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186 Commits

Author SHA1 Message Date
kjellander
dd460e2aa2 Fix lint errors to enable stricter PyLint rules 
These fixes are needed to avoid errors after submitting
https://codereview.webrtc.org/2737963003

BUG=webrtc:7303
NOTRY=True

Review-Url: https://codereview.webrtc.org/2812273002
Cr-Commit-Position: refs/heads/master@{#17679}
 2017-04-12 19:06:13 +00:00
oprypin
f250100475 Add POLQA to low bandwidth audio test 
BUG=webrtc:7229

Review-Url: https://codereview.webrtc.org/2804083003
Cr-Commit-Position: refs/heads/master@{#17671}
 2017-04-12 12:00:56 +00:00
minyue
20c84ccd48 Making FakeNetworkPipe demux audio and video packets. 
BUG=None

Review-Url: https://codereview.webrtc.org/2794243002
Cr-Commit-Position: refs/heads/master@{#17629}
 2017-04-10 23:57:57 +00:00
hbos
8d609f6b6d Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) 
Reason for revert:
Re-land, reverting did not fix bug.

https://bugs.chromium.org/p/webrtc/issues/detail?id=7465

Original issue's description:
> Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
>
> Reason for revert:
> Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
>
> https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
>
> Original issue's description:
> > Added the GetSources() to the RtpReceiverInterface and implemented
> > it for the AudioRtpReceiver.
> >
> > This method returns a vector of RtpSource(both CSRC source and SSRC
> > source) which contains the ID of a source, the timestamp, the source
> > type (SSRC or CSRC) and the audio level.
> >
> > The RtpSource objects are buffered and maintained by the
> > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> > the info of the contributing source will be pulled along the object
> > chain:
> > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> > AudioReceiveStream -> voe::Channel -> RtpRtcp module
> >
> > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
> >
> > BUG=chromium:703122
> > TBR=stefan@webrtc.org, danilchap@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2770233003
> > Cr-Commit-Position: refs/heads/master@{#17591}
> > Committed: 292084c376
>
> TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:703122
>
> Review-Url: https://codereview.webrtc.org/2809613002
> Cr-Commit-Position: refs/heads/master@{#17616}
> Committed: fbcc5cb386

TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:703122

Review-Url: https://codereview.webrtc.org/2810623003
Cr-Commit-Position: refs/heads/master@{#17621}
 2017-04-10 14:39:05 +00:00
kwiberg
37e99fd3fa Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ 
AudioDecoder and AudioDecoderFactory are in webrtc/api/ now, so move
their mocks to someplace central where tests from all over WebRTC are
allowed to #include them.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2798063004
Cr-Commit-Position: refs/heads/master@{#17619}
 2017-04-10 12:15:48 +00:00
olka
fbcc5cb386 Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) 
Reason for revert:
Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see

https://bugs.chromium.org/p/webrtc/issues/detail?id=7465

Original issue's description:
> Added the GetSources() to the RtpReceiverInterface and implemented
> it for the AudioRtpReceiver.
>
> This method returns a vector of RtpSource(both CSRC source and SSRC
> source) which contains the ID of a source, the timestamp, the source
> type (SSRC or CSRC) and the audio level.
>
> The RtpSource objects are buffered and maintained by the
> RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> the info of the contributing source will be pulled along the object
> chain:
> AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> AudioReceiveStream -> voe::Channel -> RtpRtcp module
>
> Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
>
> BUG=chromium:703122
> TBR=stefan@webrtc.org, danilchap@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2770233003
> Cr-Commit-Position: refs/heads/master@{#17591}
> Committed: 292084c376

TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:703122

Review-Url: https://codereview.webrtc.org/2809613002
Cr-Commit-Position: refs/heads/master@{#17616}
 2017-04-10 11:38:13 +00:00
stefan
fca900aa37 Fix two invalid DCHECKs related to audio BWE. 
These are invalid since:
- An allocated bitrate of 0 means that the stream should be disabled. Changing the behavior to send audio at min bitrate even though the allocator asks for the stream to be disabled.
- It should be OK to set a min bitrate equal to the start bitrate.

BUG=webrtc:5079

Review-Url: https://codereview.webrtc.org/2806163003
Cr-Commit-Position: refs/heads/master@{#17613}
 2017-04-10 10:53:00 +00:00
zhihuang
292084c376 Added the GetSources() to the RtpReceiverInterface and implemented 
it for the AudioRtpReceiver.

