Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(270)

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

Issue 2351183002: AcmReceiver: Eliminate AcmReceiver::decoders_ (Closed)
Patch Set: case-insensitive string comparison Created 4 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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 241 matching lines...) Expand 10 before | Expand all | Expand 10 after
252 252
253 // Returns the sample rate in Hz of the audio produced in the last GetAudio 253 // Returns the sample rate in Hz of the audio produced in the last GetAudio
254 // call. If GetAudio has not been called yet, the configured sample rate 254 // call. If GetAudio has not been called yet, the configured sample rate
255 // (Config::sample_rate_hz) is returned. 255 // (Config::sample_rate_hz) is returned.
256 virtual int last_output_sample_rate_hz() const = 0; 256 virtual int last_output_sample_rate_hz() const = 0;
257 257
258 // Returns info about the decoder for the given payload type, or an empty 258 // Returns info about the decoder for the given payload type, or an empty
259 // value if we have no decoder for that payload type. 259 // value if we have no decoder for that payload type.
260 virtual rtc::Optional<CodecInst> GetDecoder(int payload_type) const = 0; 260 virtual rtc::Optional<CodecInst> GetDecoder(int payload_type) const = 0;
261 261
262 // Returns the decoder format for the given payload type. Returns null if no
263 // such payload type was registered, or if it was registered without
264 // providing an SdpAudioFormat.
265 virtual const SdpAudioFormat* GetDecoderFormat(int payload_type) const = 0;
266
262 // Not implemented. 267 // Not implemented.
263 virtual int SetTargetNumberOfChannels() = 0; 268 virtual int SetTargetNumberOfChannels() = 0;
264 269
265 // Not implemented. 270 // Not implemented.
266 virtual int SetTargetSampleRate() = 0; 271 virtual int SetTargetSampleRate() = 0;
267 272
268 // Returns the error code for the last occurred error. If no error has 273 // Returns the error code for the last occurred error. If no error has
269 // occurred, 0 is returned. 274 // occurred, 0 is returned.
270 virtual int LastError() const = 0; 275 virtual int LastError() const = 0;
271 276
(...skipping 23 matching lines...) Expand all
295 300
296 protected: 301 protected:
297 NetEq() {} 302 NetEq() {}
298 303
299 private: 304 private:
300 RTC_DISALLOW_COPY_AND_ASSIGN(NetEq); 305 RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
301 }; 306 };
302 307
303 } // namespace webrtc 308 } // namespace webrtc
304 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_ 309 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
OLDNEW
« no previous file with comments | « webrtc/modules/audio_coding/neteq/decoder_database.h ('k') | webrtc/modules/audio_coding/neteq/neteq_impl.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698