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