This method returns a vector of RtpSource(both CSRC source and SSRC
source) which contains the ID of a source, the timestamp, the source
type (SSRC or CSRC) and the audio level.

The RtpSource objects are buffered and maintained by the
RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
the info of the contributing source will be pulled along the object
chain:
AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
AudioReceiveStream -> voe::Channel -> RtpRtcp module

Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource

BUG=chromium:703122
TBR=stefan@webrtc.org, danilchap@webrtc.org

Review-Url: https://codereview.webrtc.org/2770233003
Cr-Commit-Position: refs/heads/master@{#17591}
 2017-04-07 17:57:22 +00:00
oprypin
abd101b91f Support multiple connected Android devices in low bandwidth audio test 
Previously it was assumed that exactly one device is connected.
Now adb will get an argument with the device ID taken from the runner
script's stdout.

BUG=webrtc:7229
TBR=kjellander@webrtc.org
NOTRY=true

Review-Url: https://codereview.webrtc.org/2783343003
Cr-Commit-Position: refs/heads/master@{#17580}
 2017-04-07 06:21:30 +00:00
nisse
fdbfdc9786 Let PacketRouter separate send and receive modules. 
This is in preparation for merging the ViERemb logic in packet_router,
to send REMB feedback as sender reports if possible, otherwise as
receiver reports.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2774623006
Cr-Commit-Position: refs/heads/master@{#17489}
 2017-03-31 12:44:52 +00:00
oprypin
6d305baa04 Add Windows, Mac, Android support to low bandwidth audio test 
BUG=webrtc:7229

Review-Url: https://codereview.webrtc.org/2767383005
Cr-Commit-Position: refs/heads/master@{#17470}
 2017-03-30 11:01:30 +00:00
nisse
e5ad5ca06a Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ ) 
Reason for revert:
Intend to fix perf failures and reland.

Original issue's description:
> Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ )
>
> Reason for revert:
> Reverting since this seems to break multiple WebRTC Perf buildbots
>
> Original issue's description:
> > Don't hardcode MediaType::ANY in FakeNetworkPipe.
> >
> > Instead let each test set the appropriate media type. This simplifies
> > demuxing in Call and later in RtpTransportController.
> >
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/2774463003
> > Cr-Commit-Position: refs/heads/master@{#17418}
> > Committed: 9c47b00e24
>
> TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2784543002
> Cr-Commit-Position: refs/heads/master@{#17427}
> Committed: 3a3bd50610

TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,lliuu@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2783853002
Cr-Commit-Position: refs/heads/master@{#17459}
 2017-03-30 06:57:43 +00:00
lliuu
3a3bd50610 Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ ) 
Reason for revert:
Reverting since this seems to break multiple WebRTC Perf buildbots

Original issue's description:
> Don't hardcode MediaType::ANY in FakeNetworkPipe.
>
> Instead let each test set the appropriate media type. This simplifies
> demuxing in Call and later in RtpTransportController.
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2774463003
> Cr-Commit-Position: refs/heads/master@{#17418}
> Committed: 9c47b00e24

TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2784543002
Cr-Commit-Position: refs/heads/master@{#17427}
 2017-03-28 16:40:59 +00:00
nisse
9c47b00e24 Don't hardcode MediaType::ANY in FakeNetworkPipe. 
Instead let each test set the appropriate media type. This simplifies
demuxing in Call and later in RtpTransportController.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2774463003
Cr-Commit-Position: refs/heads/master@{#17418}
 2017-03-28 11:59:41 +00:00
elad.alon
4e7645118e Fix UT failure by temporarily uncommenting 
BUG=webrtc:7322, webrtc:7405

Review-Url: https://codereview.webrtc.org/2780473002
Cr-Commit-Position: refs/heads/master@{#17393}
 2017-03-27 15:53:11 +00:00
kwiberg
1c07c70d88 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" 
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2774833003
Cr-Commit-Position: refs/heads/master@{#17391}
 2017-03-27 14:15:49 +00:00
nisse
b8f9a32459 Define RtpTransportControllerSendInterface. 
Implementation owned by call, and passed to VideoSendStream and
AudioSendStream.

