This functionality already exists for video streams, so not having it
for audio is unexpected and has lead to problems.
BUG=webrtc:7631
Review-Url: https://codereview.webrtc.org/2887733002
Cr-Commit-Position: refs/heads/master@{#18231}
probing_interval as a name is used for the period that BWE attempt to increase its estimate. The name is confusing since it is not related to "probing" which is a special mechanism for estimating BWE.
BUG=None
Review-Url: https://codereview.webrtc.org/2888893002
Cr-Commit-Position: refs/heads/master@{#18203}
These are the changes made to WebRtcVoiceEngine and surrounding
code. It still contains some things that are inelegant, like how
AudioCodecSpec and AudioFormatInfo is ferried around in
SendCodecSpec. This should probably be resolved before landing.
There are also a few test still that are disabled. They should be
removed or fixed, as the case may be.
I've put this CL up to get a better overview of the changes made and
how reviewable they are.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2705093002
Cr-Commit-Position: refs/heads/master@{#17904}
Also move RtpTransportControllerSendInterface to its own header file.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2808043002
Cr-Commit-Position: refs/heads/master@{#17840}
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}
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}
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}
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}
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}
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}
And delete the method CongestionController::packet_router.
BUG=None
Review-Url: https://codereview.webrtc.org/2516983004
Cr-Commit-Position: refs/heads/master@{#15323}
Reason for revert:
Unfortunately, this change breaks internal projects. Specifically the change to the CongestionController interface means anything implementing it will be forced to change in lock-step.
Original issue's description:
> Pass time constanct to bwe smoothing filter.
>
> BUG=webrtc:6443, webrtc:6303
>
> Committed: https://crrev.com/9abbf5ae4ec7d688a9b4aa03a405f3faadb74b90
> Cr-Commit-Position: refs/heads/master@{#15266}
TBR=minyue@webrtc.org,stefan@webrtc.org,solenberg@webrtc.org,michaelt@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6443, webrtc:6303
Review-Url: https://codereview.webrtc.org/2532993002
Cr-Commit-Position: refs/heads/master@{#15272}
Payload type -> RtpCodecParameters maps added for sender and receiver.
This is a follow-up to https://codereview.webrtc.org/2484193002/ which
did the same thing for VideoMediaInfo. This information will be used to
produce RTCCodecStats[1].
Voice[Sender/Receiver]Info is updated with current codec payload type
for every stream which can be used to look up the codec in
VoiceMediaInfo.
[1] https://w3c.github.io/webrtc-stats/#codec-dict*
BUG=chromium:659117
Review-Url: https://codereview.webrtc.org/2503383002
Cr-Commit-Position: refs/heads/master@{#15144}
We support multiple payload types, and one which matches the audio codec the closest, is picked (or the one with lowest clock rate, if no perfect match is found).
The exact clock rate is then ignored and DTMF packets are time stamped with the rate of the current audio codec. This is exactly the way the code has worked up to this point, but until now we have been under the impression that we were in fact sending 8k DTMF.
In other words, this is an improvement over the current situation, since we will most likely find a payload type which matches the codec clock rate.
This CL also does a little cleaning of the DTMFQueue and RTPSenderAudio classes.
BUG=webrtc:2795
Review-Url: https://codereview.webrtc.org/2392883002
Cr-Commit-Position: refs/heads/master@{#15129}
Updates GetStats() function in AudioSendStream to use the new GetStatistics function in APM instead of the corresponding VoE functions.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2463813002
Cr-Commit-Position: refs/heads/master@{#15065}
The unit was kbps but the one default use of it is in bps. The inconsistency should be fixed.
BUG=webrtc:6670
Review-Url: https://codereview.webrtc.org/2247213005
Cr-Commit-Position: refs/heads/master@{#14955}
As a side effect:
- Moved the AudioSendStream::Config::SendCodecSpec methods into the .cc.
- Which exposed an issue with event_visualizer_utils not having a dependency on api:call_api set up.
- Which further exposed clang warnings about large inlined default methods in webrtc/config.h.
BUG=webrtc:4690
Committed: https://crrev.com/1836fd6257a692959b3b49ba99ef587ad9995871
Review-Url: https://codereview.webrtc.org/2446963003
Cr-Original-Commit-Position: refs/heads/master@{#14771}
Cr-Commit-Position: refs/heads/master@{#14780}
Reason for revert:
Breaks downstream project
Original issue's description:
> Clean up logging in AudioSendStream::SetupSendCodec().
