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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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482 return output_will_be_muted_; | 482 return output_will_be_muted_; |
483 } | 483 } |
484 | 484 |
485 int AudioProcessingImpl::ProcessStream(const float* const* src, | 485 int AudioProcessingImpl::ProcessStream(const float* const* src, |
486 int samples_per_channel, | 486 int samples_per_channel, |
487 int input_sample_rate_hz, | 487 int input_sample_rate_hz, |
488 ChannelLayout input_layout, | 488 ChannelLayout input_layout, |
489 int output_sample_rate_hz, | 489 int output_sample_rate_hz, |
490 ChannelLayout output_layout, | 490 ChannelLayout output_layout, |
491 float* const* dest) { | 491 float* const* dest) { |
492 CriticalSectionScoped crit_scoped(crit_); | |
Andrew MacDonald
2015/08/11 18:56:36
FYI: Looks like the webrtc mutexes are recursive,
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492 StreamConfig input_stream = api_format_.input_stream(); | 493 StreamConfig input_stream = api_format_.input_stream(); |
493 input_stream.set_sample_rate_hz(input_sample_rate_hz); | 494 input_stream.set_sample_rate_hz(input_sample_rate_hz); |
494 input_stream.set_num_channels(ChannelsFromLayout(input_layout)); | 495 input_stream.set_num_channels(ChannelsFromLayout(input_layout)); |
495 input_stream.set_has_keyboard(LayoutHasKeyboard(input_layout)); | 496 input_stream.set_has_keyboard(LayoutHasKeyboard(input_layout)); |
496 | 497 |
497 StreamConfig output_stream = api_format_.output_stream(); | 498 StreamConfig output_stream = api_format_.output_stream(); |
498 output_stream.set_sample_rate_hz(output_sample_rate_hz); | 499 output_stream.set_sample_rate_hz(output_sample_rate_hz); |
499 output_stream.set_num_channels(ChannelsFromLayout(output_layout)); | 500 output_stream.set_num_channels(ChannelsFromLayout(output_layout)); |
500 output_stream.set_has_keyboard(LayoutHasKeyboard(output_layout)); | 501 output_stream.set_has_keyboard(LayoutHasKeyboard(output_layout)); |
501 | 502 |
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1140 int err = WriteMessageToDebugFile(); | 1141 int err = WriteMessageToDebugFile(); |
1141 if (err != kNoError) { | 1142 if (err != kNoError) { |
1142 return err; | 1143 return err; |
1143 } | 1144 } |
1144 | 1145 |
1145 return kNoError; | 1146 return kNoError; |
1146 } | 1147 } |
1147 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP | 1148 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP |
1148 | 1149 |
1149 } // namespace webrtc | 1150 } // namespace webrtc |
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