BUG=webrtc:6847, webrtc:7135

Review-Url: https://codereview.webrtc.org/2685673003
Cr-Commit-Position: refs/heads/master@{#17389}
 2017-03-27 12:36:15 +00:00
kwiberg
670a7f3611 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) 
Reason for revert:
Makes perf and Chromium FYI bots unhappy.

Original issue's description:
> WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
>
> This removes one more place where we were unable to handle codecs not
> in the built-in set.
>
> BUG=webrtc:5805
>
> Review-Url: https://codereview.webrtc.org/2686043006
> Cr-Commit-Position: refs/heads/master@{#17370}
> Committed: 1724cfbdba

TBR=ossu@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2772043002
Cr-Commit-Position: refs/heads/master@{#17374}
 2017-03-24 12:56:21 +00:00
kwiberg
1724cfbdba WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType 
This removes one more place where we were unable to handle codecs not
in the built-in set.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2686043006
Cr-Commit-Position: refs/heads/master@{#17370}
 2017-03-24 10:16:04 +00:00
elad.alon
dadb4dc3c9 Allow ANA to receive RPLR (recoverable packet loss rate) indications 
This is part of a series of CLs. Next CLs:
1. CL for RPLR-based FecController
2. CL for allowing experiment-driven configuration of the above (through both field-trials and protobuf)

BUG=webrtc:7058

Review-Url: https://codereview.webrtc.org/2661043003
Cr-Commit-Position: refs/heads/master@{#17368}
 2017-03-23 22:29:50 +00:00
elad.alon
d12a8e1c8e Attach TransportFeedbackPacketLossTracker to ANA (PLR only) 
This CL is one in a series. To finish the work, the following CLs will be added:
1. CL for connecting RPLR as well
2. CL for RPLR-based FecController
3. CL for allowing experiment-driven configuration of the above (through both field-trials and protobuf)

BUG=webrtc:7058

Review-Url: https://codereview.webrtc.org/2638083002
Cr-Commit-Position: refs/heads/master@{#17365}
 2017-03-23 18:04:48 +00:00
oprypin
92220ffe9f Low-bandwidth audio testing 
The C++ part of the test uses CallTest to set up an audio-only call. It reads an audio file, plays it through a FakeAudioDevice which transfers data through a FakeNetworkPipe for another FakeAudioDevice to receive it and write it to a file. Information about these files is printed to stdout.

The test cases are meant to try different network and audio configs (more are planned in the future).

The Python part of the test runs the C++ part and scans stdout for tests to perform, runs the pairs of files (original and degraded) through the PESQ tool to receive a score and writes that to perf dashboard.

BUG=webrtc:7229
NOTRY=True

Review-Url: https://codereview.webrtc.org/2694203002
Cr-Commit-Position: refs/heads/master@{#17356}
 2017-03-23 10:40:03 +00:00
nisse
40854eab23 Reland of Delete class MockCongestionController. (patchset #1 id:1 of https://codereview.webrtc.org/2762133003/ ) 
Reason for revert:
Problem was the rename of the include file. Intend to keep the old name for now, and then reland.

Original issue's description:
> Revert of Delete class MockCongestionController. (patchset #4 id:60001 of https://codereview.webrtc.org/2762023004/ )
>
> Reason for revert:
> This is breaking downstream build.
>
> Original issue's description:
> > Delete class MockCongestionController.
> >
> > It became unused in cl https://codereview.webrtc.org/2516983004.
> >
> > BUG=None
> >
> > Review-Url: https://codereview.webrtc.org/2762023004
> > Cr-Commit-Position: refs/heads/master@{#17325}
> > Committed: d19bcb7116
>
> TBR=stefan@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2762133003
> Cr-Commit-Position: refs/heads/master@{#17330}
> Committed: e27f1e764e

TBR=stefan@webrtc.org,skvlad@webrtc.org
BUG=None

Review-Url: https://codereview.webrtc.org/2766133002
Cr-Commit-Position: refs/heads/master@{#17338}
 2017-03-22 10:28:54 +00:00
skvlad
e27f1e764e Revert of Delete class MockCongestionController. (patchset #4 id:60001 of https://codereview.webrtc.org/2762023004/ ) 
Reason for revert:
This is breaking downstream build.

