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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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| 245 | 245 |
| 246 // Returns the RTP timestamp for the last sample delivered by GetAudio(). | 246 // Returns the RTP timestamp for the last sample delivered by GetAudio(). |
| 247 // The return value will be empty if no valid timestamp is available. | 247 // The return value will be empty if no valid timestamp is available. |
| 248 virtual rtc::Optional<uint32_t> GetPlayoutTimestamp() const = 0; | 248 virtual rtc::Optional<uint32_t> GetPlayoutTimestamp() const = 0; |
| 249 | 249 |
| 250 // Returns the sample rate in Hz of the audio produced in the last GetAudio | 250 // Returns the sample rate in Hz of the audio produced in the last GetAudio |
| 251 // call. If GetAudio has not been called yet, the configured sample rate | 251 // call. If GetAudio has not been called yet, the configured sample rate |
| 252 // (Config::sample_rate_hz) is returned. | 252 // (Config::sample_rate_hz) is returned. |
| 253 virtual int last_output_sample_rate_hz() const = 0; | 253 virtual int last_output_sample_rate_hz() const = 0; |
| 254 | 254 |
| 255 // Returns info about the decoder for the given payload type, or an empty |
| 256 // value if we have no decoder for that payload type. |
| 257 virtual rtc::Optional<CodecInst> GetDecoder(int payload_type) const = 0; |
| 258 |
| 255 // Not implemented. | 259 // Not implemented. |
| 256 virtual int SetTargetNumberOfChannels() = 0; | 260 virtual int SetTargetNumberOfChannels() = 0; |
| 257 | 261 |
| 258 // Not implemented. | 262 // Not implemented. |
| 259 virtual int SetTargetSampleRate() = 0; | 263 virtual int SetTargetSampleRate() = 0; |
| 260 | 264 |
| 261 // 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 |
| 262 // occurred, 0 is returned. | 266 // occurred, 0 is returned. |
| 263 virtual int LastError() const = 0; | 267 virtual int LastError() const = 0; |
| 264 | 268 |
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| 288 | 292 |
| 289 protected: | 293 protected: |
| 290 NetEq() {} | 294 NetEq() {} |
| 291 | 295 |
| 292 private: | 296 private: |
| 293 RTC_DISALLOW_COPY_AND_ASSIGN(NetEq); | 297 RTC_DISALLOW_COPY_AND_ASSIGN(NetEq); |
| 294 }; | 298 }; |
| 295 | 299 |
| 296 } // namespace webrtc | 300 } // namespace webrtc |
| 297 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_ | 301 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_ |
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