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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

Issue 2067673004: Style cleanups in RtpSender. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: addressed feedback Created 4 years, 6 months ago
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Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index be8ab34a2771117680367ff7e1e4653ec4c3ab01..b3590a0a7944428e761b99471bb28960ab1fd782 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -46,24 +46,7 @@ RTPExtensionType StringToRtpExtensionType(const std::string& extension) {
}
RtpRtcp::Configuration::Configuration()
- : audio(false),
- receiver_only(false),
- clock(nullptr),
- receive_statistics(NullObjectReceiveStatistics()),
- outgoing_transport(nullptr),
- intra_frame_callback(nullptr),
- bandwidth_callback(nullptr),
- transport_feedback_callback(nullptr),
- rtt_stats(nullptr),
- rtcp_packet_type_counter_observer(nullptr),
- remote_bitrate_estimator(nullptr),
- paced_sender(nullptr),
- transport_sequence_number_allocator(nullptr),
- send_bitrate_observer(nullptr),
- send_frame_count_observer(nullptr),
- send_side_delay_observer(nullptr),
- event_log(nullptr),
- send_packet_observer(nullptr) {}
+ : receive_statistics(NullObjectReceiveStatistics()) {}
RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
if (configuration.clock) {
@@ -416,7 +399,7 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData(
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation,
- const RTPVideoHeader* rtp_video_hdr) {
+ const RTPVideoHeader* rtp_video_header) {
rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
// Make sure an RTCP report isn't queued behind a key frame.
if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) {
@@ -424,7 +407,7 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData(
}
return rtp_sender_.SendOutgoingData(
frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
- payload_size, fragmentation, rtp_video_hdr);
+ payload_size, fragmentation, rtp_video_header);
}
bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,

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