NetEq: Add codec name and RTP timestamp rate to DecoderInfo

webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc

 /*
  *  Copyright (c) 2012 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 494 matching lines @@
   receiver_.SetMinimumDelay(0);
   receiver_.SetMaximumDelay(0);
   receiver_.FlushBuffers();
 
   // Register RED and CN.
   auto db = RentACodec::Database();
   for (size_t i = 0; i < db.size(); i++) {
     if (IsCodecRED(db[i]) || IsCodecCN(db[i])) {
       if (receiver_.AddCodec(static_cast<int>(i),
                              static_cast<uint8_t>(db[i].pltype), 1,
-                             db[i].plfreq, nullptr) < 0) {
+                             db[i].plfreq, nullptr, db[i].plname) < 0) {
         WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                      "Cannot register master codec.");
         return -1;
       }
     }
   }
   receiver_initialized_ = true;
   return 0;
 }
 
@@ skipping 33 matching lines @@
   // Check if the payload-type is valid.
   if (!RentACodec::IsPayloadTypeValid(codec.pltype)) {
     LOG_F(LS_ERROR) << "Invalid payload type " << codec.pltype << " for "
                     << codec.plname;
     return -1;
   }
 
   // Get |decoder| associated with |codec|. |decoder| is NULL if |codec| does
   // not own its decoder.
   return receiver_.AddCodec(*codec_index, codec.pltype, codec.channels,
-                            codec.plfreq,
-                            codec_manager_.GetAudioDecoder(codec));
+                            codec.plfreq, codec_manager_.GetAudioDecoder(codec),
+                            codec.plname);
 }
 
 int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
     int rtp_payload_type,
     AudioDecoder* external_decoder,
     int sample_rate_hz,
-    int num_channels) {
+    int num_channels,
+    const std::string& name) {
   CriticalSectionScoped lock(acm_crit_sect_.get());
   RTC_DCHECK(receiver_initialized_);
   if (num_channels > 2 || num_channels < 0) {
     LOG_F(LS_ERROR) << "Unsupported number of channels: " << num_channels;
     return -1;
   }
 
   // Check if the payload-type is valid.
   if (!RentACodec::IsPayloadTypeValid(rtp_payload_type)) {
     LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
                     << " for external decoder.";
     return -1;
   }
 
   return receiver_.AddCodec(-1 /* external */, rtp_payload_type, num_channels,
-                            sample_rate_hz, external_decoder);
+                            sample_rate_hz, external_decoder, name);
 }
 
 // Get current received codec.
 int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const {
   CriticalSectionScoped lock(acm_crit_sect_.get());
   return receiver_.LastAudioCodec(current_codec);
 }
 
 // Incoming packet from network parsed and ready for decode.
 int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
@@ skipping 173 matching lines @@
   return receiver_.LeastRequiredDelayMs();
 }
 
 void AudioCodingModuleImpl::GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const {
   receiver_.GetDecodingCallStatistics(call_stats);
 }
 
 }  // namespace acm2
 }  // namespace webrtc