AudioCodingModule: Specify decoders using SdpAudioFormat

webrtc/modules/audio_coding/acm2/acm_receiver.h

Index: webrtc/modules/audio_coding/acm2/acm_receiver.h
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index 06946f41d8898f5e4edd05a2a06a31cc84f27ba9..ea85456458b6bd3ce23d0ab8d4815197f63d1f4d 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -113,6 +113,10 @@ class AcmReceiver {
                AudioDecoder* audio_decoder,
                const std::string& name);
 
+  // Adds a new decoder to the NetEq codec database. Returns true iff
+  // successful.
+  bool AddCodec(int rtp_payload_type, const SdpAudioFormat& audio_format);
+
   //
   // Sets a minimum delay for packet buffer. The given delay is maintained,
   // unless channel condition dictates a higher delay.