OLD | NEW |
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 |
(...skipping 233 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
244 // kOutputVADPassive when the signal contains no speech. | 244 // kOutputVADPassive when the signal contains no speech. |
245 virtual void EnableVad() = 0; | 245 virtual void EnableVad() = 0; |
246 | 246 |
247 // Disables post-decode VAD. | 247 // Disables post-decode VAD. |
248 virtual void DisableVad() = 0; | 248 virtual void DisableVad() = 0; |
249 | 249 |
250 // Gets the RTP timestamp for the last sample delivered by GetAudio(). | 250 // Gets the RTP timestamp for the last sample delivered by GetAudio(). |
251 // Returns true if the RTP timestamp is valid, otherwise false. | 251 // Returns true if the RTP timestamp is valid, otherwise false. |
252 virtual bool GetPlayoutTimestamp(uint32_t* timestamp) = 0; | 252 virtual bool GetPlayoutTimestamp(uint32_t* timestamp) = 0; |
253 | 253 |
| 254 // Returns the sample rate in Hz of the audio produced in the last GetAudio |
| 255 // call. If GetAudio has not been called yet, the configured sample rate |
| 256 // (Config::sample_rate_hz) is returned. |
| 257 virtual int last_output_sample_rate_hz() const = 0; |
| 258 |
254 // Not implemented. | 259 // Not implemented. |
255 virtual int SetTargetNumberOfChannels() = 0; | 260 virtual int SetTargetNumberOfChannels() = 0; |
256 | 261 |
257 // Not implemented. | 262 // Not implemented. |
258 virtual int SetTargetSampleRate() = 0; | 263 virtual int SetTargetSampleRate() = 0; |
259 | 264 |
260 // Returns the error code for the last occurred error. If no error has | 265 // Returns the error code for the last occurred error. If no error has |
261 // occurred, 0 is returned. | 266 // occurred, 0 is returned. |
262 virtual int LastError() const = 0; | 267 virtual int LastError() const = 0; |
263 | 268 |
(...skipping 23 matching lines...) Expand all Loading... |
287 | 292 |
288 protected: | 293 protected: |
289 NetEq() {} | 294 NetEq() {} |
290 | 295 |
291 private: | 296 private: |
292 RTC_DISALLOW_COPY_AND_ASSIGN(NetEq); | 297 RTC_DISALLOW_COPY_AND_ASSIGN(NetEq); |
293 }; | 298 }; |
294 | 299 |
295 } // namespace webrtc | 300 } // namespace webrtc |
296 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_ | 301 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_ |
OLD | NEW |