webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h - Issue 2545753002: Deprecated SetAudioPacketSize from RTPSender and removed calls to it.

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

Issue 2545753002: Deprecated SetAudioPacketSize from RTPSender and removed calls to it. (Closed)

Patch Set: constexpr; parentheses Created 4 years 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 4808a4e7eadc984ac1a6591df1de5fab11ab9acb..7f34957da56824c8a63f3f235a24c3e451237d69 100644

--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h

+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h

@@ -260,8 +260,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {

// Audio part.

- // Set audio packet size, used to determine when it's time to send a DTMF

- // packet in silence (CNG).

+ // 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).