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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender.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_sender.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index d4aca38c907254c7b61e6b4b259fb472497acb7f..9ab3e33f8a8832bba63ad2238e5df74e689f0e43 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -176,10 +176,6 @@ class RTPSender {
// Send a DTMF tone using RFC 2833 (4733).
int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
- // This function is deprecated. It was previously used to determine when it
- // was time to send a DTMF packet in silence (CNG).
- RTC_DEPRECATED int32_t SetAudioPacketSize(uint16_t packet_size_samples);
-
// Store the audio level in d_bov for
// header-extension-for-audio-level-indication.
int32_t SetAudioLevel(uint8_t level_d_bov);
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