- 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
This reverts commit 1c3e728aa9.
Reason: Fails test running on Nexus 9 bots - org.webrtc.VideoCapturerAndroidTest#testStartStopWithDifferentResolutions.
Note that all other tests pass so it seems like there is resolution supported by the device that can't use YV12.
TBR=glaznev@webrtc.org
BUG=4011
Review URL: https://webrtc-codereview.appspot.com/42389004
Cr-Commit-Position: refs/heads/master@{#8414}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8414 4adac7df-926f-26a2-2b94-8c16560cd09d
Instead of sending a black frame that's the size of the VideoFormat send
a black frame in the format we're already sending. This prevents
expensive encoder reconfiguration when the sending format is a different
resolution. This speeds up setting a null capturer (removing the
capturer) significantly as it doesn't entail an encoder reconfiguration.
R=mflodman@webrtc.org, pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/39179004
Cr-Commit-Position: refs/heads/master@{#8405}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8405 4adac7df-926f-26a2-2b94-8c16560cd09d
It’s possible to implement cricket::VideoFrame::CopyToBuffer using the virtual interface. This removes the need for subclasses to implement their own versions. This CL also fixes a bug in cricket::VideoFrame::CopyToPlanes which currently assumes that GetUPitch() == GetVPitch(), otherwise it may segfault.
I think this CL should land regardless, but the main purpose is to pave the way for for planned changes to I420VideoFrame. See https://review.webrtc.org/38879004.
R=fbarchard@google.com, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39889004
Cr-Commit-Position: refs/heads/master@{#8403}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8403 4adac7df-926f-26a2-2b94-8c16560cd09d
Modifies WebRtcVideoSendStream to use a default width/height of 16px.
This significantly reduces SetRemoteDescription time under
WebRtcVideoEngine2. Also preventing (expensive) reconfigurations due to
incoming frames when the channel is not sending yet.
Tests have been modified to generate a frame before expecting a certain
encoder size to have been configured.
Also adding tracing to WebRtcVideoSendStream::InputFrame as it can lead
to reconfigurations of the encoder which is expensive and it should show
up in chrome://tracing.
BUG=1788
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42369004
Cr-Commit-Position: refs/heads/master@{#8381}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8381 4adac7df-926f-26a2-2b94-8c16560cd09d
For now:
java_helpers - JNI convenience functions etc. Can in theory be moved to libjingle / webrtc general one day.
classreferenceholder - app/webrtc specific Java class loader.
androidvideocapturer_jni - the jni part of the video capturer I added.
peerconnection_jni - all the rest.
This also move all jni specifics into ns webrtc_jni to avoid naming collision.
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38099004
Cr-Commit-Position: refs/heads/master@{#8363}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8363 4adac7df-926f-26a2-2b94-8c16560cd09d
