Allow a received audio codec's payload type to change.

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 27 matching lines @@
 
 SdpAudioFormat::SdpAudioFormat(const std::string& name,
                                int clockrate_hz,
                                size_t num_channels,
                                const Parameters& param)
     : name(name),
       clockrate_hz(clockrate_hz),
       num_channels(num_channels),
       parameters(param) {}
 
+bool SdpAudioFormat::Matches(const SdpAudioFormat& o) const {
+  return STR_CASE_CMP(name.c_str(), o.name.c_str()) == 0 &&
+         clockrate_hz == o.clockrate_hz && num_channels == o.num_channels;
+}
+
 SdpAudioFormat::~SdpAudioFormat() = default;
 SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default;
 SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default;
 
 bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b) {
   return STR_CASE_CMP(a.name.c_str(), b.name.c_str()) == 0 &&
          a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels &&
          a.parameters == b.parameters;
 }
 
@@ skipping 39 matching lines @@
       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);
 }
 
 }  // namespace webrtc