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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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