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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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164 | 164 |
165 // Instructs NetEq to deliver 10 ms of audio data. The data is written to | 165 // Instructs NetEq to deliver 10 ms of audio data. The data is written to |
166 // |output_audio|, which can hold (at least) |max_length| elements. | 166 // |output_audio|, which can hold (at least) |max_length| elements. |
167 // The number of channels that were written to the output is provided in | 167 // The number of channels that were written to the output is provided in |
168 // the output variable |num_channels|, and each channel contains | 168 // the output variable |num_channels|, and each channel contains |
169 // |samples_per_channel| elements. If more than one channel is written, | 169 // |samples_per_channel| elements. If more than one channel is written, |
170 // the samples are interleaved. | 170 // the samples are interleaved. |
171 // The speech type is written to |type|, if |type| is not NULL. | 171 // The speech type is written to |type|, if |type| is not NULL. |
172 // Returns kOK on success, or kFail in case of an error. | 172 // Returns kOK on success, or kFail in case of an error. |
173 virtual int GetAudio(size_t max_length, int16_t* output_audio, | 173 virtual int GetAudio(size_t max_length, int16_t* output_audio, |
174 size_t* samples_per_channel, int* num_channels, | 174 size_t* samples_per_channel, size_t* num_channels, |
175 NetEqOutputType* type) = 0; | 175 NetEqOutputType* type) = 0; |
176 | 176 |
177 // Associates |rtp_payload_type| with |codec| and |codec_name|, and stores the | 177 // Associates |rtp_payload_type| with |codec| and |codec_name|, and stores the |
178 // information in the codec database. Returns 0 on success, -1 on failure. | 178 // information in the codec database. Returns 0 on success, -1 on failure. |
179 // The name is only used to provide information back to the caller about the | 179 // The name is only used to provide information back to the caller about the |
180 // decoders. Hence, the name is arbitrary, and may be empty. | 180 // decoders. Hence, the name is arbitrary, and may be empty. |
181 virtual int RegisterPayloadType(NetEqDecoder codec, | 181 virtual int RegisterPayloadType(NetEqDecoder codec, |
182 const std::string& codec_name, | 182 const std::string& codec_name, |
183 uint8_t rtp_payload_type) = 0; | 183 uint8_t rtp_payload_type) = 0; |
184 | 184 |
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299 | 299 |
300 protected: | 300 protected: |
301 NetEq() {} | 301 NetEq() {} |
302 | 302 |
303 private: | 303 private: |
304 RTC_DISALLOW_COPY_AND_ASSIGN(NetEq); | 304 RTC_DISALLOW_COPY_AND_ASSIGN(NetEq); |
305 }; | 305 }; |
306 | 306 |
307 } // namespace webrtc | 307 } // namespace webrtc |
308 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_ | 308 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_ |
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