AudioCodingModule: Specify decoders using SdpAudioFormat

webrtc/modules/audio_coding/acm2/audio_coding_module.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 103 matching lines @@
 
   // Initialize receiver, resets codec database etc.
   int InitializeReceiver() override;
 
   // Get current receive frequency.
   int ReceiveFrequency() const override;
 
   // Get current playout frequency.
   int PlayoutFrequency() const override;
 
+  bool RegisterReceiveCodec(int rtp_payload_type,
+                            const SdpAudioFormat& audio_format) override;
+
   int RegisterReceiveCodec(const CodecInst& receive_codec) override;
   int RegisterReceiveCodec(
       const CodecInst& receive_codec,
       rtc::FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory) override;
 
   int RegisterExternalReceiveCodec(int rtp_payload_type,
                                    AudioDecoder* external_decoder,
                                    int sample_rate_hz,
                                    int num_channels,
                                    const std::string& name) override;
@@ skipping 846 matching lines @@
                                  : receiver_.last_output_sample_rate_hz();
 }
 
 // Get current playout frequency.
 int AudioCodingModuleImpl::PlayoutFrequency() const {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
                "PlayoutFrequency()");
   return receiver_.last_output_sample_rate_hz();
 }
 
+bool AudioCodingModuleImpl::RegisterReceiveCodec(

[hlundin-webrtc, 2016/10/05 13:39:14]

Why do you not have to handle the "iSAC case" like we do in the other
RegisterReceiveCodec methods?

[kwiberg-webrtc, 2016/10/06 09:20:46]

In principle: Because the user has previously injected (or had the option of
injecting) the factory that'll be used to instantiate the decoders. That factory
can take care of things like coupling the state of iSAC decoders with iSAC
encoders.

In practice: Because we never used the same ACM for sending and receiving, the
encoder and decoder were never coupled anyway.

+    int rtp_payload_type,
+    const SdpAudioFormat& audio_format) {
+  rtc::CritScope lock(&acm_crit_sect_);
+  RTC_DCHECK(receiver_initialized_);
+
+  if (!acm2::RentACodec::IsPayloadTypeValid(rtp_payload_type)) {
+    LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
+                    << " for decoder.";
+    return false;
+  }
+
+  return receiver_.AddCodec(rtp_payload_type, audio_format);
+}
+
 int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) {
   rtc::CritScope lock(&acm_crit_sect_);
   auto* ef = encoder_factory_.get();
   return RegisterReceiveCodecUnlocked(
       codec, [&] { return ef->rent_a_codec.RentIsacDecoder(codec.plfreq); });
 }
 
 int AudioCodingModuleImpl::RegisterReceiveCodec(
     const CodecInst& codec,
     rtc::FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory) {
@@ skipping 356 matching lines @@
 // Checks the validity of the parameters of the given codec
 bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
   bool valid = acm2::RentACodec::IsCodecValid(codec);
   if (!valid)
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
                  "Invalid codec setting");
   return valid;
 }
 
 }  // namespace webrtc