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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender.cc

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.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index dc7b20e9db31fb673471d012d9e6610dcca16d29..024e028393057c54e2772945c8d49f0f17cb1aea 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -1132,13 +1132,6 @@ int32_t RTPSender::SendTelephoneEvent(uint8_t key,
return audio_->SendTelephoneEvent(key, time_ms, level);
}
-int32_t RTPSender::SetAudioPacketSize(uint16_t packet_size_samples) {
- if (!audio_configured_) {
- return -1;
- }
- return 0;
-}
-
int32_t RTPSender::SetAudioLevel(uint8_t level_d_bov) {
return audio_->SetAudioLevel(level_d_bov);
}
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