Original issue's description:
> Delete class MockCongestionController.
>
> It became unused in cl https://codereview.webrtc.org/2516983004.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2762023004
> Cr-Commit-Position: refs/heads/master@{#17325}
> Committed: d19bcb7116

TBR=stefan@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2762133003
Cr-Commit-Position: refs/heads/master@{#17330}
 2017-03-21 20:15:59 +00:00
nisse
d19bcb7116 Delete class MockCongestionController. 
It became unused in cl https://codereview.webrtc.org/2516983004.

BUG=None

Review-Url: https://codereview.webrtc.org/2762023004
Cr-Commit-Position: refs/heads/master@{#17325}
 2017-03-21 16:41:07 +00:00
nisse
559af38a15 Split CongestionController into send- and receive-side classes. 
New class ReceiveSideCongestionController, extracted from CongestionController, and responsible for the
OnReceivedPacket processing.

Rest of the CongestionController moved to a new class
SendSideCongestionController.

To avoid breaking applications, CongestionController is redefined
as a union of these two classes, with no intended change in behavior.

With one exception: CongestionController::SetBweBitrates used to call
remote_bitrate_estimator_.SetMinBitrate, but after remote_bitrate_estimator_ was moved to ReceiveSideCongestionController,
it no longer does this.

BUG=webrtc:6847

Review-Url: https://codereview.webrtc.org/2752233002
Cr-Commit-Position: refs/heads/master@{#17321}
 2017-03-21 13:41:12 +00:00
oprypin
5e1ca78705 Add low_bandwidth_audio_test to default build 
https://codereview.webrtc.org/2717683002/ added a placeholder binary but
it still needs to be included in the default build before enabling it on
bots.

BUG=webrtc:7229

Review-Url: https://codereview.webrtc.org/2746403002
Cr-Commit-Position: refs/heads/master@{#17302}
 2017-03-20 09:06:18 +00:00
elad.alon
5bbf43f9d4 Move delay_based_bwe_ into CongestionController 
BUG=webrtc:7058

Review-Url: https://codereview.webrtc.org/2725823002
Cr-Commit-Position: refs/heads/master@{#17146}
 2017-03-09 14:40:08 +00:00
oprypin
67fdb80837 Reland of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #1 id:1 of https://codereview.webrtc.org/2739143002/ ) 
Reason for revert:
Can reland it if backwards compatible API is kept.

Original issue's description:
> Revert of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #4 id:180001 of https://codereview.webrtc.org/2683033004/ )
>
> Reason for revert:
> The API change in audio/utility/audio_frame_operations.h caused breakage. Need to keep backward-compatible API.
>
> Original issue's description:
> > Enable cpplint and fix cpplint errors in webrtc/*audio
> >
> > Change usage accordingly throughout the codebase
> >
> > BUG=webrtc:5268
> >
> > TESTED=Fixed issues reported by:
> > find webrtc/*audio -type f -name *.cc -o -name *.h | xargs cpplint.py
> >
> > Review-Url: https://codereview.webrtc.org/2683033004
> > Cr-Commit-Position: refs/heads/master@{#17133}
> > Committed: aebe55ca6c
>
> TBR=henrika@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5268
>
> Review-Url: https://codereview.webrtc.org/2739143002
> Cr-Commit-Position: refs/heads/master@{#17138}
> Committed: e47c1d3ca1

TBR=henrika@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
BUG=webrtc:5268

Review-Url: https://codereview.webrtc.org/2739073003
Cr-Commit-Position: refs/heads/master@{#17144}
 2017-03-09 14:25:06 +00:00
oprypin
e47c1d3ca1 Revert of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #4 id:180001 of https://codereview.webrtc.org/2683033004/ ) 
Reason for revert:
The API change in audio/utility/audio_frame_operations.h caused breakage. Need to keep backward-compatible API.

