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2014-12-04Update makefiles after merge of Chromium at 40.0.2214.27Ben Murdoch
This commit was generated by merge_from_chromium.py. Change-Id: I231a22a5117516d0892018e313b21fab26b1f615
2014-12-04Merge third_party/webrtc from ↵Ben Murdoch
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 71616dbb8cdd3ea13e0a964d18456ca3fe002dab This commit was generated by merge_from_chromium.py. Change-Id: I40a02c5091b60b596c558bfc0f03c06c63d7fb7a
2014-11-27Merge r7729 into M40 branch.kjellander@webrtc.org
BUG=chromium:436400 TBR=phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27259004 git-svn-id: http://webrtc.googlecode.com/svn/branches/40/webrtc@7759 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-12Merge third_party/webrtc from ↵Torne (Richard Coles)
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 60ab669c4c545b328b5c8b0453eb2cdecf851651 This commit was generated by merge_from_chromium.py. Change-Id: I7efe0f9a490c88eb3ec522902a351f9662e8753e
2014-11-07Remove partially defined WebRtcRTPHeader from Parse().pbos@webrtc.org
It' bit ugly that RtpDepacketizer::ParsedPayload partially defines WebRtcRTPHeader, and then sent to Parse() function for internal change. To make it clearer, the CL gets rid of using partially-defined WebRtcRTPHeader. BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28919004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7660 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Volume buttons in AppRTCDemo should affect output audio volume (part II).henrika@webrtc.org
See https://webrtc-codereview.appspot.com/32399004/ for part I. BUG=3279 TEST=AppRTC demo R=perkj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27059004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7654 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at b831a9e3d5f9f0563d249b726cffa8a070e58aee This commit was generated by merge_from_chromium.py. Change-Id: I6d6255972e3c34e7797e9b46fbc3c0fe7e552d43
2014-11-06Remove the state_ member from AudioDecoderkwiberg@webrtc.org
The subclasses that need a state pointer should declare them---with the right type, not void*, to get rid of all those casts. Two small but not quite trivial cleanups are included because they blocked the state_ removal: - AudioDecoderG722Stereo now inherits directly from AudioDecoder instead of being a subclass of AudioDecoderG722. - AudioDecoder now has a CngDecoderInstance member function, which is implemented only by AudioDecoderCng. This replaces the previous practice of calling AudioDecoder::state() and casting the result to a CNG_dec_inst*. It still isn't pretty, but now the blemish is plainly visible in the AudioDecoder class declaration. R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24169005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7644 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Adjust parameter in vp9 rate control test.marpan@webrtc.org
TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25099004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7638 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Increase speed setting for VP9 (from 5 to 6) and re-enable end_to_end test.marpan@webrtc.org
TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28949004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7637 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05Update makefiles after merge of Chromium at 5a645aa13b82Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I038f8684aa804c94ca2c5175bdeaf605bf0611c5
2014-11-05Wire up bandwidth stats to the new API and webrtcvideoengine2.stefan@webrtc.org
Adds stats to verify bandwidth and pacer stats. BUG=1788 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24969004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05Update makefiles after merge of Chromium at 2d0da5605d75Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I2b5db589b04e302cb1067fe730b81f3fb21b06bb
2014-11-05Fix android_clang build.glaznev@webrtc.org
BUG= R=kjellander@webrtc.org, tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30879004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7630 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05Revert 7623 "Remove the state_ member from AudioDecoder"niklas.enbom@webrtc.org
Breaks Chrome compile: e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(131) : error C3867: 'webrtc::NetEqImpl::InsertPacketInternal': function call missing argument list; use '&webrtc::NetEqImpl::InsertPacketInternal' to create a pointer to member e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(131) : error C3861: 'LOG_FERR1': identifier not found e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(152) : error C3867: 'webrtc::NetEqImpl::InsertPacketInternal': function call missing argument list; use '&webrtc::NetEqImpl::InsertPacketInternal' to create a pointer to member e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(152) : error C3861: 'LOG_FERR1': identifier not found e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(169) : error C3867: 'webrtc::NetEqImpl::GetAudioInternal': function call missing argument list; use '&webrtc::NetEqImpl::GetAudioInternal' to create a pointer to member ... > Remove the state_ member from AudioDecoder > > The subclasses that need a state pointer should declare them---with > the right type, not void*, to get rid of all those casts. > > Two small but not quite trivial cleanups are included because they > blocked the state_ removal: > > - AudioDecoderG722Stereo now inherits directly from AudioDecoder > instead of being a subclass of AudioDecoderG722. > > - AudioDecoder now has a CngDecoderInstance member function, which > is implemented only by AudioDecoderCng. This replaces the previous > practice of calling AudioDecoder::state() and casting the result > to a CNG_dec_inst*. It still isn't pretty, but now the blemish is > plainly visible in the AudioDecoder class declaration. > > R=henrik.lundin@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/24169005 TBR=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30879005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7629 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05Revert 7625 "Don't use DCHECK when you need the side effects..."niklas.enbom@webrtc.org
