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 |