| /* | 
 |  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 
 |  * | 
 |  *  Use of this source code is governed by a BSD-style license | 
 |  *  that can be found in the LICENSE file in the root of the source | 
 |  *  tree. An additional intellectual property rights grant can be found | 
 |  *  in the file PATENTS.  All contributing project authors may | 
 |  *  be found in the AUTHORS file in the root of the source tree. | 
 |  */ | 
 |  | 
 | #ifndef API_AUDIO_CODECS_AUDIO_FORMAT_H_ | 
 | #define API_AUDIO_CODECS_AUDIO_FORMAT_H_ | 
 |  | 
 | #include <map> | 
 | #include <ostream> | 
 | #include <string> | 
 | #include <utility> | 
 |  | 
 | #include "api/optional.h" | 
 | #include "rtc_base/checks.h" | 
 |  | 
 | namespace webrtc { | 
 |  | 
 | // SDP specification for a single audio codec. | 
 | // NOTE: This class is still under development and may change without notice. | 
 | struct SdpAudioFormat { | 
 |   using Parameters = std::map<std::string, std::string>; | 
 |  | 
 |   SdpAudioFormat(const SdpAudioFormat&); | 
 |   SdpAudioFormat(SdpAudioFormat&&); | 
 |   SdpAudioFormat(const char* name, int clockrate_hz, size_t num_channels); | 
 |   SdpAudioFormat(const std::string& name, | 
 |                  int clockrate_hz, | 
 |                  size_t num_channels); | 
 |   SdpAudioFormat(const char* name, | 
 |                  int clockrate_hz, | 
 |                  size_t num_channels, | 
 |                  const Parameters& param); | 
 |   SdpAudioFormat(const std::string& name, | 
 |                  int clockrate_hz, | 
 |                  size_t num_channels, | 
 |                  const Parameters& param); | 
 |   ~SdpAudioFormat(); | 
 |  | 
 |   // Returns true if this format is compatible with |o|. In SDP terminology: | 
 |   // would it represent the same codec between an offer and an answer? As | 
 |   // opposed to operator==, this method disregards codec parameters. | 
 |   bool Matches(const SdpAudioFormat& o) const; | 
 |  | 
 |   SdpAudioFormat& operator=(const SdpAudioFormat&); | 
 |   SdpAudioFormat& operator=(SdpAudioFormat&&); | 
 |  | 
 |   friend bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b); | 
 |   friend bool operator!=(const SdpAudioFormat& a, const SdpAudioFormat& b) { | 
 |     return !(a == b); | 
 |   } | 
 |  | 
 |   std::string name; | 
 |   int clockrate_hz; | 
 |   size_t num_channels; | 
 |   Parameters parameters; | 
 | }; | 
 |  | 
 | void swap(SdpAudioFormat& a, SdpAudioFormat& b); | 
 | std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf); | 
 |  | 
 | // Information about how an audio format is treated by the codec implementation. | 
 | // Contains basic information, such as sample rate and number of channels, which | 
 | // isn't uniformly presented by SDP. Also contains flags indicating support for | 
 | // integrating with other parts of WebRTC, like external VAD and comfort noise | 
 | // level calculation. | 
 | // | 
 | // To avoid API breakage, and make the code clearer, AudioCodecInfo should not | 
 | // be directly initializable with any flags indicating optional support. If it | 
 | // were, these initializers would break any time a new flag was added. It's also | 
 | // more difficult to understand: | 
 | //   AudioCodecInfo info{16000, 1, 32000, true, false, false, true, true}; | 
 | // than | 
 | //   AudioCodecInfo info(16000, 1, 32000); | 
 | //   info.allow_comfort_noise = true; | 
 | //   info.future_flag_b = true; | 
 | //   info.future_flag_c = true; | 
 | struct AudioCodecInfo { | 
 |   AudioCodecInfo(int sample_rate_hz, size_t num_channels, int bitrate_bps); | 
 |   AudioCodecInfo(int sample_rate_hz, | 
 |                  size_t num_channels, | 
 |                  int default_bitrate_bps, | 
 |                  int min_bitrate_bps, | 
 |                  int max_bitrate_bps); | 
 |   AudioCodecInfo(const AudioCodecInfo& b) = default; | 
 |   ~AudioCodecInfo() = default; | 
 |  | 
 |   bool operator==(const AudioCodecInfo& b) const { | 
 |     return sample_rate_hz == b.sample_rate_hz && | 
 |            num_channels == b.num_channels && | 
 |            default_bitrate_bps == b.default_bitrate_bps && | 
 |            min_bitrate_bps == b.min_bitrate_bps && | 
 |            max_bitrate_bps == b.max_bitrate_bps && | 
 |            allow_comfort_noise == b.allow_comfort_noise && | 
 |            supports_network_adaption == b.supports_network_adaption; | 
 |   } | 
 |  | 
 |   bool operator!=(const AudioCodecInfo& b) const { return !(*this == b); } | 
 |  | 
 |   bool HasFixedBitrate() const { | 
 |     RTC_DCHECK_GE(min_bitrate_bps, 0); | 
 |     RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps); | 
 |     RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps); | 
 |     return min_bitrate_bps == max_bitrate_bps; | 
 |   } | 
 |  | 
 |   int sample_rate_hz; | 
 |   size_t num_channels; | 
 |   int default_bitrate_bps; | 
 |   int min_bitrate_bps; | 
 |   int max_bitrate_bps; | 
 |  | 
 |   bool allow_comfort_noise = true;  // This codec can be used with an external | 
 |                                     // comfort noise generator. | 
 |   bool supports_network_adaption = false;  // This codec can adapt to varying | 
 |                                            // network conditions. | 
 | }; | 
 |  | 
 | std::ostream& operator<<(std::ostream& os, const AudioCodecInfo& aci); | 
 |  | 
 | // AudioCodecSpec ties an audio format to specific information about the codec | 
 | // and its implementation. | 
 | struct AudioCodecSpec { | 
 |   bool operator==(const AudioCodecSpec& b) const { | 
 |     return format == b.format && info == b.info; | 
 |   } | 
 |  | 
 |   bool operator!=(const AudioCodecSpec& b) const { return !(*this == b); } | 
 |  | 
 |   SdpAudioFormat format; | 
 |   AudioCodecInfo info; | 
 | }; | 
 |  | 
 | std::ostream& operator<<(std::ostream& os, const AudioCodecSpec& acs); | 
 |  | 
 | }  // namespace webrtc | 
 |  | 
 | #endif  // API_AUDIO_CODECS_AUDIO_FORMAT_H_ |