Add support for external decoders in ACM

webrtc/modules/audio_coding/main/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 649 matching lines @@
     return -1;
   }
 
   // Get |decoder| associated with |codec|. |decoder| is NULL if |codec| does
   // not own its decoder.
   return receiver_.AddCodec(codec_id, codec.pltype, codec.channels,
                             codec.plfreq,
                             codec_manager_.GetAudioDecoder(codec));
 }
 
+int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
+    int rtp_payload_type,
+    AudioDecoder* external_decoder,
+    int sample_rate_hz,
+    int num_channels) {
+  CriticalSectionScoped lock(acm_crit_sect_);
+  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 (!ACMCodecDB::ValidPayloadType(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);
+}
+
 // Get current received codec.
 int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const {
   CriticalSectionScoped lock(acm_crit_sect_);
   return receiver_.LastAudioCodec(current_codec);
 }
 
 // Incoming packet from network parsed and ready for decode.
 int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
                                           const size_t payload_length,
                                           const WebRtcRTPHeader& rtp_header) {
@@ skipping 593 matching lines @@
       *channels = 1;
       break;
 #endif
     default:
       FATAL() << "Codec type " << codec_type << " not supported.";
   }
   return true;
 }
 
 }  // namespace webrtc