Chromium Code Reviews| 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); |