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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h

Issue 2986793002: Remove deprecated RtpRtcp::SetAudioPacketSize (Closed)
Patch Set: Created 3 years, 5 months ago
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Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 39a40f4470070886d9106edf9525849b729e7ad3..2f59d14b4d4797af064daa9ca3a33f0968196aeb 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -253,10 +253,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
// Audio part.
- // This function is deprecated. It was previously used to determine when it
- // was time to send a DTMF packet in silence (CNG).
- int32_t SetAudioPacketSize(uint16_t packet_size_samples) override;
-
// Send a TelephoneEvent tone using RFC 2833 (4733).
int32_t SendTelephoneEventOutband(uint8_t key,
uint16_t time_ms,
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