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 ffbcb817e709cb569da1ec09c0e5d43adfd14735..8a0369d0e9ce5ea5c640795d51a6c2737196a178 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
@@ -46,13 +46,11 @@ class RTPSenderInterface { |
virtual uint32_t SSRC() const = 0; |
virtual uint32_t Timestamp() const = 0; |
- virtual int32_t BuildRTPheader(uint8_t* data_buffer, |
danilchap
2016/06/15 13:40:01
this function (with this spelling) is used outside
Sergey Ulanov
2016/06/15 18:27:53
Done.
|
+ virtual int32_t BuildRtpHeader(uint8_t* data_buffer, |
int8_t payload_type, |
bool marker_bit, |
uint32_t capture_timestamp, |
- int64_t capture_time_ms, |
- bool timestamp_provided = true, |
- bool inc_sequence_number = true) = 0; |
+ int64_t capture_time_ms) = 0; |
// This returns the expected header length taking into consideration |
// the optional RTP header extensions that may not be currently active. |
@@ -153,7 +151,7 @@ class RTPSender : public RTPSenderInterface { |
const uint8_t* payload_data, |
size_t payload_size, |
const RTPFragmentationHeader* fragmentation, |
- const RTPVideoHeader* rtp_hdr = NULL); |
+ const RTPVideoHeader* rtp_header); |
// RTP header extension |
int32_t SetTransmissionTimeOffset(int32_t transmission_time_offset); |
@@ -167,7 +165,7 @@ class RTPSender : public RTPSenderInterface { |
size_t RtpHeaderExtensionLength() const; |
- uint16_t BuildRTPHeaderExtension(uint8_t* data_buffer, bool marker_bit) const; |
+ uint16_t BuildRtpHeaderExtension(uint8_t* data_buffer, bool marker_bit) const; |
uint8_t BuildTransmissionTimeOffsetExtension(uint8_t *data_buffer) const; |
uint8_t BuildAudioLevelExtension(uint8_t* data_buffer) const; |
@@ -242,13 +240,11 @@ class RTPSender : public RTPSenderInterface { |
void SetRtxPayloadType(int payload_type, int associated_payload_type); |
// Functions wrapping RTPSenderInterface. |
- int32_t BuildRTPheader(uint8_t* data_buffer, |
+ int32_t BuildRtpHeader(uint8_t* data_buffer, |
int8_t payload_type, |
bool marker_bit, |
uint32_t capture_timestamp, |
- int64_t capture_time_ms, |
- const bool timestamp_provided = true, |
danilchap
2016/06/15 13:40:01
there is use (outside webrtc) of this function wit
Sergey Ulanov
2016/06/15 18:27:53
Added deprecated version with these parameters. No
|
- const bool inc_sequence_number = true) override; |
+ int64_t capture_time_ms) override; |
size_t RtpHeaderLength() const override; |
uint16_t AllocateSequenceNumber(uint16_t packets_to_send) override; |
@@ -300,16 +296,7 @@ class RTPSender : public RTPSenderInterface { |
int32_t SetFecParameters(const FecProtectionParams *delta_params, |
const FecProtectionParams *key_params); |
- size_t SendPadData(size_t bytes, |
danilchap
2016/06/15 13:40:01
this version is used outside webrtc, (with timesta
Sergey Ulanov
2016/06/15 18:27:53
Done.
|
- bool timestamp_provided, |
- uint32_t timestamp, |
- int64_t capture_time_ms); |
- |
- size_t SendPadData(size_t bytes, |
- bool timestamp_provided, |
- uint32_t timestamp, |
- int64_t capture_time_ms, |
- int probe_cluster_id); |
+ size_t SendPadData(size_t bytes, int probe_cluster_id); |
// Called on update of RTP statistics. |
void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback); |