Because it's easy and generally useful, and because a later CL in this
series needs it.
BUG=webrtc:7459
Review-Url: https://codereview.webrtc.org/2808603002
Cr-Commit-Position: refs/heads/master@{#17633}
This CL removed a couple parameters from the method, and changed the
type of the first parameter to an android.content.Context:
https://codereview.webrtc.org/2800353002/
But applications still using the old method may have already upcast the
context parameter to an Object, in which case this is a breaking change.
So, leaving the old signature exactly as it was before, for maximum
backwards compatibility.
BUG=webrtc:3416
TBR=magjed@webrtc.org
Review-Url: https://codereview.webrtc.org/2810973002
Cr-Commit-Position: refs/heads/master@{#17630}
The "initialize audio/video" parameters are no longer needed, but
at the same time were required to be true, causing a lot of confusion.
This CL removes them, but leaves the old method signature around,
marked "deprecated".
BUG=webrtc:3416
TBR=solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2800353002
Cr-Commit-Position: refs/heads/master@{#17626}
Changes in the microphone gain are effecting the AEC in the sense
that each change in the microphone gain is a change in the echo
path seen by the AEC. This CL utilizes the ability of AEC3 to
leverage information about known changes in the analog microphone
gain.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2808073002
Cr-Commit-Position: refs/heads/master@{#17625}
During the first few capture frames, there is no way for the AEC
to tell whether there is echo in the capture signal as the echo
removal functionality in the AEC has not yet seen any render
signal. To avoid initial echo bursts due to this, this CL adds
functionality for forcing the echo suppression gain to zero during
the first 50 blocks (200 ms) after call start and after a reported
echo path change.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2808733002
Cr-Commit-Position: refs/heads/master@{#17624}
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: fbcc5cb386TBR=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}
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}
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: 292084c376TBR=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}
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}
fixing white spaces
updated authors file
Changed OLA window to use Q14 as Q5 dosnt work with 48khz. 1 ms @ 48 khz is > 2^5
BUG=webrtc:1361
Review-Url: https://codereview.webrtc.org/2763273003
Cr-Commit-Position: refs/heads/master@{#17611}
The unit test ConversationalSpeechTest.MultiEndCallWavReaderAdaptorSine uses CreateSineWavFile() and writes temporary wav files that are used for test (deleted only if the test passes).
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2774423005
Cr-Commit-Position: refs/heads/master@{#17608}
Reason for revert:
Deleting scopedptrcollection.h broke an internal project.
Original issue's description:
> Delete class ScopedPtrCollection. Replaced with vector of unique_ptr.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2808463002
> Cr-Commit-Position: refs/heads/master@{#17605}
> Committed: 188596f20fTBR=pthatcher@webrtc.org,kwiberg@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/2812553002
Cr-Commit-Position: refs/heads/master@{#17607}
This patch fixes compilation issues related to usage of std::unique_ptr
and NULL instead of nullptr. This issue pops up once you would try to
compile whole webrtc with using C++14 and gcc-4.9
BUG=webrtc:7461
Review-Url: https://codereview.webrtc.org/2806693004
Cr-Commit-Position: refs/heads/master@{#17600}
This CL includes extensive tests to match accept or reject decisions on several different timing setups. The setups are simulated using mocks (by far more light-weight than using actual timing and audio track files).
The client code, the unit tests in this case, passes information about the fake audio tracks to MockWavReaderFactory. MockWavReader instances are then created using the parameters defined in the client code. To improve the readability of the tests, generator_unittest.cc includes a docstring explaining how each MultiEndCallSetup* test is documented.
Run tests as follows:
$ out/Default/modules_unittests --gtest_filter=ConversationalSpeechTest.*
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2781573002
Cr-Commit-Position: refs/heads/master@{#17592}
This CL adds support for handling highly reverberant echoes in
AEC3. The functionality is hardcoded to be have no effect (via
a decay factor of 0), but this CL will be followed by other CLs
for which nonzero decay factors will be allowed.
Apart from this change, this CL also refactors the residual
echo estimation code to make it shorter and more readable.
The changes introduced herein are bitexact (for a decay factor
of 0).
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2804223002
Cr-Commit-Position: refs/heads/master@{#17589}
Reason for revert:
This should be landed after https://codereview.webrtc.org/2791963003.
Original issue's description:
> Revert of Adding PRESUBMIT check on google::protobuf (patchset #2 id:20001 of https://codereview.webrtc.org/2753823003/ )
>
> Reason for revert:
> We have to revert https://codereview.webrtc.org/2747863003 and this CL depends on it.
>
> Original issue's description:
> > Adding PRESUBMIT check on google::protobuf
> >
> > The goal is to avoid direct usage of google::protobuf.
> >
> > It should only be used with a 'using' directive in the header file:
> > //webrtc/base/protobuf_utils.h.
> >
> > BUG=webrtc:7340
> > NOTRY=True
> >
> > Review-Url: https://codereview.webrtc.org/2753823003
> > Cr-Commit-Position: refs/heads/master@{#17467}
> > Committed: dd27055cb7
>
> TBR=kjellander@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7340
>
> Review-Url: https://codereview.webrtc.org/2791583002
> Cr-Commit-Position: refs/heads/master@{#17481}
> Committed: 515dff40b7TBR=kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7340
Review-Url: https://codereview.webrtc.org/2792103002
Cr-Commit-Position: refs/heads/master@{#17588}
This CL ensures that the number of bands
for the render side matches that for the capture side
when AEC3 is active. Without this, there was problems
when the render rate is different from the capture rate.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2800033003
Cr-Commit-Position: refs/heads/master@{#17586}
