAME. Either new or same as one of the other synched streams. - std::string cname; - if (!GenerateCname(*current_params, streams, stream_it->sync_label, - &cname)) { - return false; - } - uint32 ssrc = GenerateSsrc(*current_params); - // TODO(perkj): Generate the more complex types of stream_params. - StreamParams stream_param(stream_it->name, ssrc, cname, - stream_it->sync_label); - content_description->AddStream(stream_param); - - // Store the new StreamParams in current_params. - // This is necessary so that we can use the CNAME for other media types. - current_params->push_back(stream_param); - } else { - content_description->AddStream(*params); - } - } - return true; -} - -void MediaSessionOptions::AddStream(MediaType type, - const std::string& name, - const std::string& sync_label) { - streams.push_back(Stream(type, name, sync_label)); - - if (type == MEDIA_TYPE_VIDEO) - has_video = true; - else if (type == MEDIA_TYPE_AUDIO) - has_audio = true; -} - -void MediaSessionOptions::RemoveStream(MediaType type, - const std::string& name) { - Streams::iterator stream_it = streams.begin(); - for (; stream_it != streams.end(); ++stream_it) { - if (stream_it->type == type && stream_it->name == name) { - streams.erase(stream_it); - break; - } - } - ASSERT(stream_it != streams.end()); -} - -MediaSessionDescriptionFactory::MediaSessionDescriptionFactory() - : secure_(SEC_DISABLED) { -} - -MediaSessionDescriptionFactory::MediaSessionDescriptionFactory( - ChannelManager* channel_manager) - : secure_(SEC_DISABLED) { - channel_manager->GetSupportedAudioCodecs(&audio_codecs_); - channel_manager->GetSupportedVideoCodecs(&video_codecs_); -} - -SessionDescription* MediaSessionDescriptionFactory::CreateOffer( - const MediaSessionOptions& options) { - return CreateOffer(options, NULL); -} - -SessionDescription* MediaSessionDescriptionFactory::CreateOffer( - const MediaSessionOptions& options, - const SessionDescription* current_description) { - scoped_ptr offer(new SessionDescription()); - - StreamParamsVec current_params; - GetCurrentStreamParams(current_description, ¤t_params); - - if (options.has_audio) { - scoped_ptr audio(new AudioContentDescription()); - for (AudioCodecs::const_iterator codec = audio_codecs_.begin(); - codec != audio_codecs_.end(); ++codec) { - audio->AddCodec(*codec); - } - audio->SortCodecs(); - if (!AddStreamParams(MEDIA_TYPE_AUDIO, options.streams, ¤t_params, - audio.get())) { - return NULL; // Abort, something went seriously wrong. - } - - if (options.streams.empty()) { - // TODO(perkj): Remove this legacy ssrc when all apps use StreamParams. - audio->set_ssrc(talk_base::CreateRandomNonZeroId()); - } - audio->set_rtcp_mux(true); - audio->set_lang(lang_); - - if (secure() != SEC_DISABLED) { - CryptoParamsVec audio_cryptos; - if (current_description) { - // Copy crypto parameters from the previous offer. - const ContentInfo* info = - GetFirstAudioContent(current_description); - if (info) { - const AudioContentDescription* desc = - static_cast(info->description); - audio_cryptos = desc->cryptos(); - } - } - if (audio_cryptos.empty()) - GetSupportedAudioCryptos(&audio_cryptos); // Generate new cryptos. - - for (CryptoParamsVec::const_iterator crypto = audio_cryptos.begin(); - crypto != audio_cryptos.end(); ++crypto) { - audio->AddCrypto(*crypto); - } - - if (secure() == SEC_REQUIRED) { - if (audio->cryptos().empty()) { - return NULL; // Abort, crypto required but none found. - } - audio->set_crypto_required(true); - } - } - - offer->AddContent(CN_AUDIO, NS_JINGLE_RTP, audio.release()); - } - - // add video codecs, if this is a video call - if (options.has_video) { - scoped_ptr video(new VideoContentDescription()); - for (VideoCodecs::const_iterator codec = video_codecs_.begin(); - codec != video_codecs_.end(); ++codec) { - video->AddCodec(*codec); - } - - video->SortCodecs(); - if (!AddStreamParams(MEDIA_TYPE_VIDEO, options.streams, ¤t_params, - video.get())) { - return NULL; // Abort, something went seriously wrong. - } - - if (options.streams.empty()) { - // TODO(perkj): Remove this legacy ssrc when all apps use StreamParams. - video->set_ssrc(talk_base::CreateRandomNonZeroId()); - } - video->set_bandwidth(options.video_bandwidth); - video->set_rtcp_mux(true); - - if (secure() != SEC_DISABLED) { - CryptoParamsVec video_cryptos; - if (current_description) { - // Copy crypto parameters from the previous offer. - const ContentInfo* info = - GetFirstVideoContent(current_description); - if (info) { - const VideoContentDescription* desc = - static_cast(info->description); - video_cryptos = desc->cryptos(); - } - } - if (video_cryptos.empty()) - GetSupportedVideoCryptos(&video_cryptos); // Generate new crypto. - for (CryptoParamsVec::const_iterator crypto = video_cryptos.begin(); - crypto != video_cryptos.end(); ++crypto) { - video->AddCrypto(*crypto); - } - if (secure() == SEC_REQUIRED) { - if (video->cryptos().empty()) { - return NULL; // Abort, crypto required but none found. - } - video->set_crypto_required(true); - } - } - - offer->AddContent(CN_VIDEO, NS_JINGLE_RTP, video.release()); - } - - return offer.release(); -} - -SessionDescription* MediaSessionDescriptionFactory::CreateAnswer( - const SessionDescription* offer, - const MediaSessionOptions& options) { - return CreateAnswer(offer, options, NULL); -} - -SessionDescription* MediaSessionDescriptionFactory::CreateAnswer( - const SessionDescription* offer, const MediaSessionOptions& options, - const SessionDescription* current_description) { - // The answer contains the intersection of the codecs in the offer with the - // codecs we support, ordered by our local preference. As indicated by - // XEP-0167, we retain the same payload ids from the offer in the answer. - scoped_ptr accept(new SessionDescription()); - - StreamParamsVec current_params; - GetCurrentStreamParams(current_description, ¤t_params); - - const ContentInfo* audio_content = GetFirstAudioContent(offer); - if (audio_content && options.has_audio) { - const AudioContentDescription* audio_offer = - static_cast(audio_content->description); - scoped_ptr audio_accept( - new AudioContentDescription()); - for (AudioCodecs::const_iterator ours = audio_codecs_.begin(); - ours != audio_codecs_.end(); ++ours) { - for (AudioCodecs::const_iterator theirs = audio_offer->codecs().begin(); - theirs != audio_offer->codecs().end(); ++theirs) { - if (ours->Matches(*theirs)) { - AudioCodec negotiated(*ours); - negotiated.id = theirs->id; - audio_accept->AddCodec(negotiated); - } - } - } - - audio_accept->SortCodecs(); - if (!AddStreamParams(MEDIA_TYPE_AUDIO, options.streams, ¤t_params, - audio_accept.get())) { - return NULL; // Abort, something went seriously wrong. - } - - if (options.streams.empty()) { - // TODO(perkj): Remove this legacy ssrc when all apps use StreamParams. - audio_accept->set_ssrc(talk_base::CreateRandomNonZeroId()); - } - audio_accept->set_rtcp_mux(audio_offer->rtcp_mux()); - - if (secure() != SEC_DISABLED) { - CryptoParams crypto; - - if (SelectCrypto(audio_offer, &crypto)) { - if (current_description) { - // Check if this crypto already exist in the previous - // session description. Use it in that case. - const ContentInfo* info = - GetFirstAudioContent(current_description); - if (info) { - const AudioContentDescription* desc = - static_cast(info->description); - const CryptoParamsVec& cryptos = desc->cryptos(); - for (CryptoParamsVec::const_iterator it = cryptos.begin(); - it != cryptos.end(); ++it) { - if (crypto.Matches(*it)) { - crypto = *it; - break; - } - } - } - } - audio_accept->AddCrypto(crypto); - } - } - - if (audio_accept->cryptos().empty() && - (audio_offer->crypto_required() || secure() == SEC_REQUIRED)) { - return NULL; // Fails the session setup. - } - accept->AddContent(audio_content->name, audio_content->type, - audio_accept.release()); - } else { - LOG(LS_INFO) << "Audio is not supported in answer"; - } - - const ContentInfo* video_content = GetFirstVideoContent(offer); - if (video_content && options.has_video) { - const VideoContentDescription* video_offer = - static_cast(video_content->description); - scoped_ptr video_accept( - new VideoContentDescription()); - for (VideoCodecs::const_iterator ours = video_codecs_.begin(); - ours != video_codecs_.end(); ++ours) { - for (VideoCodecs::const_iterator theirs = video_offer->codecs().begin(); - theirs != video_offer->codecs().end(); ++theirs) { - if (ours->Matches(*theirs)) { - VideoCodec negotiated(*ours); - negotiated.id = theirs->id; - video_accept->AddCodec(negotiated); - } - } - } - if (!AddStreamParams(MEDIA_TYPE_VIDEO, options.streams, ¤t_params, - video_accept.get())) { - return NULL; // Abort, something went seriously wrong. - } - - if (options.streams.empty()) { - // TODO(perkj): Remove this legacy ssrc when all apps use StreamParams. - video_accept->set_ssrc(talk_base::CreateRandomNonZeroId()); - } - video_accept->set_bandwidth(options.video_bandwidth); - video_accept->set_rtcp_mux(video_offer->rtcp_mux()); - video_accept->SortCodecs(); - - if (secure() != SEC_DISABLED) { - CryptoParams crypto; - - if (SelectCrypto(video_offer, &crypto)) { - if (current_description) { - // Check if this crypto already exist in the previous - // session description. Use it in that case. - const ContentInfo* info = GetFirstVideoContent(current_description); - if (info) { - const VideoContentDescription* desc = - static_cast(info->description); - const CryptoParamsVec& cryptos = desc->cryptos(); - for (CryptoParamsVec::const_iterator it = cryptos.begin(); - it != cryptos.end(); ++it) { - if (crypto.Matches(*it)) { - crypto = *it; - break; - } - } - } - } - video_accept->AddCrypto(crypto); - } - } - - if (video_accept->cryptos().empty() && - (video_offer->crypto_required() || secure() == SEC_REQUIRED)) { - return NULL; // Fails the session setup. - } - accept->AddContent(video_content->name, video_content->type, - video_accept.release()); - } else { - LOG(LS_INFO) << "Video is not supported in answer"; - } - return accept.release(); -} - -static bool IsMediaContent(const ContentInfo* content, MediaType media_type) { - if (content == NULL || content->type != NS_JINGLE_RTP) { - return false; - } - - const MediaContentDescription* media = - static_cast(content->description); - return media->type() == media_type; -} - -bool IsAudioContent(const ContentInfo* content) { - return IsMediaContent(content, MEDIA_TYPE_AUDIO); -} - -bool IsVideoContent(const ContentInfo* content) { - return IsMediaContent(content, MEDIA_TYPE_VIDEO); -} - -static const ContentInfo* GetFirstMediaContent(const SessionDescription* sdesc, - MediaType media_type) { - if (sdesc == NULL) - return NULL; - - const ContentInfos& contents = sdesc->contents(); - for (ContentInfos::const_iterator content = contents.begin(); - content != contents.end(); content++) { - if (IsMediaContent(&*content, media_type)) { - return &*content; - } - } - return NULL; -} - -const ContentInfo* GetFirstAudioContent(const SessionDescription* sdesc) { - return GetFirstMediaContent(sdesc, MEDIA_TYPE_AUDIO); -} - -const ContentInfo* GetFirstVideoContent(const