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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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| 58 ChannelLayout capture_input_layout, | 58 ChannelLayout capture_input_layout, |
| 59 ChannelLayout capture_output_layout, | 59 ChannelLayout capture_output_layout, |
| 60 ChannelLayout render_input_layout) override; | 60 ChannelLayout render_input_layout) override; |
| 61 int Initialize(const ProcessingConfig& processing_config) override; | 61 int Initialize(const ProcessingConfig& processing_config) override; |
| 62 void ApplyConfig(const AudioProcessing::Config& config) override; | 62 void ApplyConfig(const AudioProcessing::Config& config) override; |
| 63 void SetExtraOptions(const webrtc::Config& config) override; | 63 void SetExtraOptions(const webrtc::Config& config) override; |
| 64 void UpdateHistogramsOnCallEnd() override; | 64 void UpdateHistogramsOnCallEnd() override; |
| 65 int StartDebugRecording(const char filename[kMaxFilenameSize], | 65 int StartDebugRecording(const char filename[kMaxFilenameSize], |
| 66 int64_t max_log_size_bytes) override; | 66 int64_t max_log_size_bytes) override; |
| 67 int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) override; | 67 int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) override; |
| 68 | 68 int StartDebugRecording(FILE* handle) override; |
| 69 int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override; | 69 int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override; |
| 70 int StopDebugRecording() override; | 70 int StopDebugRecording() override; |
| 71 | 71 |
| 72 // Capture-side exclusive methods possibly running APM in a | 72 // Capture-side exclusive methods possibly running APM in a |
| 73 // multi-threaded manner. Acquire the capture lock. | 73 // multi-threaded manner. Acquire the capture lock. |
| 74 int ProcessStream(AudioFrame* frame) override; | 74 int ProcessStream(AudioFrame* frame) override; |
| 75 int ProcessStream(const float* const* src, | 75 int ProcessStream(const float* const* src, |
| 76 size_t samples_per_channel, | 76 size_t samples_per_channel, |
| 77 int input_sample_rate_hz, | 77 int input_sample_rate_hz, |
| 78 ChannelLayout input_layout, | 78 ChannelLayout input_layout, |
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| 360 ApmRenderState(); | 360 ApmRenderState(); |
| 361 ~ApmRenderState(); | 361 ~ApmRenderState(); |
| 362 std::unique_ptr<AudioConverter> render_converter; | 362 std::unique_ptr<AudioConverter> render_converter; |
| 363 std::unique_ptr<AudioBuffer> render_audio; | 363 std::unique_ptr<AudioBuffer> render_audio; |
| 364 } render_ GUARDED_BY(crit_render_); | 364 } render_ GUARDED_BY(crit_render_); |
| 365 }; | 365 }; |
| 366 | 366 |
| 367 } // namespace webrtc | 367 } // namespace webrtc |
| 368 | 368 |
| 369 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ | 369 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ |
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