Original issue's description:
> Enable cpplint and fix cpplint errors in webrtc/*audio
>
> Change usage accordingly throughout the codebase
>
> BUG=webrtc:5268
>
> TESTED=Fixed issues reported by:
> find webrtc/*audio -type f -name *.cc -o -name *.h | xargs cpplint.py
>
> Review-Url: https://codereview.webrtc.org/2683033004
> Cr-Commit-Position: refs/heads/master@{#17133}
> Committed: aebe55ca6c

TBR=henrika@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5268

Review-Url: https://codereview.webrtc.org/2739143002
Cr-Commit-Position: refs/heads/master@{#17138}
 2017-03-09 10:43:31 +00:00
oprypin
aebe55ca6c Enable cpplint and fix cpplint errors in webrtc/*audio 
Change usage accordingly throughout the codebase

BUG=webrtc:5268

TESTED=Fixed issues reported by:
find webrtc/*audio -type f -name *.cc -o -name *.h | xargs cpplint.py

Review-Url: https://codereview.webrtc.org/2683033004
Cr-Commit-Position: refs/heads/master@{#17133}
 2017-03-09 09:05:33 +00:00
kjellander
8f8d1a06b9 Adding placeholder for low bandwidth audio test 
This will allow the trybots to be updated to start running this new test
executable, so that they can be used when landing this CL which will
replace the dummy test with real code:
https://codereview.webrtc.org/2694203002

Most likely, the trybots will just run the test binary while the perf bots
will run a Python wrapper script that takes care of the post-processing
to calculate audio quality using PESQ.

BUG=webrtc:7229
NOTRY=True

Review-Url: https://codereview.webrtc.org/2717683002
Cr-Commit-Position: refs/heads/master@{#17063}
 2017-03-06 12:01:16 +00:00
elad.alon
fb1fa44d70 Remove MockRemoteBitrateObserver (unused) 
BUG=None

Review-Url: https://codereview.webrtc.org/2731523002
Cr-Commit-Position: refs/heads/master@{#17060}
 2017-03-06 11:48:14 +00:00
solenberg
796b8f9d71 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. 
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2721003002
Cr-Commit-Position: refs/heads/master@{#16956}
 2017-03-02 01:02:23 +00:00
jens.nielsen
228c268065 Support 4 channel mic in Windows Core Audio 
BUG=webrtc:7220

Review-Url: https://codereview.webrtc.org/2712743004
Cr-Commit-Position: refs/heads/master@{#16940}
 2017-03-01 13:11:22 +00:00
deadbeef
922246a353 Replace NULL with nullptr or null in webrtc/audio/ and common_audio/. 
BUG=webrtc:7147

Review-Url: https://codereview.webrtc.org/2719733002
Cr-Commit-Position: refs/heads/master@{#16843}
 2017-02-26 12:18:12 +00:00
nisse
657bab2455 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. 
This avoids redoing RTP header parsing already done in Call.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2697833002
Cr-Commit-Position: refs/heads/master@{#16750}
 2017-02-21 14:28:10 +00:00
solenberg
08b19dfc67 Remove VoEVideoSync interface. 
The removed tests are covered by cases in call_perf_tests.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2672583002
Cr-Commit-Position: refs/heads/master@{#16621}
 2017-02-15 08:42:31 +00:00
solenberg
06f240bc4f Clean out platform specific things from voice_engine_defines.h. 
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2689183002
Cr-Commit-Position: refs/heads/master@{#16578}
 2017-02-13 12:42:52 +00:00
stefan
7de8d64f89 Wire up audio packet loss to BWE. 
BUG=webtrc:5079

Review-Url: https://codereview.webrtc.org/2658233002
Cr-Commit-Position: refs/heads/master@{#16474}
 2017-02-07 15:14:08 +00:00
nisse
4709e8971b Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. 
We can then drop the CongestionController and RemoteBitrateEstimator
completely from the receive streams.

BUG=webrtc:6847

Review-Url: https://codereview.webrtc.org/2669463006
Cr-Commit-Position: refs/heads/master@{#16459}
 2017-02-07 09:18:43 +00:00
solenberg
6b34124a6d Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ 
BUG=webrtc:6847

Review-Url: https://codereview.webrtc.org/2663063008
Cr-Commit-Position: refs/heads/master@{#16457}
 2017-02-06 21:39:38 +00:00
solenberg
bd9a77f4e5 Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. 
(TBRing webrtc/test/ OWNER)

BUG=webrtc:4690
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2669153004
Cr-Commit-Position: refs/heads/master@{#16455}
 2017-02-06 20:53:57 +00:00
nisse
d44ce0563f Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) 
Reason for revert:
Intending to fix issues and reland.