SessionDescription* sdesc) { - return GetFirstMediaContent(sdesc, MEDIA_TYPE_VIDEO); -} - -} // namespace cricket diff --git a/third_party_mods/libjingle/source/talk/session/phone/mediasession.h b/third_party_mods/libjingle/source/talk/session/phone/mediasession.h deleted file mode 100644 index c3fe77671..000000000 --- a/third_party_mods/libjingle/source/talk/session/phone/mediasession.h +++ /dev/null @@ -1,284 +0,0 @@ -/* - * libjingle - * Copyright 2004--2005, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -// Types and classes used in media session descriptions. - -#ifndef TALK_SESSION_PHONE_MEDIASESSION_H_ -#define TALK_SESSION_PHONE_MEDIASESSION_H_ - -#include -#include -#include - -#include "talk/session/phone/codec.h" -#include "talk/session/phone/cryptoparams.h" -#include "talk/session/phone/mediachannel.h" -#include "talk/session/phone/streamparams.h" -#include "talk/p2p/base/sessiondescription.h" - -namespace cricket { - -class ChannelManager; -typedef std::vector AudioCodecs; -typedef std::vector VideoCodecs; -typedef std::vector CryptoParamsVec; -typedef std::vector StreamParamsVec; - -// SEC_ENABLED and SEC_REQUIRED should only be used if the session -// was negotiated over TLS, to protect the inline crypto material -// exchange. -// SEC_DISABLED: No crypto in outgoing offer and answer. Fail any -// offer with crypto required. -// SEC_ENABLED: Crypto in outgoing offer and answer. Fail any offer -// with unsupported required crypto. Crypto set but not -// required in outgoing offer. -// SEC_REQUIRED: Crypto in outgoing offer and answer with -// required='true'. Fail any offer with no or -// unsupported crypto (implicit crypto required='true' -// in the offer.) -enum SecureMediaPolicy { - SEC_DISABLED, - SEC_ENABLED, - SEC_REQUIRED -}; - -enum MediaType { - MEDIA_TYPE_AUDIO, - MEDIA_TYPE_VIDEO -}; - -// Options to control how session descriptions are generated. -const int kAutoBandwidth = -1; -struct MediaSessionOptions { - MediaSessionOptions() : - has_audio(true), // Audio enabled by default. - has_video(false), - is_muc(false), - video_bandwidth(kAutoBandwidth) { - } - - // Add a stream with MediaType type and id name. - // All streams with the same sync_label will get the same CNAME. - // All names must be unique. - void AddStream(MediaType type, - const std::string& name, - const std::string& sync_label); - void RemoveStream(MediaType type, const std::string& name); - - bool has_audio; - bool has_video; - bool is_muc; - // bps. -1 == auto. - int video_bandwidth; - - struct Stream { - Stream(MediaType type, - const std::string& name, - const std::string& sync_label) - : type(type), name(name), sync_label(sync_label) { - } - MediaType type; - std::string name; - std::string sync_label; - }; - - typedef std::vector Streams; - Streams streams; -}; - -// "content" (as used in XEP-0166) descriptions for voice and video. -class MediaContentDescription : public ContentDescription { - public: - MediaContentDescription() - : ssrc_(0), - ssrc_set_(false), - rtcp_mux_(false), - bandwidth_(kAutoBandwidth), - crypto_required_(false), - rtp_header_extensions_set_(false) { - } - - virtual MediaType type() const = 0; - - uint32 ssrc() const { return ssrc_; } - bool ssrc_set() const { return ssrc_set_; } - void set_ssrc(uint32 ssrc) { - ssrc_ = ssrc; - ssrc_set_ = true; - } - - bool rtcp_mux() const { return rtcp_mux_; } - void set_rtcp_mux(bool mux) { rtcp_mux_ = mux; } - - int bandwidth() const { return bandwidth_; } - void set_bandwidth(int bandwidth) { bandwidth_ = bandwidth; } - - const std::vector& cryptos() const { return cryptos_; } - void AddCrypto(const CryptoParams& params) { - cryptos_.push_back(params); - } - bool