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Author SHA1 Message Date
jmarusic@webrtc.org
50604128db Method WebRtc_g722_encode that is eventually called always returns non-negative integer (internal counter) 
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34259004

Cr-Commit-Position: refs/heads/master@{#8428}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8428 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-19 12:16:49 +00:00
mflodman@webrtc.org
47d657b68e Remove Set/Get sending status from the default RTP module. 
This is now taken care of by the payload router and the calls to set_active.

BUG=769
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42379004

Cr-Commit-Position: refs/heads/master@{#8427}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8427 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-19 10:30:19 +00:00
glaznev@webrtc.org
30540fe722 Initialize RTPVideoHeader fields to correctly set simulcastIdx for non VP8 codecs. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39199004

Cr-Commit-Position: refs/heads/master@{#8421}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 20:30:18 +00:00
minyue@webrtc.org
c0bd7be0df Adding two new stats to VoiceReceiverInfo 
There have been requests of two new stats namely

speech_expand_rate and secondary_decoded_rate.

BUG=3867
R=henrik.lundin@webrtc.org, henrika@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40789004

Cr-Commit-Position: refs/heads/master@{#8415}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8415 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 15:24:39 +00:00
jmarusic@webrtc.org
b255865e6e The PCM codecs can never fail, so we don't need to check the return value 
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37329004

Cr-Commit-Position: refs/heads/master@{#8413}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8413 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 15:02:44 +00:00
henrik.lundin@webrtc.org
78619e2714 Revert of r8378 "Switch to using AudioEncoderIsac instead of ACMISAC" 
This is a speculative revert to try to isolate a memory issue.

BUG=chromium:459483,4228
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39189004

Cr-Commit-Position: refs/heads/master@{#8412}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8412 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 14:51:15 +00:00
henrik.lundin@webrtc.org
635838bd9b Re-implementing AcmOpusTest as AcmGenericCodecOpusTest 
The old AcmOpusTest depends on the ACMOpus class, but this class was
obsoleted by AudioEncoderOpus. In this CL, the test code is re-written
to use AudioEncoderOpus and ACMGenericCodecWrapper instead of
ACMOpus. Most of the test functionality is preserved, except for the
packet loss rate tests, which where already transferred to
AudioEncoderOpusTest in r8244.

R=kwiberg@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40029004

Cr-Commit-Position: refs/heads/master@{#8410}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 13:15:45 +00:00
magjed@webrtc.org
f68e186de3 Remove EnableMirroring and MirrorRenderStream 
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35239004

Cr-Commit-Position: refs/heads/master@{#8409}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8409 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 12:55:17 +00:00
sprang@webrtc.org
131bea89d6 Offline screenshare quality test, plus loopback. 
BUG=4171
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34109004

Cr-Commit-Position: refs/heads/master@{#8408}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 12:46:44 +00:00
kwiberg@webrtc.org
0521127779 AudioEncoder: Rename virtual accessors to CamelCase 
Although sample_rate_hz(), num_channels(), and rtp_timestamp_rate_hz()
are simple accessors for almost all implementations of AudioEncoder,
they are virtual and not guaranteed to be just simple accessors. Thus,
it makes more sense to use the normal CamelCase naming scheme.

BUG=4235
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34239004

Cr-Commit-Position: refs/heads/master@{#8407}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8407 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 12:01:13 +00:00
minyue@webrtc.org
7d721eea14 Adding speech_expand_rate to NetEQ Network Statistics. 
There have been requests for separating rate of expanded speech samples from noise samples.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37309004

Cr-Commit-Position: refs/heads/master@{#8404}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8404 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 10:02:20 +00:00
aluebs@webrtc.org
27669f320b Apply good settings to Beamformer 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33219004

Cr-Commit-Position: refs/heads/master@{#8398}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8398 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 19:24:37 +00:00
mflodman@webrtc.org
0abc6011b9 Remove SetCaptureDelay from the RTP module. 
This is a small step in getting rid of the default module, but also to
eventually delete FrameProviderBase completely.

BUG=769
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34229004

Cr-Commit-Position: refs/heads/master@{#8396}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8396 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 16:36:48 +00:00
stefan@webrtc.org
7663684258 Implement the Nada rmcat proposal within the simulation framework. 
This first CL focuses only on the bandwidth estimation parts of NADA, and doesn't contain the rate smoothing. It is still missing slow start functionality.

https://datatracker.ietf.org/doc/draft-zhu-rmcat-nada/

BUG=
R=sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35219004

Cr-Commit-Position: refs/heads/master@{#8395}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 16:04:17 +00:00
jmarusic@webrtc.org
71b35a4ce4 iLBC: Use uint8_t[] for byte arrays 
BUG=909

This is the same as https://review.webrtc.org/41779004/ with the review comments addressed.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40769004

Cr-Commit-Position: refs/heads/master@{#8394}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8394 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 16:02:46 +00:00
magjed@webrtc.org
640313ce4f WebRtcVideoCapturer: Remove dead code |OnIncomingCapturedEncodedFrame| 
The end goal except cleanup is to remove webrtc::VideoFrame.