Original issue's description:
> Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ )
>
> Reason for revert:
> This change causes excessive logging when running tests, and possibly also broke perf tests, see https://build.chromium.org/p/client.webrtc.perf/builders/Linux%20Trusty/builds/1040/steps/webrtc_perf_tests/logs/stdio
>
>
> Original issue's description:
> > Always call RemoteBitrateEstimator::IncomingPacket from Call.
> >
> > Delete the calls from RtpStreamReceiver (for video) and
> > AudioReceiveStream.
> >
> > BUG=webrtc:6847
> >
> > Review-Url: https://codereview.webrtc.org/2659563002
> > Cr-Commit-Position: refs/heads/master@{#16393}
> > Committed: 6d4dd593a8
>
> TBR=stefan@webrtc.org,brandtr@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6847
>
> Review-Url: https://codereview.webrtc.org/2668973003
> Cr-Commit-Position: refs/heads/master@{#16400}
> Committed: 14245cc939

TBR=stefan@webrtc.org,brandtr@webrtc.org
BUG=webrtc:6847

Review-Url: https://codereview.webrtc.org/2673523003
Cr-Commit-Position: refs/heads/master@{#16440}
 2017-02-06 10:23:00 +00:00
nisse
14245cc939 Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) 
Reason for revert:
This change causes excessive logging when running tests, and possibly also broke perf tests, see https://build.chromium.org/p/client.webrtc.perf/builders/Linux%20Trusty/builds/1040/steps/webrtc_perf_tests/logs/stdio

Original issue's description:
> Always call RemoteBitrateEstimator::IncomingPacket from Call.
>
> Delete the calls from RtpStreamReceiver (for video) and
> AudioReceiveStream.
>
> BUG=webrtc:6847
>
> Review-Url: https://codereview.webrtc.org/2659563002
> Cr-Commit-Position: refs/heads/master@{#16393}
> Committed: 6d4dd593a8

TBR=stefan@webrtc.org,brandtr@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6847

Review-Url: https://codereview.webrtc.org/2668973003
Cr-Commit-Position: refs/heads/master@{#16400}
 2017-02-01 16:10:36 +00:00
nisse
6d4dd593a8 Always call RemoteBitrateEstimator::IncomingPacket from Call. 
Delete the calls from RtpStreamReceiver (for video) and
AudioReceiveStream.

BUG=webrtc:6847

Review-Url: https://codereview.webrtc.org/2659563002
Cr-Commit-Position: refs/heads/master@{#16393}
 2017-02-01 11:06:58 +00:00
solenberg
3ebbcb528b Stop using VoEVideoSync in Call/VideoReceiveStream. 
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2452163004
Cr-Commit-Position: refs/heads/master@{#16375}
 2017-01-31 11:58:40 +00:00
stefan
55d1ebb587 Enable periodic bitrate probing when application limited for audio BWE. 
BUG=webrtc:7043

Review-Url: https://codereview.webrtc.org/2657583005
Cr-Commit-Position: refs/heads/master@{#16325}
 2017-01-27 14:17:09 +00:00
mbonadei
9aa3f0a200 Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) 
Reason for revert:
Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file)

Original issue's description:
> Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ )
>
> Reason for revert:
> This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio
>
> Original issue's description:
> > Moving webrtc.gni up one level from build/
> >
> > BUG=webrtc:7030
> >
> > Review-Url: https://codereview.webrtc.org/2651543003
> > Cr-Commit-Position: refs/heads/master@{#16241}
> > Committed: 35a32700fc
>
> TBR=kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7030
>
> Review-Url: https://codereview.webrtc.org/2657563002
> Cr-Commit-Position: refs/heads/master@{#16244}
> Committed: 69dc7dbe24

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7030

Review-Url: https://codereview.webrtc.org/2654773002
Cr-Commit-Position: refs/heads/master@{#16247}
 2017-01-24 14:58:22 +00:00
mbonadei
69dc7dbe24 Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) 
Reason for revert:
This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio

Original issue's description:
> Moving webrtc.gni up one level from build/
>
> BUG=webrtc:7030
>
> Review-Url: https://codereview.webrtc.org/2651543003
> Cr-Commit-Position: refs/heads/master@{#16241}
> Committed: 35a32700fc

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7030

Review-Url: https://codereview.webrtc.org/2657563002
Cr-Commit-Position: refs/heads/master@{#16244}
 2017-01-24 13:14:35 +00:00