AudioCodingModule: Specify decoders using SdpAudioFormat
NetEq already uses SdpAudioFormat internally; this CL adds an
AudioCodingModule::RegisterReceiveCodec overload that accepts
SdpAudioFormat, and propagates it through AcmReceiver into NetEq.
The intention is to get rid of the other ways to specify decoders and
always use SdpAudioFormat. (And eventually to do the same for encoders
too.)
NOTRY=true
BUG=5801
Review-Url: https://codereview.webrtc.org/2365653004
Cr-Commit-Position: refs/heads/master@{#14506}
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index 06946f4..ea85456 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -113,6 +113,10 @@
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.