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Side by Side Diff: webrtc/modules/audio_coding/neteq/include/neteq.h

Issue 1863993002: Move setting of AudioFrame::timestamp_ into NetEq (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@neteq-empty-playout-ts
Patch Set: Created 4 years, 8 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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239 239
240 // Enables post-decode VAD. When enabled, GetAudio() will return 240 // Enables post-decode VAD. When enabled, GetAudio() will return
241 // kOutputVADPassive when the signal contains no speech. 241 // kOutputVADPassive when the signal contains no speech.
242 virtual void EnableVad() = 0; 242 virtual void EnableVad() = 0;
243 243
244 // Disables post-decode VAD. 244 // Disables post-decode VAD.
245 virtual void DisableVad() = 0; 245 virtual void DisableVad() = 0;
246 246
247 // Returns the RTP timestamp for the last sample delivered by GetAudio(). 247 // Returns the RTP timestamp for the last sample delivered by GetAudio().
248 // The return value will be empty if no valid timestamp is available. 248 // The return value will be empty if no valid timestamp is available.
249 virtual rtc::Optional<uint32_t> GetPlayoutTimestamp() = 0; 249 virtual rtc::Optional<uint32_t> GetPlayoutTimestamp() const = 0;
hlundin-webrtc 2016/04/06 09:34:07 This was always essentially a const method, but it
250 250
251 // Returns the sample rate in Hz of the audio produced in the last GetAudio 251 // Returns the sample rate in Hz of the audio produced in the last GetAudio
252 // call. If GetAudio has not been called yet, the configured sample rate 252 // call. If GetAudio has not been called yet, the configured sample rate
253 // (Config::sample_rate_hz) is returned. 253 // (Config::sample_rate_hz) is returned.
254 virtual int last_output_sample_rate_hz() const = 0; 254 virtual int last_output_sample_rate_hz() const = 0;
255 255
256 // Not implemented. 256 // Not implemented.
257 virtual int SetTargetNumberOfChannels() = 0; 257 virtual int SetTargetNumberOfChannels() = 0;
258 258
259 // Not implemented. 259 // Not implemented.
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289 289
290 protected: 290 protected:
291 NetEq() {} 291 NetEq() {}
292 292
293 private: 293 private:
294 RTC_DISALLOW_COPY_AND_ASSIGN(NetEq); 294 RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
295 }; 295 };
296 296
297 } // namespace webrtc 297 } // namespace webrtc
298 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_ 298 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
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