>
> As a side effect:
> - Moved the AudioSendStream::Config::SendCodecSpec methods into the .cc.
> - Which exposed an issue with event_visualizer_utils not having a dependency on api:call_api set up.
> - Which further exposed clang warnings about large inlined default methods in webrtc/config.h.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/1836fd6257a692959b3b49ba99ef587ad9995871
> Cr-Commit-Position: refs/heads/master@{#14771}
TBR=kwiberg@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:4690
Review-Url: https://codereview.webrtc.org/2452643002
Cr-Commit-Position: refs/heads/master@{#14774}
As a side effect:
- Moved the AudioSendStream::Config::SendCodecSpec methods into the .cc.
- Which exposed an issue with event_visualizer_utils not having a dependency on api:call_api set up.
- Which further exposed clang warnings about large inlined default methods in webrtc/config.h.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2446963003
Cr-Commit-Position: refs/heads/master@{#14771}
The stat is currently always set to zero until the residual echo detector has landed.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2431443003
Cr-Commit-Position: refs/heads/master@{#14721}
Reason for revert:
Speculative revert.
Intermittent memory access errors suspected to be caused by this cl.
See for instance https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/8018
UNADDRESSABLE ACCESS of freed memory: reading 0x0331d330-0x0331d334 4 byte(s)
# 0 webrtc::voe::RtcpRttStatsProxy::LastProcessedRtt
# 1 webrtc::ModuleRtpRtcpImpl::Process
Original issue's description:
> Add RtcpRttStats to AudioStream
>
> BUG=webrtc:6508
>
> Committed: https://crrev.com/e0729c56d35acfaf9738fdb32c6508cd78eaf089
> Cr-Commit-Position: refs/heads/master@{#14595}
TBR=stefan@webrtc.org,minyue@webrtc.org,solenberg@webrtc.org,michaelt@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6508
Review-Url: https://codereview.webrtc.org/2415943002
Cr-Commit-Position: refs/heads/master@{#14631}
cl was originally reviewed here:
https://codereview.webrtc.org/2060403002/
- Add task queue to Call with the intent of replacing the use of one of the process threads.
- Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.
- BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.
- VideoEncoderConfig and VideoSendStream::Config support move semantics.
- The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.
TBR=mflodman@webrtc.org
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2250123002
Cr-Commit-Position: refs/heads/master@{#14014}
Reason for revert:
Failed on Win 10 Chrome FYI.
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win10%20Tester/builds/3847/steps/content_browsertests/logs/stdio
#
# Fatal error in e:\b\c\b\win_builder\src\third_party\webrtc\base\task_queue_win.cc, line 138
# last system error: 87
# Check failed: ((DWORD)0xFFFFFFFF) != result (4294967295 vs. 4294967295)
#
WebRtcBrowserTest
#
Original issue's description:
> - Add task queue to Call with the intent of replacing the use of one of the process threads.
>
> - Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.
>
> - BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.
>
> - VideoEncoderConfig and VideoSendStream::Config support move semantics.
>
> - The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.
>
> BUG=webrtc:5687
>
> Committed: https://crrev.com/cc168360f41322332860cb075edeb1cde21aa473
> Cr-Commit-Position: refs/heads/master@{#13767}
TBR=tommi@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org,sprang@webrtc.org,pbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2248713003
Cr-Commit-Position: refs/heads/master@{#13774}
- Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.
- BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.
- VideoEncoderConfig and VideoSendStream::Config support move semantics.
- The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2060403002
Cr-Commit-Position: refs/heads/master@{#13767}
This is a first CL wiring up AudioSendStream to BitrateAllocator. This
is still experimental and there is a test added for the audio only case,
combined audio video variable bitrate test cases will be added as a
follow up.
BUG=5079
Review-Url: https://codereview.webrtc.org/2165743003
Cr-Commit-Position: refs/heads/master@{#13527}
Removes the need to use VoEVolume::SetInputMute()/GetInputMute().
BUG=webrtc:4690
NOTRY=true
Review-Url: https://codereview.webrtc.org/2066973002
Cr-Commit-Position: refs/heads/master@{#13172}
