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Unified Diff: webrtc/modules/rtp_rtcp/include/rtp_receiver.h

Issue 1848813003: Remove deprecated RtpReceiver::CreateAudioReceiver() function. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 9 months ago
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Index: webrtc/modules/rtp_rtcp/include/rtp_receiver.h
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_receiver.h b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
index 0a3e201c749b75cc74bc502afd241b1c9e671e59..f393e41eabd6110afc1f685c333f6bb21556d05b 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
@@ -16,7 +16,6 @@
namespace webrtc {
-class RtpAudioFeedback;
class RTPPayloadRegistry;
class TelephoneEventHandler {
@@ -50,15 +49,6 @@ class RtpReceiver {
RtpFeedback* incoming_messages_callback,
RTPPayloadRegistry* rtp_payload_registry);
- // DEPRECATED: Creates an audio-enabled RTP receiver.
- // TODO(solenberg): Remove, after updating downstream code.
- static RtpReceiver* CreateAudioReceiver(
- Clock* clock,
- RtpAudioFeedback* incoming_audio_feedback,
- RtpData* incoming_payload_callback,
- RtpFeedback* incoming_messages_callback,
- RTPPayloadRegistry* rtp_payload_registry);
-
virtual ~RtpReceiver() {}
// Returns a TelephoneEventHandler if available.

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