Index: webrtc/api/audio_codecs/audio_format.cc |
diff --git a/webrtc/api/audio_codecs/audio_format.cc b/webrtc/api/audio_codecs/audio_format.cc |
index b0a86e25bd8a7c2ae225ed539666cbf879cf03a5..67fdf8d7ee7b43428df883719513aa469918d5de 100644 |
--- a/webrtc/api/audio_codecs/audio_format.cc |
+++ b/webrtc/api/audio_codecs/audio_format.cc |
@@ -77,9 +77,30 @@ std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf) { |
return os; |
} |
-AudioCodecSpec::AudioCodecSpec(const SdpAudioFormat& format) : format(format) {} |
+AudioCodecInfo::AudioCodecInfo(int sample_rate_hz, |
+ int num_channels, |
+ int bitrate_bps) |
+ : AudioCodecInfo(sample_rate_hz, |
+ num_channels, |
+ bitrate_bps, |
+ bitrate_bps, |
+ bitrate_bps) {} |
-AudioCodecSpec::AudioCodecSpec(SdpAudioFormat&& format) |
- : format(std::move(format)) {} |
+AudioCodecInfo::AudioCodecInfo(int sample_rate_hz, |
+ int num_channels, |
+ int default_bitrate_bps, |
+ int min_bitrate_bps, |
+ int max_bitrate_bps) |
+ : sample_rate_hz(sample_rate_hz), |
+ num_channels(num_channels), |
+ default_bitrate_bps(default_bitrate_bps), |
+ min_bitrate_bps(min_bitrate_bps), |
+ max_bitrate_bps(max_bitrate_bps) { |
+ RTC_DCHECK_GT(sample_rate_hz, 0); |
+ RTC_DCHECK_GT(num_channels, 0); |
+ 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); |
+} |
} // namespace webrtc |