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_ |