RTPPayloadRegistry: Stop using the rate to keep track of receive codecs

webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc

 /*
  *  Copyright (c) 2013 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 11 matching lines @@
 
 namespace {
 
 bool PayloadIsCompatible(const RtpUtility::Payload& payload,
                          const CodecInst& audio_codec) {
   if (!payload.audio)
     return false;
   if (_stricmp(payload.name, audio_codec.plname) != 0)
     return false;
   const AudioPayload& audio_payload = payload.typeSpecific.Audio;
-  const uint32_t rate = std::max(0, audio_codec.rate);
   return audio_payload.frequency == static_cast<uint32_t>(audio_codec.plfreq) &&
-         audio_payload.channels == audio_codec.channels &&
-         (audio_payload.rate == rate || audio_payload.rate == 0 || rate == 0);
+         audio_payload.channels == audio_codec.channels;
 }
 
 bool PayloadIsCompatible(const RtpUtility::Payload& payload,
                          const VideoCodec& video_codec) {
   if (payload.audio || _stricmp(payload.name, video_codec.plName) != 0)
     return false;
   // For H264, profiles must match as well.
   if (video_codec.codecType == kVideoCodecH264) {
     return video_codec.H264().profile ==
            payload.typeSpecific.Video.h264_profile;
   }
   return true;
 }
 
 RtpUtility::Payload CreatePayloadType(const CodecInst& audio_codec) {
   RtpUtility::Payload payload;
   payload.name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
   strncpy(payload.name, audio_codec.plname, RTP_PAYLOAD_NAME_SIZE - 1);
   RTC_DCHECK_GE(audio_codec.plfreq, 1000);
   payload.typeSpecific.Audio.frequency = audio_codec.plfreq;
   payload.typeSpecific.Audio.channels = audio_codec.channels;
-  payload.typeSpecific.Audio.rate = std::max(0, audio_codec.rate);
+  payload.typeSpecific.Audio.rate = 0;
   payload.audio = true;
   return payload;
 }
 
 RtpVideoCodecTypes ConvertToRtpVideoCodecType(VideoCodecType type) {
   switch (type) {
     case kVideoCodecVP8:
       return kRtpVideoVp8;
     case kVideoCodecVP9:
       return kRtpVideoVp9;
@@ skipping 59 matching lines @@
   if (!IsPayloadTypeValid(audio_codec.pltype))
     return -1;
 
   rtc::CritScope cs(&crit_sect_);
 
   auto it = payload_type_map_.find(audio_codec.pltype);
   if (it != payload_type_map_.end()) {
     // We already use this payload type. Check if it's the same as we already
     // have. If same, ignore sending an error.
     if (PayloadIsCompatible(it->second, audio_codec)) {
-      it->second.typeSpecific.Audio.rate = std::max(0, audio_codec.rate);
+      it->second.typeSpecific.Audio.rate = 0;
       return 0;
     }
     LOG(LS_ERROR) << "Payload type already registered: " << audio_codec.pltype;
     return -1;
   }
 
   // Audio codecs must be unique.
   DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(audio_codec);
 
   payload_type_map_[audio_codec.pltype] = CreatePayloadType(audio_codec);
@@ skipping 240 matching lines @@
     const char* payload_name) const {
   rtc::CritScope cs(&crit_sect_);
   for (const auto& it : payload_type_map_) {
     if (_stricmp(it.second.name, payload_name) == 0)
       return it.first;
   }
   return -1;
 }
 
 }  // namespace webrtc