crypto_required() const { return crypto_required_; } - void set_crypto_required(bool crypto) { - crypto_required_ = crypto; - } - - const std::vector& rtp_header_extensions() const { - return rtp_header_extensions_; - } - void AddRtpHeaderExtension(const RtpHeaderExtension& ext) { - rtp_header_extensions_.push_back(ext); - rtp_header_extensions_set_ = true; - } - void ClearRtpHeaderExtensions() { - rtp_header_extensions_.clear(); - rtp_header_extensions_set_ = true; - } - // We can't always tell if an empty list of header extensions is - // because the other side doesn't support them, or just isn't hooked up to - // signal them. For now we assume an empty list means no signaling, but - // provide the ClearRtpHeaderExtensions method to allow "no support" to be - // clearly indicated (i.e. when derived from other information). - bool rtp_header_extensions_set() const { - return rtp_header_extensions_set_; - } - const StreamParamsVec& streams() const { - return streams_; - } - void AddStream(const StreamParams& stream) { - streams_.push_back(stream); - } - - protected: - uint32 ssrc_; - bool ssrc_set_; - bool rtcp_mux_; - int bandwidth_; - std::vector cryptos_; - bool crypto_required_; - std::vector rtp_header_extensions_; - bool rtp_header_extensions_set_; - StreamParamsVec streams_; -}; - -template -class MediaContentDescriptionImpl : public MediaContentDescription { - public: - struct PreferenceSort { - bool operator()(C a, C b) { return a.preference > b.preference; } - }; - - const std::vector& codecs() const { return codecs_; } - void AddCodec(const C& codec) { - codecs_.push_back(codec); - } - void SortCodecs() { - std::sort(codecs_.begin(), codecs_.end(), PreferenceSort()); - } - - private: - std::vector codecs_; -}; - -class AudioContentDescription : public MediaContentDescriptionImpl { - public: - AudioContentDescription() : - conference_mode_(false) {} - - virtual MediaType type() const { return MEDIA_TYPE_AUDIO; } - - bool conference_mode() const { return conference_mode_; } - void set_conference_mode(bool enable) { - conference_mode_ = enable; - } - - const std::string &lang() const { return lang_; } - void set_lang(const std::string &lang) { lang_ = lang; } - - private: - bool conference_mode_; - std::string lang_; -}; - -class VideoContentDescription : public MediaContentDescriptionImpl { - public: - virtual MediaType type() const { return MEDIA_TYPE_VIDEO; } -}; - -// Creates media session descriptions according to the supplied codecs and -// other fields, as well as the supplied per-call options. -// When creating answers, performs the appropriate negotiation -// of the various fields to determine the proper result. -class MediaSessionDescriptionFactory { - public: - // Default ctor; use methods below to set configuration. - MediaSessionDescriptionFactory(); - // Helper, to allow configuration to be loaded from a ChannelManager. - explicit MediaSessionDescriptionFactory(ChannelManager* manager); - - const AudioCodecs& audio_codecs() const { return audio_codecs_; } - void set_audio_codecs(const AudioCodecs& codecs) { audio_codecs_ = codecs; } - const VideoCodecs& video_codecs() const { return video_codecs_; } - void set_video_codecs(const VideoCodecs& codecs) { video_codecs_ = codecs; } - SecureMediaPolicy secure() const { return secure_; } - void set_secure(SecureMediaPolicy s) { secure_ = s; } - - // TODO(perkj) Deprecate this version of CreateOffer and - // force to use the second alternative. - SessionDescription* CreateOffer( - const MediaSessionOptions& options); - - SessionDescription* CreateOffer( - const MediaSessionOptions& options, - const SessionDescription* current_description); - - // TODO(perkj) Deprecate this