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path: root/content/renderer/media/webrtc_audio_renderer.cc
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* Add support for 384kHz sample rates; works in other browsers.dalecurtis2016-01-071-2/+2
* Convert Pass()→std::move() in //content/rendererdcheng2015-12-261-1/+3
* Switch to standard integer types in content/renderer/.avi2015-12-251-0/+1
* Forward the number of skipped frames by the OS in audio playout.grunell2015-12-181-2/+4
* Revert of Forward the number of skipped frames by the OS in audio playout. (p...grunell2015-12-171-4/+2
* Forward the number of skipped frames by the OS in audio playout.grunell2015-12-141-2/+4
* Add support for unmixed audio from remote WebRTC remote tracks.tommi2015-12-121-1/+2
* Refactored Chrome MediaStream to not contain an is_local flag or a webrtc spe...perkj2015-12-021-19/+29
* Fix for FIFO parameters calculation in WebRtcAudioRenderer::PrepareSink()olka2015-12-011-20/+23
* Switch output device in WebRTC renderers by creating new sinks.guidou2015-10-131-75/+155
* Allow initializing WebRTC audio renderers with explicit output device ID.guidou2015-10-061-3/+8
* Read output parameters from the output device in WebRTC renderers.guidou2015-10-051-14/+13
* Allow AudioOutputDevice objects to be initialized with a specific hardware ou...guidou2015-09-191-3/+3
* Refactor AudioParameters member setting.ajm2015-09-081-8/+6
* Include default communication devices in audio device enumerations. This remo...tommi2015-08-281-17/+1
* Introduce OutputDevice interface.guidou2015-07-311-14/+6
* Add support for SwitchOutputDevice to the mediastream WebMediaPlayer and rend...guidou2015-06-181-0/+21
* Remove render view id from the audio input and output, part two!dalecurtis2015-04-031-9/+3
* Add support for a signaling thread message loop to WebRtcAudioRenderer.tommi2014-10-291-1/+11
* Use the optimal buffer size for the local audio renderer.xians2014-10-221-4/+4
* Standardize usage of virtual/override/final in content/renderer/dcheng2014-10-211-8/+8
* Replace FINAL and OVERRIDE with their C++11 counterparts in content/renderermohan.reddy2014-10-081-7/+7
* Adds time measurement of AudioOutputDevice::AudioThreadCallback::Processhenrika2014-09-191-1/+15
* Add support for 24kHz audio sample rate and remove the validation checktommi2014-09-081-28/+0
* Used 10ms native buffer size for webrtc audio renderer on Linux and Mac. And ...xians2014-09-031-72/+35
* Revert of Revert of Remove the last piece of deprecated synchronous IO code. ...xians2014-08-281-3/+3
* Revert of Remove the last piece of deprecated synchronous IO code. (patchset ...tnagel2014-08-271-3/+3
* Remove the last piece of deprecated synchronous IO code.xians2014-08-271-3/+3
* [Cross-Site Isolation] Migrate entire MediaStream verticals to be per-RenderF...miu@chromium.org2014-07-171-6/+8
* Turn audio ducking on by default on Windows again.tommi@chromium.org2014-07-081-11/+20
* Turn off audio ducking for webrtc output.tommi@chromium.org2014-07-061-1/+9
* Pass the elapsed time from VoE to WebRtcAudioRenderer as the current time for...ronghuawu@chromium.org2014-06-121-2/+5
* Remove unused RenderIO() interface.xians@chromium.org2014-05-201-3/+1
* Revert 255158 "Avoid hitting the thread check when WebRtcAudioRe..."vrk@chromium.org2014-03-121-7/+5
* Avoid hitting the thread check when WebRtcAudioRenderer is going away.xians@chromium.org2014-03-051-5/+7
* Fix audio ducking support for the output side on Windows.tommi@chromium.org2014-03-051-2/+2
* Clean up histogram'd media enum max values.rileya@chromium.org2014-02-281-3/+3
* Remove WebRTC.AudioOutputChannelLayout UMA histogram and mark it obsolete.joi@chromium.org2014-02-261-3/+0
* Feed the render data to MediaStreamAudioProcessor and used AudioBus in render...xians@chromium.org2014-02-211-52/+27
* Allow individual volume control of remote webrtc audio tracks.tommi@chromium.org2014-02-171-22/+153
* A lot of plumbing to get render_frame_id to PrerenderManager::OnCreatingAudio...jam@chromium.org2014-01-211-1/+4
* Add diagnostic WebRTC logging.grunell@chromium.org2013-11-281-0/+9
* Improves output audio on most Android devices.henrika@chromium.org2013-09-241-9/+16
* Ensure that the shared WebRtcAudioRenderer instance gets deleted when not used.tommi@chromium.org2013-09-111-12/+115
* Implicit audio output device selection for getUserMedia.tommi@chromium.org2013-09-101-11/+19
* Fixed the UMA for webrtc sample rates. We need to map the sample rate to medi...xians@chromium.org2013-06-201-2/+7
* Enable low latency mode for audio playback on Androidwjia@chromium.org2013-06-171-0/+6
* Use a direct include of strings headers in content/renderer and content/shell.avi@chromium.org2013-06-111-1/+1
* Update content/ to use scoped_refptr<T>::get() rather than implicit "operator...rsleevi@chromium.org2013-06-021-1/+1
* Adjust delay calculations on Mac to properly account for the FIFO buffers.ajm@chromium.org2013-05-311-16/+10
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