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..5d9b10b10617dd6ef88c25a84cdf35e362b96b39 100644 |
--- a/webrtc/api/audio_codecs/audio_format.cc |
+++ b/webrtc/api/audio_codecs/audio_format.cc |
@@ -77,9 +77,31 @@ std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf) { |
return os; |
} |
-AudioCodecSpec::AudioCodecSpec(const SdpAudioFormat& format) : format(format) {} |
+AudioFormatInfo::AudioFormatInfo() |
+ : AudioFormatInfo(0, 0, 0) {} |
kwiberg-webrtc
2017/03/15 13:33:17
Sorry if I've already asked about this, but why is
ossu
2017/03/16 18:03:56
It was necessary for one of the previous incarnati
|
-AudioCodecSpec::AudioCodecSpec(SdpAudioFormat&& format) |
- : format(std::move(format)) {} |
+AudioFormatInfo::AudioFormatInfo(int sample_rate_hz, |
+ int num_channels, |
+ int bitrate_bps) |
+ : AudioFormatInfo(sample_rate_hz, |
+ num_channels, |
+ bitrate_bps, |
+ bitrate_bps, |
+ bitrate_bps) {} |
+ |
+AudioFormatInfo::AudioFormatInfo(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_GE(min_bitrate_bps, 0); |
+ RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps); |
+ RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps); |
kwiberg-webrtc
2017/03/15 13:33:17
Also check that sample_rate_hz and num_channels ar
ossu
2017/03/16 18:03:56
Sure. I've just done the most basic of checks, tha
kwiberg-webrtc
2017/03/17 10:20:01
Excellent, that was exactly what I meant.
|
+} |
} // namespace webrtc |