Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(344)

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: fix compilation Created 4 years, 6 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
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);

Powered by Google App Engine
This is Rietveld 408576698