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

Issue 131213003: Fixed the racing in the resampler of webrtc when two streams are calling OnData() (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: Created 6 years, 11 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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74 } 74 }
75 75
76 // Write audio samples in blocks of 10 milliseconds to the registered 76 // Write audio samples in blocks of 10 milliseconds to the registered
77 // webrtc::AudioTransport sink. Keep writing until our internal byte 77 // webrtc::AudioTransport sink. Keep writing until our internal byte
78 // buffer is empty. 78 // buffer is empty.
79 const int16* audio_buffer = audio_data; 79 const int16* audio_buffer = audio_data;
80 const int samples_per_10_msec = (sample_rate / 100); 80 const int samples_per_10_msec = (sample_rate / 100);
81 CHECK_EQ(number_of_frames % samples_per_10_msec, 0); 81 CHECK_EQ(number_of_frames % samples_per_10_msec, 0);
82 int accumulated_audio_samples = 0; 82 int accumulated_audio_samples = 0;
83 uint32_t new_volume = 0; 83 uint32_t new_volume = 0;
84
85 // The lock here is to protect the racing in the resampler inside webrtc when
86 // there are more than one input streams calling OnData(), which can happen
henrika (OOO until Aug 14) 2014/01/09 10:07:14 nit, ...input stream calling...
no longer working on chromium 2014/01/09 10:18:32 Done.
87 // when the users setup two getUserMedia, one for the microphone, another
88 // for WebAudio. Currently we don't have a better way to fix it except for
89 // adding a lock here to sequence the call.
90 // TODO(xians): Remove this workaround after we move the
henrika (OOO until Aug 14) 2014/01/09 10:07:14 Add link to crbug issue here.
no longer working on chromium 2014/01/09 10:18:32 Done.
91 // webrtc::AudioProcessing module to Chrome.
92 base::AutoLock auto_lock(capture_callback_lock_);
84 while (accumulated_audio_samples < number_of_frames) { 93 while (accumulated_audio_samples < number_of_frames) {
85 // Deliver 10ms of recorded 16-bit linear PCM audio. 94 // Deliver 10ms of recorded 16-bit linear PCM audio.
86 int new_mic_level = audio_transport_callback_->OnDataAvailable( 95 int new_mic_level = audio_transport_callback_->OnDataAvailable(
87 &channels[0], 96 &channels[0],
88 channels.size(), 97 channels.size(),
89 audio_buffer, 98 audio_buffer,
90 sample_rate, 99 sample_rate,
91 number_of_channels, 100 number_of_channels,
92 samples_per_10_msec, 101 samples_per_10_msec,
93 total_delay_ms, 102 total_delay_ms,
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440 for (CapturerList::const_iterator iter = capturers_.begin(); 449 for (CapturerList::const_iterator iter = capturers_.begin();
441 iter != capturers_.end(); ++iter) { 450 iter != capturers_.end(); ++iter) {
442 if (!(*iter)->device_id().empty()) 451 if (!(*iter)->device_id().empty())
443 return *iter; 452 return *iter;
444 } 453 }
445 454
446 return NULL; 455 return NULL;
447 } 456 }
448 457
449 } // namespace content 458 } // namespace content
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