| 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..ab438e1e1d0a7bb27a1ac5c02a97169c7136cc53 100644
|
| --- a/webrtc/api/audio_codecs/audio_format.cc
|
| +++ b/webrtc/api/audio_codecs/audio_format.cc
|
| @@ -77,9 +77,34 @@ std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf) {
|
| return os;
|
| }
|
|
|
| -AudioCodecSpec::AudioCodecSpec(const SdpAudioFormat& format) : format(format) {}
|
| +AudioFormatInfo::AudioFormatInfo()
|
| + : sample_rate_hz(0),
|
| + num_channels(0),
|
| + min_bitrate_bps(0),
|
| + max_bitrate_bps(0) {}
|
|
|
| -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);
|
| +}
|
|
|
| } // namespace webrtc
|
|
|