Implement operator<< for AudioCodecInfo and AudioCodecSpec

webrtc/api/audio_codecs/audio_format.cc

 /*
  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 90 matching lines @@
       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);
 }
 
+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