We have collected the most relevant information on Webrtc Audio Renderer Cc. Open the URLs, which are collected below, and you will find all the info you are interested in.
content/renderer/media/webrtc/processed_local_audio_source ...
https://codereview.chromium.org/1834323002/patch/200001/210059
+++ b/content/renderer/media/webrtc/processed_local_audio_source.cc @@ -2,141 +2,169 @@ // Use of this source code is governed by a BSD-style license that can be // found in the LICENSE file. -#include "content/renderer/media/webrtc_audio_capturer.h" +#include "content/renderer/media/webrtc/processed_local_audio_source.h" -#include "base/bind.h"
webrtc/audio_device_core_win.cc at master · ReadyTalk ...
https://github.com/ReadyTalk/webrtc/blob/master/webrtc/modules/audio_device/win/audio_device_core_win.cc
Cannot retrieve contributors at this time. 5121 lines (4221 sloc) 160 KB Raw Blame
webrtc/block_processor.cc at master · webrtc …
https://github.com/webrtc-uwp/webrtc/blob/master/modules/audio_processing/aec3/block_processor.cc
All contributing project authors may. * be found in the AUTHORS file in the root of the source tree. // If no render data has yet arrived, do not process the capture signal. // capture block. // Reset the delay controller at render buffer underrun. // alignment. // Remove the echo from the capture signal. // Update the metrics.
content/renderer/media/webrtc/processed_local_audio_source ...
https://codereview.chromium.org/1834323002/patch/200001/210060
Patch Set: REBASE + Workaround to ensure MediaStreamAudioProcessor is destroyed on the main thread. Created 4 years, 7 months ago
webrtc/modules/audio_processing/aec3/echo_remover.cc ...
https://codereview.webrtc.org/2678423005/diff/300001/webrtc/modules/audio_processing/aec3/echo_remover.cc
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libjingle - WebRTC library remote audio rendering via ...
https://stackoverflow.com/questions/24160034/webrtc-library-remote-audio-rendering-via-addsink
It's defined in Chromium's audio_renderer.h and uses all sorts of Chromium internal types. If you figure out what to do with it, please let me know, because I am trying to solve the same problem myself. I did notice some code in WebRTC's mediastreamhandler.cc that uses OnData() in the same way you and I are trying to do.
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