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| 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2012 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_local_audio_renderer.h" | 5 #include "content/renderer/media/webrtc_local_audio_renderer.h" |
| 6 | 6 |
| 7 #include "base/debug/trace_event.h" | 7 #include "base/debug/trace_event.h" |
| 8 #include "base/logging.h" | 8 #include "base/logging.h" |
| 9 #include "base/message_loop/message_loop_proxy.h" | 9 #include "base/message_loop/message_loop_proxy.h" |
| 10 #include "base/synchronization/lock.h" | 10 #include "base/synchronization/lock.h" |
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| 45 | 45 |
| 46 return audio_bus->frames(); | 46 return audio_bus->frames(); |
| 47 } | 47 } |
| 48 | 48 |
| 49 void WebRtcLocalAudioRenderer::OnRenderError() { | 49 void WebRtcLocalAudioRenderer::OnRenderError() { |
| 50 NOTIMPLEMENTED(); | 50 NOTIMPLEMENTED(); |
| 51 } | 51 } |
| 52 | 52 |
| 53 // content::WebRtcAudioCapturerSink implementation | 53 // content::WebRtcAudioCapturerSink implementation |
| 54 int WebRtcLocalAudioRenderer::CaptureData(const std::vector<int>& channels, | 54 int WebRtcLocalAudioRenderer::CaptureData(const std::vector<int>& channels, |
| 55 const int16* audio_data, | 55 const int16* audio_data, |
| 56 int sample_rate, | 56 int sample_rate, |
| 57 int number_of_channels, | 57 int number_of_channels, |
| 58 int number_of_frames, | 58 int number_of_frames, |
| 59 int audio_delay_milliseconds, | 59 int audio_delay_milliseconds, |
| 60 int current_volume, | 60 int current_volume, |
| 61 bool need_audio_processing, | 61 bool need_audio_processing) { |
| 62 bool key_pressed) { | |
| 63 TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::CaptureData"); | 62 TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::CaptureData"); |
| 64 base::AutoLock auto_lock(thread_lock_); | 63 base::AutoLock auto_lock(thread_lock_); |
| 65 | 64 |
| 66 if (!playing_) | 65 if (!playing_) |
| 67 return 0; | 66 return 0; |
| 68 | 67 |
| 69 // Push captured audio to FIFO so it can be read by a local sink. | 68 // Push captured audio to FIFO so it can be read by a local sink. |
| 70 if (loopback_fifo_) { | 69 if (loopback_fifo_) { |
| 71 if (loopback_fifo_->frames() + number_of_frames <= | 70 if (loopback_fifo_->frames() + number_of_frames <= |
| 72 loopback_fifo_->max_frames()) { | 71 loopback_fifo_->max_frames()) { |
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| 240 if (!sink_.get()) | 239 if (!sink_.get()) |
| 241 return base::TimeDelta(); | 240 return base::TimeDelta(); |
| 242 return total_render_time(); | 241 return total_render_time(); |
| 243 } | 242 } |
| 244 | 243 |
| 245 bool WebRtcLocalAudioRenderer::IsLocalRenderer() const { | 244 bool WebRtcLocalAudioRenderer::IsLocalRenderer() const { |
| 246 return true; | 245 return true; |
| 247 } | 246 } |
| 248 | 247 |
| 249 } // namespace content | 248 } // namespace content |
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