NetEq: Add codec name and RTP timestamp rate to DecoderInfo

webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
 
+#include <string>
 #include <vector>
 
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/acm2/codec_manager.h"
@@ skipping 92 matching lines @@
   // Get current playout frequency.
   int PlayoutFrequency() const override;
 
   // Register possible receive codecs, can be called multiple times,
   // for codecs, CNG, DTMF, RED.
   int RegisterReceiveCodec(const CodecInst& receive_codec) override;
 
   int RegisterExternalReceiveCodec(int rtp_payload_type,
                                    AudioDecoder* external_decoder,
                                    int sample_rate_hz,
-                                   int num_channels) override;
+                                   int num_channels,
+                                   const std::string& name) override;
 
   // Get current received codec.
   int ReceiveCodec(CodecInst* current_codec) const override;
 
   // Incoming packet from network parsed and ready for decode.
   int IncomingPacket(const uint8_t* incoming_payload,
                      const size_t payload_length,
                      const WebRtcRTPHeader& rtp_info) override;
 
   // Incoming payloads, without rtp-info, the rtp-info will be created in ACM.
@@ skipping 134 matching lines @@
   const rtc::scoped_ptr<CriticalSectionWrapper> callback_crit_sect_;
   AudioPacketizationCallback* packetization_callback_
       GUARDED_BY(callback_crit_sect_);
   ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_);
 };
 
 }  // namespace acm2
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_