AudioCodingModule: Specify decoders using SdpAudioFormat

webrtc/modules/audio_coding/acm2/acm_receiver.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 212 matching lines @@
   }
   if (ret_val != NetEq::kOK) {
     LOG(LERROR) << "AcmReceiver::AddCodec " << acm_codec_id
                 << static_cast<int>(payload_type)
                 << " channels: " << channels;
     return -1;
   }
   return 0;
 }
 
+bool AcmReceiver::AddCodec(int rtp_payload_type,
+                           const SdpAudioFormat& audio_format) {
+  const auto old_format = neteq_->GetDecoderFormat(rtp_payload_type);
+  if (old_format && *old_format == audio_format) {
+    // Re-registering the same codec. Do nothing and return.
+    return true;
+  }
+
+  if (neteq_->RemovePayloadType(rtp_payload_type) != NetEq::kOK &&
+      neteq_->LastError() != NetEq::kDecoderNotFound) {
+    LOG(LERROR) << "AcmReceiver::AddCodec: Could not remove existing decoder"
+                   " for payload type "
+                << rtp_payload_type;
+    return false;
+  }
+
+  const bool success =
+      neteq_->RegisterPayloadType(rtp_payload_type, audio_format);
+  if (!success) {
+    LOG(LERROR) << "AcmReceiver::AddCodec failed for payload type "
+                << rtp_payload_type << ", decoder format " << audio_format;
+  }
+  return success;
+}
+
 void AcmReceiver::FlushBuffers() {
   neteq_->FlushBuffers();
 }
 
 void AcmReceiver::RemoveAllCodecs() {
   rtc::CritScope lock(&crit_sect_);
   neteq_->RemoveAllPayloadTypes();
   last_audio_decoder_ = rtc::Optional<CodecInst>();
   last_packet_sample_rate_hz_ = rtc::Optional<int>();
 }
@@ skipping 111 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   rtc::CritScope lock(&crit_sect_);
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc