- 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
ACMGenericCodecWrapper was the only remaining subclass of
ACMGenericCodec, and was the only class that was ever instantiated.
This CL merges the two, essentially keeping the function implementations
from ACMGenericCodecWrapper except where the base class's code was
invoked.
As it turns out, a lot of functions were never used, but in some cases
they were refernced in AudioCodingModuleImpl. In these cases, the
referencing code is commented out and marked FATAL(). This will be
further cleaned up in follow-up CLs.
BUG=4228
COAUTHOR=kwiberg@webrtc.orgR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38209004
Cr-Commit-Position: refs/heads/master@{#8463}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8463 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
They have all been replaced by AudioEncoder subclasses, accessed throgh
ACMGenericCodecWrapper objects. After this change, the only subclass of
ACMGenericCodec is ACMGenericCodecWrapper. (The two will be consolidated
in a future cl.)
This CL also deletes acm_opus_unittest.cc. This test file was already
replaced audio_encoder_opus_unittest.cc in r8244.
BUG=4228
COAUTHOR=kwiberg@webrtc.orgR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40729004
Cr-Commit-Position: refs/heads/master@{#8457}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8457 4adac7df-926f-26a2-2b94-8c16560cd09d
This cl just moves the logic form the default module
SetTargetSendBitrates to PayloadRouter. There might be glitch / mismatch
in size between trate the vector and rtp modules. This was the same in
the default module and is quite hard to protect from before we have the
new video API.
I also removed some test form rtp_rtcp_impl_unittest that were affected
by this change. The test tests code that isn't implemented, hence the
DISABLED_, and this will never be implemented in the RTP module, rather
the payload router in the future.
BUG=769
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42419004
Cr-Commit-Position: refs/heads/master@{#8453}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8453 4adac7df-926f-26a2-2b94-8c16560cd09d
There is a potential race when deleting a channel and there is a frame
in the encoder. ViEEncoder::SendData can be called after
ViEEncoder::StopThreadsAndRemovePayloadRouter and payload_router is
then already removed.
Until we have the new API in place, use scoped_refptr in ViEChannel and
ViEEncoder and deregister channel/encoder before deleting.
BUG=769
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42019004
Cr-Commit-Position: refs/heads/master@{#8443}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8443 4adac7df-926f-26a2-2b94-8c16560cd09d
* Don't use ConjugateDotProduct to calculate the norm.
* Only resize Matrix when needed.
This makes the Beamformer run in 93.6% the original time.
The error between the new and original output is really small and is caused by the new norm calculation.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37339004
Cr-Commit-Position: refs/heads/master@{#8438}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8438 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
When an "empty frame" was produced by ACMGenericCodecWrapper::Encode,
the timestamp value was not set. This is now fixed, and the first byte
of the bitstream is set to something as well to avoid similar problems.
BUG=chromium:459483
R=jmarusic@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34269004
Cr-Commit-Position: refs/heads/master@{#8431}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8431 4adac7df-926f-26a2-2b94-8c16560cd09d
