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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_API_AUDIO_CODECS_AUDIO_FORMAT_H_ | 11 #ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_ |
12 #define WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_ | 12 #define WEBRTC_API_AUDIO_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/base/optional.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); |
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47 | 49 |
48 std::string name; | 50 std::string name; |
49 int clockrate_hz; | 51 int clockrate_hz; |
50 int num_channels; | 52 int num_channels; |
51 Parameters parameters; | 53 Parameters parameters; |
52 }; | 54 }; |
53 | 55 |
54 void swap(SdpAudioFormat& a, SdpAudioFormat& b); | 56 void swap(SdpAudioFormat& a, SdpAudioFormat& b); |
55 std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf); | 57 std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf); |
56 | 58 |
57 // To avoid API breakage, and make the code clearer, AudioCodecSpec should not | 59 // Information about how an audio format is treated by the codec implementation. |
| 60 // Contains basic information, such as sample rate and number of channels, which |
| 61 // isn't uniformly presented by SDP. Also contains flags indicating support for |
| 62 // integrating with other parts of WebRTC, like external VAD and comfort noise |
| 63 // level calculation. |
| 64 // |
| 65 // To avoid API breakage, and make the code clearer, AudioCodecInfo should not |
58 // be directly initializable with any flags indicating optional support. If it | 66 // be directly initializable with any flags indicating optional support. If it |
59 // were, these initializers would break any time a new flag was added. It's also | 67 // were, these initializers would break any time a new flag was added. It's also |
60 // more difficult to understand: | 68 // more difficult to understand: |
61 // AudioCodecSpec spec{{"format", 8000, 1}, true, false, false, true, true}; | 69 // AudioCodecInfo info{16000, 1, 32000, true, false, false, true, true}; |
62 // than | 70 // than |
63 // AudioCodecSpec spec({"format", 8000, 1}); | 71 // AudioCodecInfo info(16000, 1, 32000); |
64 // spec.allow_comfort_noise = true; | 72 // info.allow_comfort_noise = true; |
65 // spec.future_flag_b = true; | 73 // info.future_flag_b = true; |
66 // spec.future_flag_c = true; | 74 // info.future_flag_c = true; |
| 75 struct AudioCodecInfo { |
| 76 AudioCodecInfo(int sample_rate_hz, int num_channels, int bitrate_bps); |
| 77 AudioCodecInfo(int sample_rate_hz, |
| 78 int num_channels, |
| 79 int default_bitrate_bps, |
| 80 int min_bitrate_bps, |
| 81 int max_bitrate_bps); |
| 82 AudioCodecInfo(const AudioCodecInfo& b) = default; |
| 83 ~AudioCodecInfo() = default; |
| 84 |
| 85 bool operator==(const AudioCodecInfo& b) const { |
| 86 return sample_rate_hz == b.sample_rate_hz && |
| 87 num_channels == b.num_channels && |
| 88 default_bitrate_bps == b.default_bitrate_bps && |
| 89 min_bitrate_bps == b.min_bitrate_bps && |
| 90 max_bitrate_bps == b.max_bitrate_bps && |
| 91 allow_comfort_noise == b.allow_comfort_noise && |
| 92 supports_network_adaption == b.supports_network_adaption; |
| 93 } |
| 94 |
| 95 bool operator!=(const AudioCodecInfo& b) const { |
| 96 return !(*this == b); |
| 97 } |
| 98 |
| 99 bool HasFixedBitrate() const { |
| 100 RTC_DCHECK_GE(min_bitrate_bps, 0); |
| 101 RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps); |
| 102 RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps); |
| 103 return min_bitrate_bps == max_bitrate_bps; |
| 104 } |
| 105 |
| 106 int sample_rate_hz; |
| 107 int num_channels; |
| 108 int default_bitrate_bps; |
| 109 int min_bitrate_bps; |
| 110 int max_bitrate_bps; |
| 111 |
| 112 bool allow_comfort_noise = true; // This codec can be used with an external |
| 113 // comfort noise generator. |
| 114 bool supports_network_adaption = false; // This codec can adapt to varying |
| 115 // network conditions. |
| 116 }; |
| 117 |
| 118 // AudioCodecSpec ties an audio format to specific information about the codec |
| 119 // and its implementation. |
67 struct AudioCodecSpec { | 120 struct AudioCodecSpec { |
68 explicit AudioCodecSpec(const SdpAudioFormat& format); | 121 bool operator==(const AudioCodecSpec& b) const { |
69 explicit AudioCodecSpec(SdpAudioFormat&& format); | 122 return format == b.format && info == b.info; |
70 ~AudioCodecSpec() = default; | 123 } |
| 124 |
| 125 bool operator!=(const AudioCodecSpec& b) const { |
| 126 return !(*this == b); |
| 127 } |
71 | 128 |
72 SdpAudioFormat format; | 129 SdpAudioFormat format; |
73 bool allow_comfort_noise = true; // This codec can be used with an external | 130 AudioCodecInfo info; |
74 // comfort noise generator. | |
75 bool supports_network_adaption = false; // This codec can adapt to varying | |
76 // network conditions. | |
77 }; | 131 }; |
78 | 132 |
79 } // namespace webrtc | 133 } // namespace webrtc |
80 | 134 |
81 #endif // WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_ | 135 #endif // WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_ |
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