diff --git a/webrtc/config.cc b/webrtc/config.cc deleted file mode 100644 index 021bbbfcd..000000000 --- a/webrtc/config.cc +++ /dev/null @@ -1,17 +0,0 @@ -/* - * Copyright (c) 2013 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 "webrtc/config.h" - -namespace webrtc { -const char* RtpExtension::kTOffset = "urn:ietf:params:rtp-hdrext:toffset"; -const char* RtpExtension::kAbsSendTime = - "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"; -} // namespace webrtc diff --git a/webrtc/system_wrappers/source/system_wrappers_tests.gyp b/webrtc/system_wrappers/source/system_wrappers_tests.gyp index b8587eb90..5686105e4 100644 --- a/webrtc/system_wrappers/source/system_wrappers_tests.gyp +++ b/webrtc/system_wrappers/source/system_wrappers_tests.gyp @@ -18,7 +18,6 @@ '<(webrtc_root)/test/test.gyp:test_support_main', ], 'sources': [ - '../../common_unittest.cc', 'aligned_malloc_unittest.cc', 'clock_unittest.cc', 'condition_variable_unittest.cc', diff --git a/webrtc/common_unittest.cc b/webrtc/test/common_unittest.cc similarity index 100% rename from webrtc/common_unittest.cc rename to webrtc/test/common_unittest.cc diff --git a/webrtc/call.cc b/webrtc/video/call.cc similarity index 97% rename from webrtc/call.cc rename to webrtc/video/call.cc index ff184cdf0..e0df6fe1e 100644 --- a/webrtc/call.cc +++ b/webrtc/video/call.cc @@ -16,6 +16,7 @@ #include "webrtc/call.h" #include "webrtc/common.h" +#include "webrtc/config.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" @@ -28,6 +29,9 @@ #include "webrtc/video_engine/include/vie_rtp_rtcp.h" namespace webrtc { +const char* RtpExtension::kTOffset = "urn:ietf:params:rtp-hdrext:toffset"; +const char* RtpExtension::kAbsSendTime = + "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"; namespace internal { class Call : public webrtc::Call, public PacketReceiver { public: @@ -80,7 +84,7 @@ class Call : public webrtc::Call, public PacketReceiver { DISALLOW_COPY_AND_ASSIGN(Call); }; -} // internal +} // namespace internal class TraceDispatcher : public TraceCallback { public: diff --git a/webrtc/call_tests.cc b/webrtc/video/call_tests.cc similarity index 99% rename from webrtc/call_tests.cc rename to webrtc/video/call_tests.cc index ec7d4ccb1..67e01131a 100644 --- a/webrtc/call_tests.cc +++ b/webrtc/video/call_tests.cc @@ -255,7 +255,7 @@ TEST_F(CallTest, UsesTraceCallback) { const unsigned int kReceiverTraceFilter = kTraceDefault & (~kTraceDebug); class TraceObserver : public TraceCallback { public: - TraceObserver(unsigned int filter) + explicit TraceObserver(unsigned int filter) : filter_(filter), messages_left_(50), done_(EventWrapper::Create()) {} virtual void Print(TraceLevel level, @@ -685,7 +685,7 @@ void CallTest::RespectsRtcpMode(newapi::RtcpMode rtcp_mode) { static const int kNumCompoundRtcpPacketsToObserve = 10; class RtcpModeObserver : public test::RtpRtcpObserver { public: - RtcpModeObserver(newapi::RtcpMode rtcp_mode) + explicit RtcpModeObserver(newapi::RtcpMode rtcp_mode) : test::RtpRtcpObserver(kDefaultTimeoutMs), rtcp_mode_(rtcp_mode), sent_rtp_(0), @@ -865,7 +865,7 @@ TEST_F(CallTest, SendsAndReceivesMultipleStreams) { class SyncRtcpObserver : public test::RtpRtcpObserver { public: - SyncRtcpObserver(int delay_ms) + explicit SyncRtcpObserver(int delay_ms) : test::RtpRtcpObserver(kLongTimeoutMs, delay_ms), critical_section_(CriticalSectionWrapper::CreateCriticalSection()) {} diff --git a/webrtc/loopback.cc b/webrtc/video/loopback.cc similarity index 99% rename from webrtc/loopback.cc rename to webrtc/video/loopback.cc index dc5afce42..e9b08931f 100644 --- a/webrtc/loopback.cc +++ b/webrtc/video/loopback.cc @@ -99,4 +99,4 @@ TEST_F(LoopbackTest, Test) { transport.StopSending(); } -} // webrtc +} // namespace webrtc diff --git a/webrtc/video/webrtc_video.gypi b/webrtc/video/webrtc_video.gypi index 82e17f461..5f7784d31 100644 --- a/webrtc/video/webrtc_video.gypi +++ b/webrtc/video/webrtc_video.gypi @@ -11,6 +11,7 @@ '<(webrtc_root)/video_engine/video_engine.gyp:video_engine_core', ], 'webrtc_video_sources': [ + 'video/call.cc', 'video/encoded_frame_callback_adapter.cc', 'video/encoded_frame_callback_adapter.h', 'video/transport_adapter.cc', diff --git a/webrtc/webrtc.gyp b/webrtc/webrtc.gyp index 03a7b69e8..7da4433c4 100644 --- a/webrtc/webrtc.gyp +++ b/webrtc/webrtc.gyp @@ -59,9 +59,7 @@ 'target_name': 'webrtc', 'type': 'static_library', 'sources': [ - 'call.cc', 'call.h', - 'config.cc', 'config.h', 'experiments.h', 'frame_callback.h', diff --git a/webrtc/webrtc_tests.gypi b/webrtc/webrtc_tests.gypi index 6ea74f432..6827fd5dd 100644 --- a/webrtc/webrtc_tests.gypi +++ b/webrtc/webrtc_tests.gypi @@ -19,7 +19,7 @@ 'target_name': 'video_loopback', 'type': 'executable', 'sources': [ - 'loopback.cc', + 'video/loopback.cc', 'test/test_main.cc', ], 'dependencies': [ @@ -32,12 +32,13 @@ 'target_name': 'video_engine_tests', 'type': '<(gtest_target_type)', 'sources': [ - 'call_tests.cc', + 'video/call_tests.cc', 'video/full_stack.cc', 'video/rampup_tests.cc', 'video/video_send_stream_tests.cc', 'voice_engine/test/auto_test/resource_manager.cc', 'voice_engine/test/auto_test/resource_manager.h', + 'test/common_unittest.cc', 'test/test_main.cc', ], 'dependencies': [