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

Issue 2004283002: AudioConverter: Express delay in frames rather than msec. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 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_local_audio_source_provider.h" 5 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
6 6
7 #include "base/logging.h" 7 #include "base/logging.h"
8 #include "content/renderer/render_thread_impl.h" 8 #include "content/renderer/render_thread_impl.h"
9 #include "media/base/audio_fifo.h" 9 #include "media/base/audio_fifo.h"
10 #include "media/base/audio_hardware_config.h" 10 #include "media/base/audio_hardware_config.h"
(...skipping 112 matching lines...) Expand 10 before | Expand all | Expand 10 after
123 output_wrapper_->SetChannelData(i, audio_data[i]); 123 output_wrapper_->SetChannelData(i, audio_data[i]);
124 124
125 base::AutoLock auto_lock(lock_); 125 base::AutoLock auto_lock(lock_);
126 if (!audio_converter_) 126 if (!audio_converter_)
127 return; 127 return;
128 128
129 is_enabled_ = true; 129 is_enabled_ = true;
130 audio_converter_->Convert(output_wrapper_.get()); 130 audio_converter_->Convert(output_wrapper_.get());
131 } 131 }
132 132
133 double WebRtcLocalAudioSourceProvider::ProvideInput( 133 double WebRtcLocalAudioSourceProvider::ProvideInput(media::AudioBus* audio_bus,
134 media::AudioBus* audio_bus, base::TimeDelta buffer_delay) { 134 uint32_t frames_delayed) {
135 if (fifo_->frames() >= audio_bus->frames()) { 135 if (fifo_->frames() >= audio_bus->frames()) {
136 fifo_->Consume(audio_bus, 0, audio_bus->frames()); 136 fifo_->Consume(audio_bus, 0, audio_bus->frames());
137 } else { 137 } else {
138 audio_bus->Zero(); 138 audio_bus->Zero();
139 DVLOG(1) << "WARNING: Underrun, FIFO has data " << fifo_->frames() 139 DVLOG(1) << "WARNING: Underrun, FIFO has data " << fifo_->frames()
140 << " samples but " << audio_bus->frames() 140 << " samples but " << audio_bus->frames()
141 << " samples are needed"; 141 << " samples are needed";
142 } 142 }
143 143
144 return 1.0; 144 return 1.0;
145 } 145 }
146 146
147 void WebRtcLocalAudioSourceProvider::SetSinkParamsForTesting( 147 void WebRtcLocalAudioSourceProvider::SetSinkParamsForTesting(
148 const media::AudioParameters& sink_params) { 148 const media::AudioParameters& sink_params) {
149 sink_params_ = sink_params; 149 sink_params_ = sink_params;
150 } 150 }
151 151
152 } // namespace content 152 } // namespace content
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