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

Issue 1525173002: Bugfix that fixes the integration issue that cause WebRTC AEC mobile to fail (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Added missing input_sample_rate_hz() to FakeAudioProcessing Created 5 years 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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272 int output_sample_rate_hz, 272 int output_sample_rate_hz,
273 int reverse_sample_rate_hz, 273 int reverse_sample_rate_hz,
274 ChannelLayout input_layout, 274 ChannelLayout input_layout,
275 ChannelLayout output_layout, 275 ChannelLayout output_layout,
276 ChannelLayout reverse_layout) = 0; 276 ChannelLayout reverse_layout) = 0;
277 277
278 // Pass down additional options which don't have explicit setters. This 278 // Pass down additional options which don't have explicit setters. This
279 // ensures the options are applied immediately. 279 // ensures the options are applied immediately.
280 virtual void SetExtraOptions(const Config& config) = 0; 280 virtual void SetExtraOptions(const Config& config) = 0;
281 281
282 // TODO(peah): Remove after voice engine no longer requires it to resample
283 // the reverse stream to the forward rate.
284 virtual int input_sample_rate_hz() const = 0;
285
282 // TODO(ajm): Only intended for internal use. Make private and friend the 286 // TODO(ajm): Only intended for internal use. Make private and friend the
283 // necessary classes? 287 // necessary classes?
284 virtual int proc_sample_rate_hz() const = 0; 288 virtual int proc_sample_rate_hz() const = 0;
285 virtual int proc_split_sample_rate_hz() const = 0; 289 virtual int proc_split_sample_rate_hz() const = 0;
286 virtual int num_input_channels() const = 0; 290 virtual int num_input_channels() const = 0;
287 virtual int num_output_channels() const = 0; 291 virtual int num_output_channels() const = 0;
288 virtual int num_reverse_channels() const = 0; 292 virtual int num_reverse_channels() const = 0;
289 293
290 // Set to true when the output of AudioProcessing will be muted or in some 294 // Set to true when the output of AudioProcessing will be muted or in some
291 // other way not used. Ideally, the captured audio would still be processed, 295 // other way not used. Ideally, the captured audio would still be processed,
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944 // This does not impact the size of frames passed to |ProcessStream()|. 948 // This does not impact the size of frames passed to |ProcessStream()|.
945 virtual int set_frame_size_ms(int size) = 0; 949 virtual int set_frame_size_ms(int size) = 0;
946 virtual int frame_size_ms() const = 0; 950 virtual int frame_size_ms() const = 0;
947 951
948 protected: 952 protected:
949 virtual ~VoiceDetection() {} 953 virtual ~VoiceDetection() {}
950 }; 954 };
951 } // namespace webrtc 955 } // namespace webrtc
952 956
953 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_ 957 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
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