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Side by Side Diff: content/renderer/media/webrtc_audio_device_impl.cc

Issue 328493003: Pass the elapsed time from VoE to WebRtcAudioRenderer as the current time for the audio/video eleme… (Closed) Base URL: svn://chrome-svn/chrome/trunk/src/
Patch Set: Created 6 years, 6 months ago
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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"
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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.
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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
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