| 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 ea5f1839231814fc92cf11b04aed8d3d4e1610d3..de8b1fd8d4ed9199f04296a50815ca3c845c9401 100644
|
| --- a/webrtc/api/audio_codecs/audio_format.cc
|
| +++ b/webrtc/api/audio_codecs/audio_format.cc
|
| @@ -108,4 +108,23 @@ AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
|
| RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps);
|
| }
|
|
|
| +std::ostream& operator<<(std::ostream& os, const AudioCodecInfo& aci) {
|
| + os << "{sample_rate_hz: " << aci.sample_rate_hz;
|
| + os << ", num_channels: " << aci.num_channels;
|
| + os << ", default_bitrate_bps: " << aci.default_bitrate_bps;
|
| + os << ", min_bitrate_bps: " << aci.min_bitrate_bps;
|
| + os << ", max_bitrate_bps: " << aci.max_bitrate_bps;
|
| + os << ", allow_comfort_noise: " << aci.allow_comfort_noise;
|
| + os << ", supports_network_adaption: " << aci.supports_network_adaption;
|
| + os << "}";
|
| + return os;
|
| +}
|
| +
|
| +std::ostream& operator<<(std::ostream& os, const AudioCodecSpec& acs) {
|
| + os << "{format: " << acs.format;
|
| + os << ", info: " << acs.info;
|
| + os << "}";
|
| + return os;
|
| +}
|
| +
|
| } // namespace webrtc
|
|
|