AudioCodingModule: Specify decoders using SdpAudioFormat

webrtc/modules/audio_coding/acm2/audio_coding_module.cc

Index: webrtc/modules/audio_coding/acm2/audio_coding_module.cc
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
index 99b539ab6440745769efcd58606f9fc6823cf4ca..a845c017d1bd4efce439a5b23fbf3d3bc2475114 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -121,6 +121,9 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   // 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,
@@ -987,6 +990,21 @@ int AudioCodingModuleImpl::PlayoutFrequency() const {
   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();