Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(305)

Side by Side Diff: webrtc/modules/audio_coding/codecs/audio_format.h

Issue 2388153004: Stop using old AudioCodingModule::RegisterReceiveCodec overloads (Closed)
Patch Set: Created 4 years, 2 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) 2016 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2016 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
11 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_ 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_
12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_ 12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_
13 13
14 #include <map> 14 #include <map>
15 #include <ostream> 15 #include <ostream>
16 #include <string> 16 #include <string>
17 #include <utility> 17 #include <utility>
18 18
19 #include "webrtc/common_types.h"
20
19 namespace webrtc { 21 namespace webrtc {
20 22
21 // SDP specification for a single audio codec. 23 // SDP specification for a single audio codec.
22 // NOTE: This class is still under development and may change without notice. 24 // NOTE: This class is still under development and may change without notice.
23 struct SdpAudioFormat { 25 struct SdpAudioFormat {
24 using Parameters = std::map<std::string, std::string>; 26 using Parameters = std::map<std::string, std::string>;
25 27
26 SdpAudioFormat(const SdpAudioFormat&); 28 SdpAudioFormat(const SdpAudioFormat&);
27 SdpAudioFormat(SdpAudioFormat&&); 29 SdpAudioFormat(SdpAudioFormat&&);
28 SdpAudioFormat(const char* name, int clockrate_hz, int num_channels); 30 SdpAudioFormat(const char* name, int clockrate_hz, int num_channels);
(...skipping 19 matching lines...) Expand all
48 50
49 void swap(SdpAudioFormat& a, SdpAudioFormat& b); 51 void swap(SdpAudioFormat& a, SdpAudioFormat& b);
50 std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf); 52 std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf);
51 53
52 struct AudioCodecSpec { 54 struct AudioCodecSpec {
53 SdpAudioFormat format; 55 SdpAudioFormat format;
54 bool allow_comfort_noise; // This encoder can be used with an external 56 bool allow_comfort_noise; // This encoder can be used with an external
55 // comfort noise generator. 57 // comfort noise generator.
56 }; 58 };
57 59
60 SdpAudioFormat CodecInstToSdp(const CodecInst& codec_inst);
ossu 2016/10/05 13:39:58 You can have this, but not here. CodecInst is deci
kwiberg-webrtc 2016/10/06 12:14:52 OK, makes sense to put the legacy->current convert
61
58 } // namespace webrtc 62 } // namespace webrtc
59 63
60 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_ 64 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698