WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
This removes one more place where we were unable to handle codecs not
in the built-in set.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2686043006
Cr-Commit-Position: refs/heads/master@{#17370}
diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
index ea05940..450318e 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -157,6 +157,9 @@
// Returns kOK on success, or kFail in case of an error.
virtual int GetAudio(AudioFrame* audio_frame, bool* muted) = 0;
+ // Replaces the current set of decoders with the given one.
+ virtual void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) = 0;
+
// Associates |rtp_payload_type| with |codec| and |codec_name|, and stores the
// information in the codec database. Returns 0 on success, -1 on failure.
// The name is only used to provide information back to the caller about the