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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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476 agc_manager_->SetCaptureMuted(output_will_be_muted_); | 476 agc_manager_->SetCaptureMuted(output_will_be_muted_); |
477 } | 477 } |
478 } | 478 } |
479 | 479 |
480 bool AudioProcessingImpl::output_will_be_muted() const { | 480 bool AudioProcessingImpl::output_will_be_muted() const { |
481 CriticalSectionScoped lock(crit_); | 481 CriticalSectionScoped lock(crit_); |
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 size_t 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 StreamConfig input_stream = api_format_.input_stream(); | 492 StreamConfig input_stream = api_format_.input_stream(); |
493 input_stream.set_sample_rate_hz(input_sample_rate_hz); | 493 input_stream.set_sample_rate_hz(input_sample_rate_hz); |
494 input_stream.set_num_channels(ChannelsFromLayout(input_layout)); | 494 input_stream.set_num_channels(ChannelsFromLayout(input_layout)); |
495 input_stream.set_has_keyboard(LayoutHasKeyboard(input_layout)); | 495 input_stream.set_has_keyboard(LayoutHasKeyboard(input_layout)); |
496 | 496 |
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675 } | 675 } |
676 | 676 |
677 // The level estimator operates on the recombined data. | 677 // The level estimator operates on the recombined data. |
678 RETURN_ON_ERR(level_estimator_->ProcessStream(ca)); | 678 RETURN_ON_ERR(level_estimator_->ProcessStream(ca)); |
679 | 679 |
680 was_stream_delay_set_ = false; | 680 was_stream_delay_set_ = false; |
681 return kNoError; | 681 return kNoError; |
682 } | 682 } |
683 | 683 |
684 int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data, | 684 int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data, |
685 int samples_per_channel, | 685 size_t samples_per_channel, |
686 int sample_rate_hz, | 686 int sample_rate_hz, |
687 ChannelLayout layout) { | 687 ChannelLayout layout) { |
688 const StreamConfig reverse_config = { | 688 const StreamConfig reverse_config = { |
689 sample_rate_hz, ChannelsFromLayout(layout), LayoutHasKeyboard(layout), | 689 sample_rate_hz, ChannelsFromLayout(layout), LayoutHasKeyboard(layout), |
690 }; | 690 }; |
691 if (samples_per_channel != reverse_config.num_frames()) { | 691 if (samples_per_channel != reverse_config.num_frames()) { |
692 return kBadDataLengthError; | 692 return kBadDataLengthError; |
693 } | 693 } |
694 return AnalyzeReverseStream(data, reverse_config); | 694 return AnalyzeReverseStream(data, reverse_config); |
695 } | 695 } |
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1140 int err = WriteMessageToDebugFile(); | 1140 int err = WriteMessageToDebugFile(); |
1141 if (err != kNoError) { | 1141 if (err != kNoError) { |
1142 return err; | 1142 return err; |
1143 } | 1143 } |
1144 | 1144 |
1145 return kNoError; | 1145 return kNoError; |
1146 } | 1146 } |
1147 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP | 1147 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP |
1148 | 1148 |
1149 } // namespace webrtc | 1149 } // namespace webrtc |
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