Reverting since 7623 might depend on this one > Don't use DCHECK when you need the side effects... > > R=pbos@webrtc.org > TBR=henrik.lundin@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/32369004 TBR=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7628 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 9a8c28f10f329c5ce91e77057933e60224000627 This commit was generated by merge_from_chromium.py. Change-Id: If9e805f5024e1fcdd99127626811f9650e109b1d
2014-11-04Don't use DCHECK when you need the side effects...kwiberg@webrtc.org
R=pbos@webrtc.org TBR=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32369004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7625 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Remove the state_ member from AudioDecoderkwiberg@webrtc.org
The subclasses that need a state pointer should declare them---with the right type, not void*, to get rid of all those casts. Two small but not quite trivial cleanups are included because they blocked the state_ removal: - AudioDecoderG722Stereo now inherits directly from AudioDecoder instead of being a subclass of AudioDecoderG722. - AudioDecoder now has a CngDecoderInstance member function, which is implemented only by AudioDecoderCng. This replaces the previous practice of calling AudioDecoder::state() and casting the result to a CNG_dec_inst*. It still isn't pretty, but now the blemish is plainly visible in the AudioDecoder class declaration. R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24169005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Reduce to 2 probes when probing for initial bandwidth.stefan@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23359005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7621 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Add UMA for measuring the diff between the BWE at 2 seconds compared to the ↵stefan@webrtc.org
BWE at 20 seconds when the BWE should have converged. BUG=crbug/425925 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30819005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7620 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Update makefiles after merge of Chromium at a99b7ad25d02Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I172fda810eb6cb37d17ba35571733f9eaeb9b230
2014-11-04Reworked paced sender queuesprang@webrtc.org
Packet queue in the paced sender is now based on a priority queue rather than having a separate fifo-queue per priority level. This allows more flexible sorting and cleaner usage. Packets with earlier capture times are now prioritized higher. In situations with high packet loss, the queue might contain packets from several subsequent frames. Retransmit packets from the earlier frames first, since the later ones will probably be dependent on these. Also, don't force sending of packets after a certain time of inactivity or when packets grow too old, since this was causing consistent overuse on poor connections. Instead, drop frames in vie encoder if pacer queue is too long. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27869004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7617 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Update makefiles after merge of Chromium at 30ec995cdb2dAndroid Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I77678e9f2e5044a6457f21cada6ee13b75fbfb0c
2014-11-04Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at a1fd19c12e4efdf4b8a5f92323070443d50dc34e This commit was generated by merge_from_chromium.py. Change-Id: Ibd48eca2d93e6324a2e886e451f27307aab45e9b
2014-11-04Fix problem with late packets in NetEqhenrik.lundin@webrtc.org
Since r7255, it could happen that an old packet would block the decoding process until enough packet was received for the buffer to flush. This CL fixes that by: - Partially reverting r7255; - Remove recent old packets before taking a decision for GetAudio; - Remove all old packets after a packet has been extracted for decoding; - Adding tests for reordered packets. BUG=chrome:423985 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25079004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7612 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16kwiberg@webrtc.org
R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26039004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7610 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Remove the useless dummy state parameter to WebRtcG711_*kwiberg@webrtc.org
R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27029004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7609 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Remove the codec_type_ member from AudioDecoderkwiberg@webrtc.org
It isn't actually required, as evidenced by the comparative ease with which it can be removed. R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31939004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7606 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Improving error message from neteq_rtpplayhenrik.lundin@webrtc.org
If a packet with unknown RTP payload type is inserted, this CL will make sure that the error message is a little more detailed and gives a better understadning of what to do. BUG=2692 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7603 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Update rate control parameter in vp9 test.marpan@webrtc.org
TBR=hellner@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7599 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Roll chromium_revision: 28d1981..d3db2ffmarpan@webrtc.org
Pick up the libvpx roll: https://codereview.chromium.org/674753002 Summary of changes (https://chromium.googlesource.com/chromium/src/+/28d1981..d3db2ff/DEPS): * third_party/android_tools 36bf7ac..ea50ccc * third_party/boringssl 7ea8481..751e889 * third_party/icu 8ac906f..d8b2a9d * third_party/libvpx efe9712..2e5ced5 * third_party/usrsctp/usrsctplib * tools/gyp 1990:1991 * tools/swarming_client a57d7db..bcb3bc3 Clang is not updated in this roll. Made the change getchar() --> getc(stdin) as seems like getchar() isn't supported on android anymore. (getchar() was causing the error: undefined reference to '__srget') Update rate control parameter in vp9 test. R=andrew@webrtc.org TBR=ajm@google.com Review URL: https://webrtc-codereview.appspot.com/23229004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7598 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03replace inline assembly WebRtcNsx_AnalysisUpdate by intrinsics.andrew@webrtc.org
