| Commit message (Expand) | Author | Age | Files | Lines |
|---|
| * | Associate audio streams with their source/destination RenderView. | miu@chromium.org | 2012-12-04 | 1 | -1/+0 |
| * | Break down the webrtc code and AudioInputDevice into a WebRtcAudioCapturer. | xians@chromium.org | 2012-11-16 | 1 | -345/+83 |
| * | Revert 167387 - Break down the webrtc code and AudioInputDevice into a WebRtc... | wez@chromium.org | 2012-11-13 | 1 | -83/+365 |
| * | Break down the webrtc code and AudioInputDevice into a WebRtcAudioCapturer. | xians@chromium.org | 2012-11-13 | 1 | -365/+83 |
| * | Set supported sampling rate for platforms other than linux, bsd, win and mac. | wjia@chromium.org | 2012-11-07 | 1 | -6/+11 |
| * | PeerConnection will setup the call before connecting the audio tag, in this c... | xians@chromium.org | 2012-11-05 | 1 | -10/+4 |
| * | Otherwise it will prevent the second renderer connecting to the ADM | xians@chromium.org | 2012-11-01 | 1 | -0/+10 |
| * | getUserMedia video-only requests should not fail if no audio input device is ... | xians@chromium.org | 2012-10-30 | 1 | -18/+44 |
| * | This allow connection remote stream to audio tag, and enable users to control... | xians@chromium.org | 2012-10-30 | 1 | -103/+49 |
| * | Removing the audio_hardware namespace and move the code into the content name... | tommi@chromium.org | 2012-10-23 | 1 | -3/+3 |
| * | Move the remaining code in content\renderer that wasn't in the content namesp... | jam@chromium.org | 2012-10-22 | 1 | -3/+3 |
| * | Move a bunch of code in content\renderer to the content namespace. | jam@chromium.org | 2012-10-19 | 1 | -1/+4 |
| * | Move ChannelLayout into media namespace. | dalecurtis@google.com | 2012-10-17 | 1 | -7/+8 |
| * | Reland 10907055 which does: | xians@chromium.org | 2012-09-06 | 1 | -4/+2 |
| * | Revert 154804 - We need to set the session id to the WebRtcAudioDevice to be ... | wjia@chromium.org | 2012-09-05 | 1 | -2/+4 |
| * | We need to set the session id to the WebRtcAudioDevice to be able to use the ... | xians@chromium.org | 2012-09-04 | 1 | -4/+2 |
| * | Upgrade AudioBus to support wrapping, interleaving. | dalecurtis@chromium.org | 2012-08-21 | 1 | -14/+6 |
| * | Switch AudioRenderSink::Callback to use AudioBus. | dalecurtis@chromium.org | 2012-08-09 | 1 | -15/+13 |
| * | Move AudioDevice and AudioInputDevice to media. | tommi@chromium.org | 2012-07-27 | 1 | -1/+2 |
| * | First step towards moving AudioDevice and AudioInputDevice from content/ to m... | tommi@chromium.org | 2012-07-26 | 1 | -3/+5 |
| * | This CL adds a new factory method called AudioDeviceFactory. It is a template... | henrika@chromium.org | 2012-06-29 | 1 | -11/+13 |
| * | Ensures that the user can select 8kHz input sample rate as default rate and s... | henrika@chromium.org | 2012-05-07 | 1 | -1/+5 |
| * | Adds WebRTC histogram data to be uploaded as part of UMA logging events. | henrika@chromium.org | 2012-04-11 | 1 | -6/+126 |
| * | Merge AudioRendererImpl and AudioRendererBase; add NullAudioSink | vrk@chromium.org | 2012-04-06 | 1 | -8/+9 |
| * | Move media/audio files into media namespace (relanding) | vrk@google.com | 2012-04-03 | 1 | -0/+2 |
| * | Change InterleaveFloatToInt16() to work with uint8 and int32 | vrk@google.com | 2012-04-02 | 1 | -4/+5 |
| * | Revert 130180 - Move media/audio files into media namespace | vrk@google.com | 2012-04-02 | 1 | -2/+0 |
| * | Move media/audio files into media namespace | vrk@google.com | 2012-04-02 | 1 | -0/+2 |
| * | Adds Analog Gain Control (AGC) to the WebRTC client. | henrika@chromium.org | 2012-03-28 | 1 | -35/+82 |
| * | Make AudioParameters a class instead of a struct | vrk@google.com | 2012-03-21 | 1 | -66/+64 |
| * | Added support for 96kHz capture rate on Windows 7 and Mac OS X for WebRTC. | henrika@chromium.org | 2012-02-24 | 1 | -25/+39 |
| * | Adds support for 96kHz output sample rate in WebRTC. | henrika@chromium.org | 2012-02-22 | 1 | -4/+8 |
| * | Extends error handling for audio capture and rendering. | henrika@chromium.org | 2012-02-13 | 1 | -4/+14 |
| * | Roll webrtc 1538 and libjingle 112 | grunell@chromium.org | 2012-01-27 | 1 | -20/+0 |
| * | Detect errors in audio output and report them upstream. | fischman@chromium.org | 2012-01-27 | 1 | -1/+5 |
| * | Removes invalid DCHECK() calls in the WebRTC ADM. | henrika@chromium.org | 2012-01-24 | 1 | -5/+3 |
| * | Adds support for 16kHz input sample rate and mono channel config. in WebRTC. | henrika@chromium.org | 2012-01-19 | 1 | -23/+20 |
| * | Fix start/stop of html5 audio stream and race condition when pausing. | enal@chromium.org | 2011-12-21 | 1 | -1/+2 |
| * | Coverity: Initialize member variables. | jhawkins@chromium.org | 2011-12-20 | 1 | -0/+1 |
| * | remove the race related to output_delay_ms_ in ADM and races in the unittests. | xians@chromium.org | 2011-12-12 | 1 | -9/+24 |
| * | There is a racing between SyncSocket::Receive in audio_thread_ and SyncSocket... | xians@chromium.org | 2011-12-07 | 1 | -4/+6 |
| * | Resolves crash related to conflict with latest WebRTC (1008) and the WebRTCAu... | henrika@chromium.org | 2011-11-28 | 1 | -5/+4 |
| * | Enable the device selection for linux and mac by passing a device unique id f... | xians@chromium.org | 2011-11-22 | 1 | -5/+5 |
| * | Refactor the Get*Hardware* routines a bit. | tommi@chromium.org | 2011-11-18 | 1 | -12/+4 |
| * | Low-latency AudioOutputStream implementation based on WASAPI for Windows. | henrika@chromium.org | 2011-11-16 | 1 | -96/+125 |
| * | share all the needed linux code with OpenBSD in chrome and content | robert.nagy@gmail.com | 2011-11-11 | 1 | -1/+1 |
| * | Low-latency AudioInputStream implementation based on WASAPI for Windows. | henrika@chromium.org | 2011-10-24 | 1 | -24/+45 |
| * | ALSA should support 10ms playout buffer size now. | xians@chromium.org | 2011-10-07 | 1 | -3/+3 |
| * | remove NewRunnableMethod and switch to base::Bind | wjia@chromium.org | 2011-10-07 | 1 | -3/+3 |
| * | Rename RenderThread to RenderThreadImpl | jam@chromium.org | 2011-10-06 | 1 | -2/+2 |
