RtpVideoSender nits
The following private methods needlessly took a reference to the
RtpConfig on which they had worked, which was itself a member.
* ConfigureProtection
* ConfigureSsrcs
* ConfigureRids
Bug: None
Change-Id: I013ca438915336d1b8f3477fe8b9f1bf87f514f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138205
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28041}
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index b73ad13..115cd1f 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -249,10 +249,10 @@
encoder_target_rate_bps_(0),
frame_counts_(rtp_config.ssrcs.size()),
frame_count_observer_(observers.frame_count_observer) {
- RTC_DCHECK_EQ(rtp_config.ssrcs.size(), rtp_streams_.size());
+ RTC_DCHECK_EQ(rtp_config_.ssrcs.size(), rtp_streams_.size());
module_process_thread_checker_.Detach();
// SSRCs are assumed to be sorted in the same order as |rtp_modules|.
- for (uint32_t ssrc : rtp_config.ssrcs) {
+ for (uint32_t ssrc : rtp_config_.ssrcs) {
// Restore state if it previously existed.
const RtpPayloadState* state = nullptr;
auto it = states.find(ssrc);
@@ -286,28 +286,28 @@
}
}
- ConfigureProtection(rtp_config);
- ConfigureSsrcs(rtp_config);
- ConfigureRids(rtp_config);
+ ConfigureProtection();
+ ConfigureSsrcs();
+ ConfigureRids();
- if (!rtp_config.mid.empty()) {
+ if (!rtp_config_.mid.empty()) {
for (const RtpStreamSender& stream : rtp_streams_) {
- stream.rtp_rtcp->SetMid(rtp_config.mid);
+ stream.rtp_rtcp->SetMid(rtp_config_.mid);
}
}
for (const RtpStreamSender& stream : rtp_streams_) {
// Simulcast has one module for each layer. Set the CNAME on all modules.
- stream.rtp_rtcp->SetCNAME(rtp_config.c_name.c_str());
+ stream.rtp_rtcp->SetCNAME(rtp_config_.c_name.c_str());
stream.rtp_rtcp->RegisterRtcpStatisticsCallback(observers.rtcp_stats);
stream.rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(
observers.rtp_stats);
- stream.rtp_rtcp->SetMaxRtpPacketSize(rtp_config.max_packet_size);
- stream.rtp_rtcp->RegisterSendPayloadFrequency(rtp_config.payload_type,
+ stream.rtp_rtcp->SetMaxRtpPacketSize(rtp_config_.max_packet_size);
+ stream.rtp_rtcp->RegisterSendPayloadFrequency(rtp_config_.payload_type,
kVideoPayloadTypeFrequency);
- stream.sender_video->RegisterPayloadType(rtp_config.payload_type,
- rtp_config.payload_name,
- rtp_config.raw_payload);
+ stream.sender_video->RegisterPayloadType(rtp_config_.payload_type,
+ rtp_config_.payload_name,
+ rtp_config_.raw_payload);
}
// Currently, both ULPFEC and FlexFEC use the same FEC rate calculation logic,
// so enable that logic if either of those FEC schemes are enabled.
@@ -464,14 +464,14 @@
}
}
-void RtpVideoSender::ConfigureProtection(const RtpConfig& rtp_config) {
+void RtpVideoSender::ConfigureProtection() {
// Consistency of FlexFEC parameters is checked in MaybeCreateFlexfecSender.
const bool flexfec_enabled = (flexfec_sender_ != nullptr);
// Consistency of NACK and RED+ULPFEC parameters is checked in this function.
- const bool nack_enabled = rtp_config.nack.rtp_history_ms > 0;
- int red_payload_type = rtp_config.ulpfec.red_payload_type;
- int ulpfec_payload_type = rtp_config.ulpfec.ulpfec_payload_type;
+ const bool nack_enabled = rtp_config_.nack.rtp_history_ms > 0;
+ int red_payload_type = rtp_config_.ulpfec.red_payload_type;
+ int ulpfec_payload_type = rtp_config_.ulpfec.ulpfec_payload_type;
// Shorthands.
auto IsRedEnabled = [&]() { return red_payload_type >= 0; };
@@ -500,7 +500,7 @@
// is a waste of bandwidth since FEC packets still have to be transmitted.
// Note that this is not the case with FlexFEC.
if (nack_enabled && IsUlpfecEnabled() &&
- !PayloadTypeSupportsSkippingFecPackets(rtp_config.payload_name)) {
+ !PayloadTypeSupportsSkippingFecPackets(rtp_config_.payload_name)) {
RTC_LOG(LS_WARNING)
<< "Transmitting payload type without picture ID using "
"NACK+ULPFEC is a waste of bandwidth since ULPFEC packets "
@@ -553,11 +553,11 @@
stream.rtp_rtcp->IncomingRtcpPacket(packet, length);
}
-void RtpVideoSender::ConfigureSsrcs(const RtpConfig& rtp_config) {
+void RtpVideoSender::ConfigureSsrcs() {
// Configure regular SSRCs.