The modification only uses the unique part of the analysis_update function. Pass byte to byte conformance test on both ARMv7 and AArch64, and the single function performance is similar with original assembly version on different platforms. If not specified, the code is compiled by GCC 4.6. The result is the "X version / C version" ratio, and the less is better. | run 100k times | cortex-a7 | cortex-a9 | cortex-a15 | | use C as the base on each | (1.2Ghz) | (1.0Ghz) | (1.7Ghz) | | CPU target | | | | |----------------------------+-----------+-----------+------------| | Neon asm | 15.61% | 20.15% | 14.89% | | Neon inline asm (LLVM 3.4) | 25.98% | 33.96% | 18.18% | | Neon intrinsics (GCC 4.6) | 22.06% | 27.01% | 19.24% | | Neon intrinsics (GCC 4.8) | 17.28% | 18.23% | 18.55% | | Neon intrinsics (LLVM 3.4) | 21.02% | 19.98% | 16.76% | BUG= R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28849004 Patch from Zhongwei Yao <zhongwei.yao@arm.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7596 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Add Opus support to neteq_rtpplayhenrik.lundin@webrtc.org
BUG=2416 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30859004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7595 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Add UMA metrics for the initial (after two seconds) packet loss, round-trip ↵stefan@webrtc.org
time and bandwidth estimate of a WebRTC call. BUG=crbug/425925 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7593 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Add stats for video:asapersson@webrtc.org
- number of sent/received RTCP NACK/FIR/PLI per minute - percentage of unique sent/received NACK requests - percentage of discarded/duplicated packets by the jitter buffer - permille of sent/received key frames BUG=crbug/419657 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24959004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7592 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Add more sanity checks to workaround the unidentified problem that ↵braveyao@webrtc.org
CaptureThread is still running while related resouces are destroyed already. BUG= TEST=auto test R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28859004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7590 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01Adjust/increase rate control thresold for a vp9 test.marpan@webrtc.org
TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25049004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7589 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01Add VP9 codec to VCM and vie_auto_test.marpan@webrtc.org
Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31replace inline assembly WebRtcNsx_SynthesisUpdateNeon by intrinsics.andrew@webrtc.org
The modification only uses the unique part of the synthesis_update function. Pass byte to byte conformance test both on ARMv7 and ARMv8, and the single function performance is similar with original assembly version on different platforms (if not specified, the code is compiled by GCC 4.6): | run 100k times | cortex-a7 | cortex-a9 | cortex-a15 | | use C as the base | (1.2Ghz) | (1.0Ghz) | (1.7Ghz) | | (the smaller the better) | | | | |----------------------------+-----------+-----------+------------| | C | 100% | 100% | 100% | | Neon asm | 15.93% | 17.01% | 12.50% | | Neon inline asm | 27.74% | 31.41% | 14.64% | | Neon intrinsics (GCC 4.8) | 17.84% | 14.10% | 13.84% | | Neon intrinsics (LLVM 3.4) | 16.63% | 14.01% | 12.98% | BUG=3580 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23159004 Patch from Zhongwei Yao <zhongwei.yao@arm.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7586 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Add a WavReader counterpart to WavWriter.andrew@webrtc.org
Don't bother with a C interface as we currently have no need to call this from C code. The first use will be in the audioproc tool. R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7585 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Update makefiles after merge of Chromium at a41c404b1c7fAndroid Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: Ifcab5d7c5bd698b1a0a72100960585183048352d
2014-10-31Update makefiles after merge of Chromium at b210e2d62956Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I3b99d06f861694a90ee0f32a97380e1c99cfaa07
2014-10-31Update all .isolate files for the new format.kjellander@webrtc.org
R=kjellander@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/27809004 Patch from Marc-Antoine Ruel <maruel@chromium.org>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7583 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Fix N7 camera aspect ratio.glaznev@webrtc.org
N7 video preview generates stretched output: https://code.google.com/p/android/issues/detail?id=70830. To workaround the problem set camera picture size in addition to video preview size with the same resolution. BUG=3971 R=braveyao@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25009004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7581 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Build fix for MIPS32R6.andrew@webrtc.org
Exclude MIPS optimizations for MIPS32R6 build since some of the instructions are not supported. This is temporary fix, until the MIPS32R6 code is added. R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25989004 Patch from Ljubomir Papuga <lpapuga@mips.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7580 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 1a02faa335e7d8076b5cf8dd9a584e72669b0c8e This commit was generated by merge_from_chromium.py. Change-Id: I6a99300dc0ba646ed641a90356f632d782c83dbe
2014-10-31Simplify bwe tests.stefan@webrtc.org
R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7576 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Revert "Revert part of r7561, "Refactor audio conversion functions.""andrew@webrtc.org
This restores the conversion changes to AudioProcessing originally added in r7561, with minor alterations to ensure it passes all tests. TBR=kwiberg Review URL: https://webrtc-codereview.appspot.com/28899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7574 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30Add 15 fps support for Android devices with missing 15 fpsglaznev@webrtc.org
camera mode. Some latest Android devices support only 30 fps for front camera, but HW VP8 encoder performance is not enough for 720p 30 fps encoding. Add 15 fps support for these devices by allowing frame drop in Android camera wrapper. BUG= R=tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7571 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-26 02:51:44 +0000