| 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 7a6677b408e29b29a716d996b761d4059d0607d5..6c9928c79312b98ce505d0b16b05c98adb10d397 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
|
| @@ -203,6 +203,14 @@ class RTPSender {
|
|
|
| void SetRtxPayloadType(int payload_type, int associated_payload_type);
|
|
|
| + // Create empty packet, fills ssrc, csrcs and reserve place for header
|
| + // extensions RtpSender updates before sending.
|
| + std::unique_ptr<RtpPacketToSend> AllocatePacket() const;
|
| + // Allocate sequence number for provided packet.
|
| + // Save packet's fields to generate padding that doesn't break media stream.
|
| + // Return false if sending was turned off.
|
| + bool AssignSequenceNumber(RtpPacketToSend* packet);
|
| +
|
| // Functions wrapping RTPSenderInterface.
|
| int32_t BuildRTPheader(uint8_t* data_buffer,
|
| int8_t payload_type,
|
|
|