| Index: webrtc/modules/pacing/packet_router.cc
|
| diff --git a/webrtc/modules/pacing/packet_router.cc b/webrtc/modules/pacing/packet_router.cc
|
| index 1884958aca48f8c6286c13a45d64fe7ec9aca8df..be45615ca5b89c0dc23dad4be9c85775cab8744e 100644
|
| --- a/webrtc/modules/pacing/packet_router.cc
|
| +++ b/webrtc/modules/pacing/packet_router.cc
|
| @@ -50,20 +50,22 @@ bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
|
| for (auto* rtp_module : rtp_modules_) {
|
| if (rtp_module->SendingMedia() && ssrc == rtp_module->SSRC()) {
|
| return rtp_module->TimeToSendPacket(ssrc, sequence_number,
|
| - capture_timestamp, retransmission);
|
| + capture_timestamp, retransmission,
|
| + probe_cluster_id);
|
| }
|
| }
|
| return true;
|
| }
|
|
|
| -size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send) {
|
| +size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send,
|
| + int probe_cluster_id) {
|
| RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
|
| size_t total_bytes_sent = 0;
|
| rtc::CritScope cs(&modules_crit_);
|
| for (RtpRtcp* module : rtp_modules_) {
|
| if (module->SendingMedia()) {
|
| - size_t bytes_sent =
|
| - module->TimeToSendPadding(bytes_to_send - total_bytes_sent);
|
| + size_t bytes_sent = module->TimeToSendPadding(
|
| + bytes_to_send - total_bytes_sent, probe_cluster_id);
|
| total_bytes_sent += bytes_sent;
|
| if (total_bytes_sent >= bytes_to_send)
|
| break;
|
|
|