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. |