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Side by Side Diff: webrtc/modules/audio_processing/audio_processing_impl.h

Issue 2406193002: Made the AudioProcessing class a pure interface. (Closed)
Patch Set: Corrected mock of the AudioProcessing interface Created 4 years, 2 months ago
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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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