R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36079004

Cr-Commit-Position: refs/heads/master@{#8393}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8393 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 15:10:41 +00:00
pbos@webrtc.org
a28a91d2f0 Fix data race for RTCPReceiver stats callback. 
Annotates the callback which identifies the bug, then fixes it.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/40009004

Cr-Commit-Position: refs/heads/master@{#8390}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 14:45:44 +00:00
magjed@webrtc.org
959dac7498 VideoCaptureImpl: Remove unused member variable |_capture_encoded_frame| 
The end goal except cleanup is to remove webrtc::VideoFrame.

R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37219004

Cr-Commit-Position: refs/heads/master@{#8388}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8388 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 13:44:46 +00:00
pbos@webrtc.org
4dd40d6b88 Signal threads for faster receiver destruction. 
Unblocks pending threads (render thread + decoder thread) when
destroying renderers and shutting down decoders.

Speeds up SetLocalDescription significantly (10x or so) under
WebRtcVideoEngine2 but also shutdown times in ~ViEChannel and
~ViEReceiver in general.

BUG=1788
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41959004

Cr-Commit-Position: refs/heads/master@{#8387}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 13:23:27 +00:00
minyue@webrtc.org
f9b5c1b3d0 Removing CELT. 
CELT is not supported in WebRTC/Libjingle. There are a few left-over in our code base. They are cleaned up in this CL.

BUG=
R=pbos@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36099004

Cr-Commit-Position: refs/heads/master@{#8385}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 12:37:14 +00:00
minyue@webrtc.org
2c1bcf2cb4 Adding decoded_fec_rate to NetEQ Network Statistics. 
A statistic is introduced to reflect the actual benefits of Opus FEC. It shows what percentage of samples in the rendered audio come from FEC data.

BUG=3867
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34969004

Cr-Commit-Position: refs/heads/master@{#8384}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8384 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 10:17:48 +00:00
mflodman@webrtc.org
290cb56dca Remove TimeToSendPacket and TimeToSendPadding from the default module. 
Thie CL moves the default RTP module logic for TimeToSendPacket and
TimeToSendPadding to PayloadRouter class and asserts on usage of the
default module.

BUG=769
TEST=New unittest.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33319004

Cr-Commit-Position: refs/heads/master@{#8383}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 10:15:47 +00:00
pbos@webrtc.org
86196c4f48 Setup encoders inexpensively before first frame. 
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
 2015-02-16 21:02:20 +00:00
henrik.lundin@webrtc.org
34509d9f33 Fix an issue with comfort noise in ACMGenericCodecWrapper 
In some cases it was not possible to set another payload type for CNG
than the default one. This CL fixes this. The problem was also
dependent on whether the comfort noise codec was registered before or
after the speech codec.

A test is implement to expose the bug, registering comfort noise at a
non-default payload type, and both before and after the speech codec.

BUG=4228
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35199004

Cr-Commit-Position: refs/heads/master@{#8380}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8380 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 16:02:40 +00:00
stefan@webrtc.org
e9f0f591b5 Enable bitrate probing by default in PacedSender. 
BUG=crbug:425925
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33359004

Cr-Commit-Position: refs/heads/master@{#8379}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8379 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 15:48:29 +00:00
henrik.lundin@webrtc.org
fbc347f2ef Re-land r8342 "Switch to using AudioEncoderIsac instead of ACMISAC"" 
This reverts r8372, with a bug fix: allowing zero rate in
AudioEncoderIsac::Config. Without this fix, setting the rate to zero
triggered a CHECK. Existing callers assumed that zero was a valid
value. Setting the rate to zero will result in the default rate 32000
being set.

BUG=4228,chromium:458638
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org
TBR=tina.legrand@webrtc.org
CC=jmarusic@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39159004

Cr-Commit-Position: refs/heads/master@{#8378}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8378 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 14:28:45 +00:00
kjellander@webrtc.org
ce22f13f0e GN: Changes for vp9, opus and direct trace 
Corresponding GN changes for
https://webrtc-codereview.appspot.com/34099004/

BUG=4185
R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/40669004

Cr-Commit-Position: refs/heads/master@{#8377}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8377 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 12:47:45 +00:00
kjellander@webrtc.org
e35fa96cbe Move isacfix.gypi and isac.gypi 
Move webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.gypi
and webrtc/modules/audio_coding/codecs/isac/main/source/isac.gypi to
webrtc/modules/audio_coding/codecs/isac to pass recently
added _CheckNoSourcesAboveGyp presubmit rule.