RTC_CHECK(ssrc_to_acknowledged_packets_observers_.empty());
- for (size_t i = 0; i < rtp_config.ssrcs.size(); ++i) {
- uint32_t ssrc = rtp_config.ssrcs[i];
+ for (size_t i = 0; i < rtp_config_.ssrcs.size(); ++i) {
+ uint32_t ssrc = rtp_config_.ssrcs[i];
RtpRtcp* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.get();
rtp_rtcp->SetSSRC(ssrc);
@@ -573,13 +573,13 @@
}
// Set up RTX if available.
- if (rtp_config.rtx.ssrcs.empty())
+ if (rtp_config_.rtx.ssrcs.empty())
return;
// Configure RTX SSRCs.
- RTC_DCHECK_EQ(rtp_config.rtx.ssrcs.size(), rtp_config.ssrcs.size());
- for (size_t i = 0; i < rtp_config.rtx.ssrcs.size(); ++i) {
- uint32_t ssrc = rtp_config.rtx.ssrcs[i];
+ RTC_DCHECK_EQ(rtp_config_.rtx.ssrcs.size(), rtp_config_.ssrcs.size());
+ for (size_t i = 0; i < rtp_config_.rtx.ssrcs.size(); ++i) {
+ uint32_t ssrc = rtp_config_.rtx.ssrcs[i];
RtpRtcp* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.get();
rtp_rtcp->SetRtxSsrc(ssrc);
auto it = suspended_ssrcs_.find(ssrc);
@@ -588,30 +588,30 @@
}
// Configure RTX payload types.
- RTC_DCHECK_GE(rtp_config.rtx.payload_type, 0);
+ RTC_DCHECK_GE(rtp_config_.rtx.payload_type, 0);
for (const RtpStreamSender& stream : rtp_streams_) {
- stream.rtp_rtcp->SetRtxSendPayloadType(rtp_config.rtx.payload_type,
- rtp_config.payload_type);
+ stream.rtp_rtcp->SetRtxSendPayloadType(rtp_config_.rtx.payload_type,
+ rtp_config_.payload_type);
stream.rtp_rtcp->SetRtxSendStatus(kRtxRetransmitted |
kRtxRedundantPayloads);
}
- if (rtp_config.ulpfec.red_payload_type != -1 &&
- rtp_config.ulpfec.red_rtx_payload_type != -1) {
+ if (rtp_config_.ulpfec.red_payload_type != -1 &&
+ rtp_config_.ulpfec.red_rtx_payload_type != -1) {
for (const RtpStreamSender& stream : rtp_streams_) {
stream.rtp_rtcp->SetRtxSendPayloadType(
- rtp_config.ulpfec.red_rtx_payload_type,
- rtp_config.ulpfec.red_payload_type);
+ rtp_config_.ulpfec.red_rtx_payload_type,
+ rtp_config_.ulpfec.red_payload_type);
}
}
}
-void RtpVideoSender::ConfigureRids(const RtpConfig& rtp_config) {
- RTC_DCHECK(rtp_config.rids.empty() ||
- rtp_config.rids.size() == rtp_config.ssrcs.size());
- RTC_DCHECK(rtp_config.rids.empty() ||
- rtp_config.rids.size() == rtp_streams_.size());
- for (size_t i = 0; i < rtp_config.rids.size(); ++i) {
- const std::string& rid = rtp_config.rids[i];
+void RtpVideoSender::ConfigureRids() {
+ RTC_DCHECK(rtp_config_.rids.empty() ||
+ rtp_config_.rids.size() == rtp_config_.ssrcs.size());
+ RTC_DCHECK(rtp_config_.rids.empty() ||
+ rtp_config_.rids.size() == rtp_streams_.size());
+ for (size_t i = 0; i < rtp_config_.rids.size(); ++i) {
+ const std::string& rid = rtp_config_.rids[i];
rtp_streams_[i].rtp_rtcp->SetRid(rid);
}
}
diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h
index db8ce85..b437769 100644
--- a/call/rtp_video_sender.h
+++ b/call/rtp_video_sender.h
@@ -152,9 +152,9 @@
private:
void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
- void ConfigureProtection(const RtpConfig& rtp_config);
- void ConfigureSsrcs(const RtpConfig& rtp_config);
- void ConfigureRids(const RtpConfig& rtp_config);
+ void ConfigureProtection();
+ void ConfigureSsrcs();
+ void ConfigureRids();
bool FecEnabled() const;
bool NackEnabled() const;
uint32_t GetPacketizationOverheadRate() const;
