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Unified Diff: webrtc/modules/rtp_rtcp/include/rtp_rtcp.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/include/rtp_rtcp.h
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
index 693a6f8373ea351c8ffc47791bf742dc8f03e4f5..29c3e02efaf9de2055e2cb9287f7a0e438bbfaf2 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -413,12 +413,6 @@ class RtpRtcp : public Module {
// Audio
// **************************************************************************
- // This function is deprecated. It was previously used to determine when it
- // was time to send a DTMF packet in silence (CNG).
- // Returns -1 on failure else 0.
- RTC_DEPRECATED virtual int32_t SetAudioPacketSize(
- uint16_t packet_size_samples) = 0;
-
// Sends a TelephoneEvent tone using RFC 2833 (4733).
// Returns -1 on failure else 0.
virtual int32_t SendTelephoneEventOutband(uint8_t key,
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