BUG=4002
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37269004

Cr-Commit-Position: refs/heads/master@{#8376}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8376 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 12:47:22 +00:00
sprang@webrtc.org
0200f70792 Set webrtc_rtp category to be disabled by default. 
Should not affect webrtc standalone. For chromium, disabling helps
mitigate viewing performance problems.

BUG=chromium:441440
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41909004

Cr-Commit-Position: refs/heads/master@{#8375}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8375 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 12:06:48 +00:00
stefan@webrtc.org
14b0279416 Break out code from bloated files in the BWE simulator. 
No changes to functionality.

BUG=4173
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34209004

Cr-Commit-Position: refs/heads/master@{#8374}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8374 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 12:03:19 +00:00
kjellander@webrtc.org
0f7f161ed6 Add audio_coding module OWNERS file. 
It should simplify things to have an
OWNERS file at the top level of audio_coding, in addition
to the lower ones.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39149004

Cr-Commit-Position: refs/heads/master@{#8373}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 09:53:47 +00:00
henrik.lundin@webrtc.org
4dc0003bed Revert r8342 "Switch to using AudioEncoderIsac instead of ACMISAC" 
BUG=chromium:458638
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33349004

Cr-Commit-Position: refs/heads/master@{#8372}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-14 09:42:41 +00:00
aluebs@webrtc.org
92a19bcbd7 Simplify mask calculation 
There are only 2 things that prevent the output to be bit-exact:
* The zero initialization of the postfilter_mask_ and high_pass_postfilter_mask_, which only afects the first blocks.
* The re-tuning of the target presence estimation, since only the bins between low_average_start_bin_ and high_average_end_bin_ are of interest.
This latter was not taken into account before.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35139004

Cr-Commit-Position: refs/heads/master@{#8368}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8368 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 19:38:22 +00:00
stefan@webrtc.org
56cb0ea99c Add support for bi-directional simulations by having both an uplink and a downlink. 
Partially make PacketReceiver a source which should be possible to hook up to the network.

Require that flow ids are set in all constructors.

BUG=4173
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36059004

Cr-Commit-Position: refs/heads/master@{#8367}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 15:46:52 +00:00
pbos@webrtc.org
d5ce2e63df Remove EventWrapper::Reset(). 
This simplifies the event wrapper which we've recently found issues in.
Also refactoring EndToEndTest.RespectsNetworkState to not depend on it.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41939004

Cr-Commit-Position: refs/heads/master@{#8366}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 14:58:38 +00:00
guoweis@webrtc.org
5a7dc39277 This is a code clean up. No functional change intended. 
Consolidate the enum for capturer/frame rotation we use through out the code base.

BUG=4145
R=mflodman@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39859004

Cr-Commit-Position: refs/heads/master@{#8365}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 14:32:13 +00:00
minyue@webrtc.org
a8cc3440b1 Allowing RED decoding for Opus. 
BUG=4247
TEST=reproduced and fixed the bug
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41809004

Cr-Commit-Position: refs/heads/master@{#8364}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8364 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 14:02:17 +00:00
solenberg@webrtc.org
2adf4c4edd Re-enable BWE tests using baseline files. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39119004

Cr-Commit-Position: refs/heads/master@{#8361}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8361 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 12:05:16 +00:00
henrika@webrtc.org
58f6f01acc WebRTC now compiles for enable_android_opensl=1. 
Default is enable_android_opensl=0 but we should build for OpenSL as well.

BUG=4293
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40719004

Cr-Commit-Position: refs/heads/master@{#8360}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8360 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 11:36:12 +00:00
bjornv@webrtc.org
ba97ea69f0 audio_coding/codec/ilbc: Removed usage of macro WEBRTC_SPL_MUL_16_16 
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)

The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)

Some other minor code cleanup also exists.

BUG=3348, 3353
TESTED=locally on Mac and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34179004

Cr-Commit-Position: refs/heads/master@{#8358}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8358 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 09:52:42 +00:00
mflodman@webrtc.org
2bd299a172 Remove call to RtpRtcp::RegisterSendPayload for the default RTP module. 
The send payload type is only checked in RTPSender::CheckPayloadType,
which in turn is only called from SendOutgoingData and never from the
default module anylonger.

