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 dc35a038e1de6f9d09cc24eec04ff326f6e039c0..b0165c628da023c9a8925ffb63e410ece64b2f2d 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
@@ -17,6 +17,7 @@ |
#include <vector> |
#include "webrtc/api/call/transport.h" |
+#include "webrtc/base/array_view.h" |
#include "webrtc/base/constructormagic.h" |
#include "webrtc/base/criticalsection.h" |
#include "webrtc/base/deprecation.h" |
@@ -71,9 +72,6 @@ class RTPSender { |
uint32_t FecOverheadRate() const; |
uint32_t NackOverheadRate() const; |
- // Excluding size of RTP and FEC headers. |
- size_t MaxPayloadSize() const; |
- |
int32_t RegisterPayload(const char* payload_name, |
const int8_t payload_type, |
const uint32_t frequency, |
@@ -148,6 +146,9 @@ class RTPSender { |
void SetRtxPayloadType(int payload_type, int associated_payload_type); |
+ // Size info for header extensions used by FEC packets. |
+ static rtc::ArrayView<const RtpExtensionSize> FecExtensionSizes(); |
+ |
// Create empty packet, fills ssrc, csrcs and reserve place for header |
// extensions RtpSender updates before sending. |
std::unique_ptr<RtpPacketToSend> AllocatePacket() const; |
@@ -156,6 +157,7 @@ class RTPSender { |
// Return false if sending was turned off. |
bool AssignSequenceNumber(RtpPacketToSend* packet); |
+ // Used for padding and FEC packets only. |
size_t RtpHeaderLength() const; |
uint16_t AllocateSequenceNumber(uint16_t packets_to_send); |
// Including RTP headers. |