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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
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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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