BUG=769
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39949004

Cr-Commit-Position: refs/heads/master@{#8357}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 09:52:17 +00:00
henrik.lundin@webrtc.org
bb1219eca3 Add a unit test for callbacks with empty frames and fix bug in code 
This change adds a couple of new tests that verify that callbacks
with frame type kFrameEmpty are sent in between comfort noise packets.
This used to be the case until r8268, and with the fix included in
this CL is once again so.

COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37229004

Cr-Commit-Position: refs/heads/master@{#8353}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8353 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-12 15:53:55 +00:00
kjellander@webrtc.org
e01264306b Remove temporary GYP targets 
The Chromium libjingle.gyp has now been updated in
https://codereview.chromium.org/907343002/ and the changes
in https://webrtc-codereview.appspot.com/35099004/ are rolled
into Chromium. Therefore these targets are no longer needed.

BUG=
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41919004

Cr-Commit-Position: refs/heads/master@{#8352}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8352 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-12 13:48:39 +00:00
perkj@webrtc.org
a9eaeebc6a Fix problem where Android VoE can not record on multiple channels. 
The issue was introduced in https://webrtc-codereview.appspot.com/33969004/
R8325

TEST=  Build libjingle_peerconnection_android_unittest and then run "CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest"
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38089004

Cr-Commit-Position: refs/heads/master@{#8349}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-12 12:33:28 +00:00
pbos@webrtc.org
7c4d20fd6c Remove potential deadlock in RTPSenderAudio. 
Removes lock-order inversion formed by RTPSenderAudio->RTPSender calls
by doing a lot shorter locking which fetches a current state of
RTPSenderAudio variables before sending.

Thread annotates locked variables and removes one lock in
RTPSenderAudio, bonus fixes data races reported in voe_auto_test
--automated under TSan (DTMF data race).

Also includes some bonus cleanup of RTPSenderVideo which removes the
send critsect completely as all methods using it was always called
from RTPSender under its send_critsect.

R=henrik.lundin@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
BUG=3001, chromium:454654

Review URL: https://webrtc-codereview.appspot.com/41869004

Cr-Commit-Position: refs/heads/master@{#8348}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8348 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-12 12:20:50 +00:00
mflodman@webrtc.org
a4ef2ce29d Remove getting max payload length from default module. 
Moving functionality to get max payload length from default RTP module
to the payload router.

I'll make a follow up CL changing asserts to DCHECK in rtp_rtcp_impl.cc.

BUG=769
TEST=New unittest and existing sender mtu test
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36119004

Cr-Commit-Position: refs/heads/master@{#8345}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8345 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-12 09:55:05 +00:00
henrik.lundin@webrtc.org
76b4ac96cd Switch to using AudioEncoderIsac instead of ACMISAC 
This change switches from the old codec wrapper ACMISAC to the new
AudioEncoderIsac wrapped in an ACMGenericCodecWrapper.

This is also the CL where the old codec for producing redundancy (RED)
is inactivated. All RED payloads are now produces through the
AudioEncoderCopyRed or AudioEncoderIsacRed classes.

BUG=4228
TEST=Please, try the iSAC codec extensively.
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33249005

Cr-Commit-Position: refs/heads/master@{#8342}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8342 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-11 21:37:26 +00:00
henrik.lundin@webrtc.org
6c68c85b46 Switch to using AudioEncoderOpus instead of ACMOpus 
This change switches from the old codec wrapper ACMOpus to the new
AudioEncoderOpus wrapped in an ACMGenericCodecWrapper.

BUG=4228
TEST=Please, try the Opus codec extensively.
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33259004

Cr-Commit-Position: refs/heads/master@{#8341}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8341 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-11 21:34:06 +00:00
guoweis@webrtc.org
1226e926e6 CVO capturer feature: allow unrotated frame flows through the capture pipeline. 
split from https://webrtc-codereview.appspot.com/37029004/

This is based on clean up code change at https://webrtc-codereview.appspot.com/37129004

BUG=4145
R=perkj@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org, tommi@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8337

Committed: https://code.google.com/p/webrtc/source/detail?r=8338

Review URL: https://webrtc-codereview.appspot.com/39799004

Cr-Commit-Position: refs/heads/master@{#8339}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8339 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-11 18:38:53 +00:00
guoweis@webrtc.org
dc7b02277c CVO capturer feature: allow unrotated frame flows through the capture pipeline. 
split from https://webrtc-codereview.appspot.com/37029004/

This is based on clean up code change at https://webrtc-codereview.appspot.com/37129004

BUG=4145
R=perkj@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org, tommi@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8337

Review URL: https://webrtc-codereview.appspot.com/39799004

Cr-Commit-Position: refs/heads/master@{#8338}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8338 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-11 18:06:10 +00:00