| Index: webrtc/modules/pacing/packet_router.cc
|
| diff --git a/webrtc/modules/pacing/packet_router.cc b/webrtc/modules/pacing/packet_router.cc
|
| index 871ba0fcec984b99d7696a498e21b967bd2d5b8a..d5fd8b98d2f4a65d4d54fa1e1f448562e9ff74b3 100644
|
| --- a/webrtc/modules/pacing/packet_router.cc
|
| +++ b/webrtc/modules/pacing/packet_router.cc
|
| @@ -22,9 +22,7 @@ namespace webrtc {
|
| PacketRouter::PacketRouter()
|
| : last_remb_time_ms_(rtc::TimeMillis()),
|
| last_send_bitrate_bps_(0),
|
| - transport_seq_(0) {
|
| - pacer_thread_checker_.DetachFromThread();
|
| -}
|
| + transport_seq_(0) {}
|
|
|
| PacketRouter::~PacketRouter() {
|
| RTC_DCHECK(rtp_send_modules_.empty());
|
| @@ -98,7 +96,7 @@ bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
|
| int64_t capture_timestamp,
|
| bool retransmission,
|
| const PacedPacketInfo& pacing_info) {
|
| - RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
|
| + RTC_DCHECK_RUNS_SERIALIZED(&pacer_race_);
|
| rtc::CritScope cs(&modules_crit_);
|
| for (auto* rtp_module : rtp_send_modules_) {
|
| if (!rtp_module->SendingMedia())
|
| @@ -114,7 +112,7 @@ bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
|
|
|
| size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send,
|
| const PacedPacketInfo& pacing_info) {
|
| - RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
|
| + RTC_DCHECK_RUNS_SERIALIZED(&pacer_race_);
|
| size_t total_bytes_sent = 0;
|
| rtc::CritScope cs(&modules_crit_);
|
| // Rtp modules are ordered by which stream can most benefit from padding.
|
| @@ -208,7 +206,7 @@ bool PacketRouter::SendRemb(uint32_t bitrate_bps,
|
| }
|
|
|
| bool PacketRouter::SendTransportFeedback(rtcp::TransportFeedback* packet) {
|
| - RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
|
| + RTC_DCHECK_RUNS_SERIALIZED(&pacer_race_);
|
| rtc::CritScope cs(&modules_crit_);
|
| // Prefer send modules.
|
| for (auto* rtp_module : rtp_send_modules_) {
|
|
|