webrtc/modules/rtp_rtcp/source/rtp_sender.h - Issue 2067673004: Style cleanups in RtpSender.

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

Issue 2067673004: Style cleanups in RtpSender. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: . Created 4 years, 6 months ago

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« webrtc/modules/rtp_rtcp/include/rtp_rtcp.h ('K') | « webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_sender.cc » ('j') | webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h » ('J')
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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..efdbe442f074f173274a973729bd6acbabf23c16 100644

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

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

@@ -46,6 +46,9 @@ class RTPSenderInterface {

virtual uint32_t SSRC() const = 0;

virtual uint32_t Timestamp() const = 0;

+ // Deprecated version of BuildRtpHeader(). |timestamp_provided| and

+ // |inc_sequence_number| are ignored.

+ // TODO(sergeyu): Remove this method.

virtual int32_t BuildRTPheader(uint8_t* data_buffer,

int8_t payload_type,

bool marker_bit,

@@ -54,6 +57,12 @@ class RTPSenderInterface {

bool timestamp_provided = true,

bool inc_sequence_number = true) = 0;

+ virtual int32_t BuildRtpHeader(uint8_t* data_buffer,

+ int8_t payload_type,

stefan-webrtc 2016/07/28 09:31:42 Maybe change this to int?

Sergey Ulanov 2016/07/28 18:01:34 There are many other places where int8_t is used p

+ bool marker_bit,

+ uint32_t capture_timestamp,

+ 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.

virtual size_t RtpHeaderLength() const = 0;

@@ -153,7 +162,7 @@ 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 +176,7 @@ 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;

@@ -247,8 +256,13 @@ class RTPSender : public RTPSenderInterface {

bool marker_bit,

uint32_t capture_timestamp,

int64_t capture_time_ms,

- const bool timestamp_provided = true,

- const bool inc_sequence_number = true) override;

+ bool timestamp_provided = true,

+ bool inc_sequence_number = true) override;

+ int32_t BuildRtpHeader(uint8_t* data_buffer,

+ int8_t payload_type,

stefan-webrtc 2016/07/28 09:31:42 same here

+ bool marker_bit,

+ uint32_t capture_timestamp,

+ int64_t capture_time_ms) override;

size_t RtpHeaderLength() const override;

uint16_t AllocateSequenceNumber(uint16_t packets_to_send) override;

@@ -304,7 +318,6 @@ class RTPSender : public RTPSenderInterface {

bool timestamp_provided,

uint32_t timestamp,

int64_t capture_time_ms);

size_t SendPadData(size_t bytes,

bool timestamp_provided,

uint32_t timestamp,