Mostly, it's about moving constructors and descructors to the .cc
files, so that they won't be inlined everywhere.
The reason this CL is so big is that a lot of code was using
common_types.h without declaring a dependency on webrtc_common, which
broke the build once common_types.h started to depend on
common_types.cc.
BUG=163
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26089004
Cr-Commit-Position: refs/heads/master@{#8516}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
Peer connection may generate answer and ICE candidates before
websocket client is registered. Remove check from sendAnswer()
and sendLocalIceCandidate() functions and allow websocket client
to accumulate messages and send them later once it will be
registered.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44379004
Cr-Commit-Position: refs/heads/master@{#8508}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8508 4adac7df-926f-26a2-2b94-8c16560cd09d
Add some const safety by DCHECK(HasOneRef()) in non-const GetYPlane. This CL also replaces all incorrect non-const calls with const calls for pixel data access in cricket::VideoFrame. It's easy to call the non-const version of e.g. GetYPlane by mistake, even if only const-access is needed. For example:
const scoped_ptr<cricket::VideoFrame> foo;
const uint8_t* y = foo->GetYPlane();
will actually call the non-const version of GetYPlane.
R=mflodman@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39079004
Cr-Commit-Position: refs/heads/master@{#8507}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8507 4adac7df-926f-26a2-2b94-8c16560cd09d
It looks like it is being used single threadedly, except that in some cases it is created and/or destroyed in threads other than the one running its operations. As such, CaptureManager() contains 'thread_checker_.DetachFromThread()' and ~CaptureManager() does not have a DCHECK.
BUG=
R=perkj@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36279004
Cr-Commit-Position: refs/heads/master@{#8498}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8498 4adac7df-926f-26a2-2b94-8c16560cd09d
With this cl, video buffers are now allocated using direct buffers.
These buffers are guaranteed to live as long as the capturer is running.
We can now post frames in c++ from the Java thread to the c++ worker thread and let c++ post the buffers back when it has finished
processing them.
This cl also reverts back to make Stop asynchronouse so that it is guaranteed that the c++ worker thread is not used and no frames are delivered to VideoCapturerAndroid after Stop completes.
BUG=4318
TESTED= On a N5, N6, N9 and Samsung device.
R=glaznev@webrtc.org, magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43369004
Cr-Commit-Position: refs/heads/master@{#8493}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8493 4adac7df-926f-26a2-2b94-8c16560cd09d
- Qualcomm H.264 HW decoder on KK and older requires
a few video frames before it can generate output. Increase
maximum allowed pending frames for H.264 decoder to 30.
Plus changes in the logging to track decoder buffers
timestamps.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36319004
Cr-Commit-Position: refs/heads/master@{#8490}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8490 4adac7df-926f-26a2-2b94-8c16560cd09d
- Add a setting option to disable outgoing video in a call.
- Add an option to select audio codec.
- Add an option to specify audio bitrate for Opus codec.
- Plus add an option to select H.264 as default video codec.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42449004
Cr-Commit-Position: refs/heads/master@{#8468}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8468 4adac7df-926f-26a2-2b94-8c16560cd09d
* Move constants into the files/functions that use them
* Declare variables in the narrowest scope possible
* Use correct (expected, actual) order for gtest macros
* Remove unused functions
* Untabify
* 80-column limit
* Avoid C-style casts
* Prefer true typed constants to "enum hack" constants
* Print size_t using the right format macro
* Shorten and simplify code
* Other random cleanup bits and style fixes
BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36179004
Cr-Commit-Position: refs/heads/master@{#8467}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
- Allow to configure MediaCodec Java wrapper to use VP8
and H.264 codec.
- Save H.264 config frames with SPS and PPS NALUs and append them to every key frame.
- Correctly handle the case when one encoded frame may generate several output NALUs.
- Add code to find H.264 start codes.
- Add a flag (non configurable yet) to use H.264 in AppRTCDemo.
- Improve MediaCodec logging.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43379004
Cr-Commit-Position: refs/heads/master@{#8465}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8465 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL is part II in a major refactoring effort. See https://webrtc-codereview.appspot.com/33969004 for part I.
- Removes unused code and old WEBRTC logging macros
- Now uses optimal sample rate and buffer size in Java AudioTrack (used hard-coded sample rate before)
- Makes code more inline with the implementation in Chrome
- Adds helper methods for JNI handling to improve readability
- Changes the threading model (high-prio audio thread now lives in Java-land and C++ only works as proxy)
- Simplified the delay estimate
- Adds basic thread checks
- Removes all locks in C++ land
- Removes all locks in Java
- Improves construction/destruction
- Additional cleanup
Tested using AppRTCDemo and WebRTCDemo APKs on N6, N5, N7, Samsung Galaxy S4 and
Samsung Galaxy S4 mini (which uses 44.1kHz as native sample rate).
BUG=NONE
R=magjed@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39169004
Cr-Commit-Position: refs/heads/master@{#8460}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8460 4adac7df-926f-26a2-2b94-8c16560cd09d
The Codec class is a big class and objects of the Codec class are passed
around by value. That means that inlined operations would be duplicated
at many places, in particular inside STL.
By not inlining Codec methods, webrtc shrinks by 50 KB in
a Linux x64 clang build.
Total change: -54147 bytes
==========================
+2810 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/base/codec.cc - (gained 2920, lost 110)
-1003 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/base/codec.h - (gained 0, lost 1003)
-1129 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/sctp/sctpdataengine.cc - (gained 1660, lost 2789)
-1190 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/base/rtpdataengine.cc - (gained 1408, lost 2598)
-1747 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/session/media/mediasession.cc - (gained 803, lost 2550)
-2141 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine.cc - (gained 1679, lost 3820)
-2250 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/app/webrtc/webrtcsdp.cc - (gained 1224, lost 3474)
-2927 - Source: /usr/include/c++/4.8/bits/stl_vector.h - (gained 0, lost 2927)
-3729 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine2.cc - (gained 10925, lost 14654)
-6369 - Source: /usr/include/c++/4.8/bits/vector.tcc - (gained 0, lost 6369)
-10582 - Source: /usr/include/c++/4.8/bits/stl_heap.h - (gained 0, lost 10582)
-19324 - Source: /usr/include/c++/4.8/bits/stl_algo.h - (gained 743, lost 20067)
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40729005
Cr-Commit-Position: refs/heads/master@{#8436}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8436 4adac7df-926f-26a2-2b94-8c16560cd09d
Fetching the current codec for sake of gathering stats, is frequently blocked since it's done by acquiring the same lock as is held while encoding frames. This can mean tens of milliseconds.
To improve this, I'm taking advantage of the fact that the codec information is set on the same thread as is used to query the information. This means that locking isn't needed for querying this information. I'm adding checks to make sure debug builds will crash if this isn't followed.
An alternative to this approach could be to add one more lock that is specifically used for the codec information variable. This would also decouple querying codec information from the encoder itself, but still requires a lock.
This patch depends on making ThreadChecker part of rtc_base_approved:
https://webrtc-codereview.appspot.com/40539004/
BUG=2822
R=mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37779004
Cr-Commit-Position: refs/heads/master@{#8435}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8435 4adac7df-926f-26a2-2b94-8c16560cd09d
