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Eliminate RtpVideoStreamReceiver::receive_cs_ in favor of using a SequencedTaskChecker

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RtpVideoStreamReceiver::receive_cs_ is not really necessary, since all of the functions where that lock is acquired, are arrived at from functions of BaseChannel which DCHECK being called from BaseChannel::worker_thread_.

BUG=webrtc:8037

Review-Url: https://codereview.webrtc.org/2987933003
Cr-Commit-Position: refs/heads/master@{#19508}
This commit is contained in:
eladalon 2017-08-25 00:49:08 -07:00 committed by Commit Bot
parent 8cee56f254
commit 8b07305b04
2 changed files with 40 additions and 49 deletions
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71
webrtc/video/rtp_video_stream_receiver.cc
View File

@ -314,37 +314,33 @@ int32_t RtpVideoStreamReceiver::OnInitializeDecoder(
// This method handles both regular RTP packets and packets recovered
// via FlexFEC.
void RtpVideoStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
// TODO(eladalon): https://bugs.chromium.org/p/webrtc/issues/detail?id=8056
// RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
{
rtc::CritScope lock(&receive_cs_);
if (!receiving_) {
return;
}
if (!receiving_) {
return;
}
if (!packet.recovered()) {
int64_t now_ms = clock_->TimeInMilliseconds();
if (!packet.recovered()) {
int64_t now_ms = clock_->TimeInMilliseconds();
// Periodically log the RTP header of incoming packets.
if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
std::stringstream ss;
ss << "Packet received on SSRC: " << packet.Ssrc()
<< " with payload type: " << static_cast<int>(packet.PayloadType())
<< ", timestamp: " << packet.Timestamp()
<< ", sequence number: " << packet.SequenceNumber()
<< ", arrival time: " << packet.arrival_time_ms();
int32_t time_offset;
if (packet.GetExtension<TransmissionOffset>(&time_offset)) {
ss << ", toffset: " << time_offset;
}
uint32_t send_time;
if (packet.GetExtension<AbsoluteSendTime>(&send_time)) {
ss << ", abs send time: " << send_time;
}
LOG(LS_INFO) << ss.str();
last_packet_log_ms_ = now_ms;
// Periodically log the RTP header of incoming packets.
if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
std::stringstream ss;
ss << "Packet received on SSRC: " << packet.Ssrc()
<< " with payload type: " << static_cast<int>(packet.PayloadType())
<< ", timestamp: " << packet.Timestamp()
<< ", sequence number: " << packet.SequenceNumber()
<< ", arrival time: " << packet.arrival_time_ms();
int32_t time_offset;
if (packet.GetExtension<TransmissionOffset>(&time_offset)) {
ss << ", toffset: " << time_offset;
}
uint32_t send_time;
if (packet.GetExtension<AbsoluteSendTime>(&send_time)) {
ss << ", abs send time: " << send_time;
}
LOG(LS_INFO) << ss.str();
last_packet_log_ms_ = now_ms;
}
}
@ -442,8 +438,7 @@ rtc::Optional<int64_t> RtpVideoStreamReceiver::LastReceivedKeyframePacketMs()
}
void RtpVideoStreamReceiver::AddSecondarySink(RtpPacketSinkInterface* sink) {
// TODO(eladalon): https://bugs.chromium.org/p/webrtc/issues/detail?id=8056
// RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
RTC_DCHECK(std::find(secondary_sinks_.cbegin(), secondary_sinks_.cend(),
sink) == secondary_sinks_.cend());
secondary_sinks_.push_back(sink);
@ -451,8 +446,7 @@ void RtpVideoStreamReceiver::AddSecondarySink(RtpPacketSinkInterface* sink) {
void RtpVideoStreamReceiver::RemoveSecondarySink(
const RtpPacketSinkInterface* sink) {
// TODO(eladalon): https://bugs.chromium.org/p/webrtc/issues/detail?id=8056
// RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
auto it = std::find(secondary_sinks_.begin(), secondary_sinks_.end(), sink);
if (it == secondary_sinks_.end()) {
// We might be rolling-back a call whose setup failed mid-way. In such a
@ -486,6 +480,7 @@ void RtpVideoStreamReceiver::ReceivePacket(const uint8_t* packet,
void RtpVideoStreamReceiver::ParseAndHandleEncapsulatingHeader(
const uint8_t* packet, size_t packet_length, const RTPHeader& header) {
RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
if (rtp_payload_registry_.IsRed(header)) {
int8_t ulpfec_pt = rtp_payload_registry_.ulpfec_payload_type();
if (packet[header.headerLength] == ulpfec_pt) {
@ -510,7 +505,6 @@ void RtpVideoStreamReceiver::ParseAndHandleEncapsulatingHeader(
return;
if (packet_length > sizeof(restored_packet_))
return;
rtc::CritScope lock(&receive_cs_);
if (restored_packet_in_use_) {
LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet.";
return;
@ -568,11 +562,10 @@ void RtpVideoStreamReceiver::NotifyReceiverOfFecPacket(
bool RtpVideoStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
size_t rtcp_packet_length) {
{
rtc::CritScope lock(&receive_cs_);
if (!receiving_) {
return false;
}
RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
if (!receiving_) {
return false;
}
rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
@ -634,12 +627,12 @@ void RtpVideoStreamReceiver::SignalNetworkState(NetworkState state) {
}
void RtpVideoStreamReceiver::StartReceive() {
rtc::CritScope lock(&receive_cs_);
RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
receiving_ = true;
}
void RtpVideoStreamReceiver::StopReceive() {
rtc::CritScope lock(&receive_cs_);
RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
receiving_ = false;
}

18
webrtc/video/rtp_video_stream_receiver.h
View File

@ -32,7 +32,7 @@
#include "webrtc/modules/video_coding/sequence_number_util.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/rtc_base/thread_checker.h"
#include "webrtc/rtc_base/sequenced_task_checker.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -183,11 +183,11 @@ class RtpVideoStreamReceiver : public RtpData,
const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
std::unique_ptr<UlpfecReceiver> ulpfec_receiver_;
rtc::CriticalSection receive_cs_;
bool receiving_ GUARDED_BY(receive_cs_);
uint8_t restored_packet_[IP_PACKET_SIZE] GUARDED_BY(receive_cs_);
bool restored_packet_in_use_ GUARDED_BY(receive_cs_);
int64_t last_packet_log_ms_ GUARDED_BY(receive_cs_);
rtc::SequencedTaskChecker worker_task_checker_;
bool receiving_ GUARDED_BY(worker_task_checker_);
uint8_t restored_packet_[IP_PACKET_SIZE] GUARDED_BY(worker_task_checker_);
bool restored_packet_in_use_ GUARDED_BY(worker_task_checker_);
int64_t last_packet_log_ms_ GUARDED_BY(worker_task_checker_);
const std::unique_ptr<RtpRtcp> rtp_rtcp_;
@ -210,10 +210,8 @@ class RtpVideoStreamReceiver : public RtpData,
bool has_received_frame_;
// TODO(eladalon): https://bugs.chromium.org/p/webrtc/issues/detail?id=8056
// rtc::ThreadChecker worker_thread_checker_;
std::vector<RtpPacketSinkInterface*> secondary_sinks_; // This needs
// to be GUARDED_BY(worker_thread_checker_).
std::vector<RtpPacketSinkInterface*> secondary_sinks_
GUARDED_BY(worker_task_checker_);
};
} // namespace webrtc