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Issue 37793005: move the APM to chrome. (Closed)

Created:
7 years, 2 months ago by no longer working on chromium
Modified:
6 years, 9 months ago
CC:
chromium-reviews, fischman+watch_chromium.org, jam, joi+watch-content_chromium.org, feature-media-reviews_chromium.org, darin-cc_chromium.org, wjia+watch_chromium.org
Visibility:
Public.

Description

move the APM to chrome. BUG=

Patch Set 1 #

Total comments: 2

Patch Set 2 : Moved APM to the track, made interface to get render data, some optimizations. #

Patch Set 3 : fixed the webaudio. #

Patch Set 4 : added a switch, it uses the APM in WebRtc if the switch is off, otherwise use the APM in Chrome. #

Total comments: 6
Unified diffs Side-by-side diffs Delta from patch set Stats (+842 lines, -359 lines) Patch
M content/browser/renderer_host/render_process_host_impl.cc View 1 2 3 1 chunk +1 line, -0 lines 0 comments Download
M content/content_renderer.gypi View 1 1 chunk +2 lines, -0 lines 0 comments Download
M content/public/common/content_switches.h View 1 2 3 1 chunk +1 line, -0 lines 0 comments Download
M content/public/common/content_switches.cc View 1 2 3 1 chunk +3 lines, -0 lines 1 comment Download
M content/renderer/media/media_stream_dependency_factory.cc View 1 2 3 3 chunks +7 lines, -6 lines 0 comments Download
M content/renderer/media/webaudio_capturer_source.h View 1 3 chunks +5 lines, -6 lines 0 comments Download
M content/renderer/media/webaudio_capturer_source.cc View 1 2 5 chunks +11 lines, -10 lines 0 comments Download
M content/renderer/media/webrtc_audio_capturer.h View 1 6 chunks +15 lines, -18 lines 0 comments Download
M content/renderer/media/webrtc_audio_capturer.cc View 1 13 chunks +43 lines, -29 lines 0 comments Download
M content/renderer/media/webrtc_audio_capturer_sink_owner.h View 1 2 3 1 chunk +0 lines, -2 lines 0 comments Download
M content/renderer/media/webrtc_audio_device_impl.h View 1 2 3 5 chunks +23 lines, -6 lines 3 comments Download
M content/renderer/media/webrtc_audio_device_impl.cc View 1 2 3 5 chunks +49 lines, -30 lines 0 comments Download
A content/renderer/media/webrtc_audio_processor.h View 1 2 3 1 chunk +88 lines, -0 lines 1 comment Download
A content/renderer/media/webrtc_audio_processor.cc View 1 2 3 1 chunk +424 lines, -0 lines 1 comment Download
M content/renderer/media/webrtc_local_audio_source_provider.h View 1 2 3 3 chunks +16 lines, -19 lines 0 comments Download
M content/renderer/media/webrtc_local_audio_source_provider.cc View 1 2 3 6 chunks +39 lines, -48 lines 0 comments Download
M content/renderer/media/webrtc_local_audio_source_provider_unittest.cc View 1 1 chunk +0 lines, -35 lines 0 comments Download
M content/renderer/media/webrtc_local_audio_track.h View 1 7 chunks +35 lines, -9 lines 0 comments Download
M content/renderer/media/webrtc_local_audio_track.cc View 1 2 3 10 chunks +60 lines, -133 lines 0 comments Download
M content/renderer/media/webrtc_local_audio_track_unittest.cc View 1 12 chunks +20 lines, -8 lines 0 comments Download

Messages

Total messages: 2 (0 generated)
perkj_chrome
Nice hacking. A few initial comments. Also what we talked about- the feedback method should ...
7 years, 2 months ago (2013-10-24 12:31:30 UTC) #1
Henrik Grunell
7 years, 1 month ago (2013-10-31 11:56:11 UTC) #2
I'm assuming that
- the description of the issue will be updated once the design is more stable,
- a unit test will be added for the audio processor.

Some random comments below.

https://codereview.chromium.org/37793005/diff/100001/content/public/common/co...
File content/public/common/content_switches.cc (right):

https://codereview.chromium.org/37793005/diff/100001/content/public/common/co...
content/public/common/content_switches.cc:917: const char
kEnableWebRtcAudioProcessor[]    = "enable-audio-processor";
I think the name should reflect that the switch changes where processing takes
place. I can't think of a very good name, maybe something like
"switch-audio-processing-from-webrtc-to-chrome"? Or
"use-audio-processing-per-track"?

https://codereview.chromium.org/37793005/diff/100001/content/renderer/media/w...
File content/renderer/media/webrtc_audio_device_impl.h (right):

https://codereview.chromium.org/37793005/diff/100001/content/renderer/media/w...
content/renderer/media/webrtc_audio_device_impl.h:213: virtual int
CaptureData(const std::vector<int>& channels,
Why remove the description of some input parameters and the return value?

https://codereview.chromium.org/37793005/diff/100001/content/renderer/media/w...
content/renderer/media/webrtc_audio_device_impl.h:230: class
WebRtcAudioRenderDataObserver {
Add comment. See also next comment.

https://codereview.chromium.org/37793005/diff/100001/content/renderer/media/w...
content/renderer/media/webrtc_audio_device_impl.h:318: // Add/Remove the
observer to get the mixed render data for AEC.
I think the description "to get the mixed render data for AEC" should be moved
to a comment for WebRtcAudioRenderDataObserver above. Also change the text so
that it's clear that using the data for AEC is one example. There may be other
uses in the future.

https://codereview.chromium.org/37793005/diff/100001/content/renderer/media/w...
File content/renderer/media/webrtc_audio_processor.cc (right):

https://codereview.chromium.org/37793005/diff/100001/content/renderer/media/w...
content/renderer/media/webrtc_audio_processor.cc:99: static const char
kAecDumpFilename[] = "/tmp/audio.aecdump";
This should be different for different platforms. See webrtcvoiceengine.cc in
libjingle ~line 138. Also move the comment and todo (for crbug.com/264223) here.

https://codereview.chromium.org/37793005/diff/100001/content/renderer/media/w...
File content/renderer/media/webrtc_audio_processor.h (right):

https://codereview.chromium.org/37793005/diff/100001/content/renderer/media/w...
content/renderer/media/webrtc_audio_processor.h:37: void Push(media::AudioBus*
audio_source);
Add comment.

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