version of CreateAnswer and - // force to use the second alternative. - SessionDescription* CreateAnswer( - const SessionDescription* offer, - const MediaSessionOptions& options); - - SessionDescription* CreateAnswer( - const SessionDescription* offer, - const MediaSessionOptions& options, - const SessionDescription* current_description); - - private: - AudioCodecs audio_codecs_; - VideoCodecs video_codecs_; - SecureMediaPolicy secure_; - std::string lang_; -}; - -// Convenience functions. -bool IsAudioContent(const ContentInfo* content); -bool IsVideoContent(const ContentInfo* content); -const ContentInfo* GetFirstAudioContent(const SessionDescription* sdesc); -const ContentInfo* GetFirstVideoContent(const SessionDescription* sdesc); - -} // namespace cricket - -#endif // TALK_SESSION_PHONE_MEDIASESSION_H_ diff --git a/third_party_mods/libjingle/source/talk/session/phone/streamparams.h b/third_party_mods/libjingle/source/talk/session/phone/streamparams.h deleted file mode 100644 index 527d9b704..000000000 --- a/third_party_mods/libjingle/source/talk/session/phone/streamparams.h +++ /dev/null @@ -1,105 +0,0 @@ -/* - * libjingle - * Copyright 2011, Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -// This file contains structures for describing SSRCs from a media source such -// as a MediaStreamTrack when it is sent across an RTP session. Multiple media -// sources may be sent across the same RTP session, each of them will be -// described by one StreamParams object -// SsrcGroup is used to describe the relationship between the SSRCs that -// are used for this media source. -// E.x: Consider a source that is sent as 3 simulcast streams -// Let the simulcast elements have SSRC 10, 20, 30. -// Let each simulcast element use FEC and let the protection packets have -// SSRC 11,21,31. -// To describe this 4 SsrcGroups are needed, -// StreamParams would then contain ssrc = {10,11,20,21,30,31} and -// ssrc_groups = {{SIM,{10,20,30}, {FEC,{10,11}, {FEC, {20,21}, {FEC {30,31}}} -// Please see RFC 5576. - -#ifndef TALK_SESSION_PHONE_STREAMPARAMS_H_ -#define TALK_SESSION_PHONE_STREAMPARAMS_H_ - -#include -#include - -namespace cricket { - -struct SsrcGroup { - SsrcGroup(const std::string& usage, const std::vector& ssrcs) - : semantics(usage), ssrcs(ssrcs) { - } - bool operator==(const SsrcGroup& other) const { - return (semantics == other.semantics && ssrcs == other.ssrcs); - } - bool operator!=(const SsrcGroup &other) const { - return !(*this == other); - } - - std::string semantics; // e.g FIX, FEC, SIM. - std::vector ssrcs; // SSRCs of this type. -}; - -struct StreamParams { - StreamParams(const std::string& name, - const std::vector& ssrcs, - const std::vector& ssrc_groups, - const std::string& cname, - const std::string& sync_label) - : name(name), - ssrcs(ssrcs), - ssrc_groups(ssrc_groups), - cname(cname), - sync_label(sync_label) { - } - StreamParams(const std::string& name, - uint32 ssrc, - const std::string& cname, - const std::string& sync_label) - : name(name), - cname(cname), - sync_label(sync_label) { - ssrcs.push_back(ssrc); - } - bool operator==(const StreamParams& other) const { - return (name == other.name && ssrcs == other.ssrcs && - ssrc_groups == other.ssrc_groups && cname == other.cname && - sync_label == sync_label); - } - bool operator!=(const StreamParams &other) const { - return !(*this == other); - } - - std::string name; // Unique name of this source. - std::vector ssrcs; // All SSRCs for this source - std::vector ssrc_groups; // e.g. FID, FEC, SIM - std::string cname; // RTCP CNAME - std::string sync_label; // Friendly name of cname. -}; - -} // namespace cricket - -#endif // TALK_SESSION_PHONE_STREAMPARAMS_H_