Chromium Code Reviews| Index: webrtc/api/audio_codecs/audio_format.h |
| diff --git a/webrtc/api/audio_codecs/audio_format.h b/webrtc/api/audio_codecs/audio_format.h |
| index db3990f1143bf5fffc5ff233abb837dda79b533c..08bcd38008bf24e04c4b98b50d951ac692a3040c 100644 |
| --- a/webrtc/api/audio_codecs/audio_format.h |
| +++ b/webrtc/api/audio_codecs/audio_format.h |
| @@ -16,6 +16,8 @@ |
| #include <string> |
| #include <utility> |
| +#include "webrtc/base/optional.h" |
| + |
| namespace webrtc { |
| // SDP specification for a single audio codec. |
| @@ -54,28 +56,71 @@ struct SdpAudioFormat { |
| void swap(SdpAudioFormat& a, SdpAudioFormat& b); |
| std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf); |
| -// To avoid API breakage, and make the code clearer, AudioCodecSpec should not |
| +// 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: |
| -// AudioCodecSpec spec{{"format", 8000, 1}, true, false, false, true, true}; |
| +// AudioCodecInfo info{16000, 1, 32000, true, false, false, true, true}; |
| // than |
| -// AudioCodecSpec spec({"format", 8000, 1}); |
| -// spec.allow_comfort_noise = true; |
| -// spec.future_flag_b = true; |
| -// spec.future_flag_c = true; |
| -struct AudioCodecSpec { |
| - explicit AudioCodecSpec(const SdpAudioFormat& format); |
| - explicit AudioCodecSpec(SdpAudioFormat&& format); |
| - ~AudioCodecSpec() = default; |
| +// 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, int num_channels, int bitrate_bps); |
| + AudioCodecInfo(int sample_rate_hz, |
| + int num_channels, |
| + int default_bitrate_bps, |
| + int min_bitrate_bps, |
| + int max_bitrate_bps); |
| + AudioCodecInfo(const AudioCodecInfo& b) = default; |
| + ~AudioCodecInfo() = default; |
| - SdpAudioFormat format; |
| - bool allow_comfort_noise = true; // This codec can be used with an external |
| - // comfort noise generator. |
| + 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(min_bitrate_bps != max_bitrate_bps || |
| + min_bitrate_bps == default_bitrate_bps == max_bitrate_bps); |
|
the sun
2017/03/17 09:07:31
Do you really want to compare max_bitrate_bps with
ossu
2017/03/17 09:55:39
I... no, I don't know what I was thinking here. I'
|
| + return min_bitrate_bps == max_bitrate_bps; |
| + } |
| + |
| + int sample_rate_hz; |
| + int 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. |
| }; |
| +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; |
| +}; |
| + |
| } // namespace webrtc |
| #endif // WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_ |