OLD | NEW |
---|---|
1 // Copyright 2013 The Chromium Authors. All rights reserved. | 1 // Copyright 2013 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include "content/renderer/media/webrtc_audio_device_impl.h" | 5 #include "content/renderer/media/webrtc_audio_device_impl.h" |
6 | 6 |
7 #include "base/bind.h" | 7 #include "base/bind.h" |
8 #include "base/metrics/histogram.h" | 8 #include "base/metrics/histogram.h" |
9 #include "base/strings/string_util.h" | 9 #include "base/strings/string_util.h" |
10 #include "base/win/windows_version.h" | 10 #include "base/win/windows_version.h" |
(...skipping 161 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
172 audio_transport_callback_->NeedMorePlayData(frames_per_10_ms, | 172 audio_transport_callback_->NeedMorePlayData(frames_per_10_ms, |
173 bytes_per_sample, | 173 bytes_per_sample, |
174 audio_bus->channels(), | 174 audio_bus->channels(), |
175 sample_rate, | 175 sample_rate, |
176 audio_data, | 176 audio_data, |
177 num_audio_frames, | 177 num_audio_frames, |
178 &elapsed_time_ms, | 178 &elapsed_time_ms, |
179 &ntp_time_ms); | 179 &ntp_time_ms); |
180 accumulated_audio_frames += num_audio_frames; | 180 accumulated_audio_frames += num_audio_frames; |
181 } | 181 } |
182 | 182 if (elapsed_time_ms >= 0) { |
183 renderer_->SetCurrentRenderTime( | |
no longer working on chromium
2014/06/10 16:21:38
this is on audio render thread.
Ronghua Wu (Left Chromium)
2014/06/10 17:01:07
Done.
| |
184 base::TimeDelta::FromMilliseconds(elapsed_time_ms)); | |
185 } | |
183 audio_data += bytes_per_10_ms; | 186 audio_data += bytes_per_10_ms; |
184 } | 187 } |
185 | 188 |
186 // De-interleave each channel and convert to 32-bit floating-point | 189 // De-interleave each channel and convert to 32-bit floating-point |
187 // with nominal range -1.0 -> +1.0 to match the callback format. | 190 // with nominal range -1.0 -> +1.0 to match the callback format. |
188 audio_bus->FromInterleaved(&render_buffer_[0], | 191 audio_bus->FromInterleaved(&render_buffer_[0], |
189 audio_bus->frames(), | 192 audio_bus->frames(), |
190 bytes_per_sample); | 193 bytes_per_sample); |
191 | 194 |
192 // Pass the render data to the playout sinks. | 195 // Pass the render data to the playout sinks. |
(...skipping 366 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
559 | 562 |
560 // Start the Aec dump on the current default capturer. | 563 // Start the Aec dump on the current default capturer. |
561 scoped_refptr<WebRtcAudioCapturer> default_capturer(GetDefaultCapturer()); | 564 scoped_refptr<WebRtcAudioCapturer> default_capturer(GetDefaultCapturer()); |
562 if (!default_capturer) | 565 if (!default_capturer) |
563 return; | 566 return; |
564 | 567 |
565 default_capturer->StartAecDump(aec_dump_file_.Pass()); | 568 default_capturer->StartAecDump(aec_dump_file_.Pass()); |
566 } | 569 } |
567 | 570 |
568 } // namespace content | 571 } // namespace content |
OLD | NEW |