зеркало из https://github.com/mozilla/gecko-dev.git
134 строки
5.0 KiB
Diff
134 строки
5.0 KiB
Diff
From: Andreas Pehrson <apehrson@mozilla.com>
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Date: Mon, 18 Jan 2021 11:04:00 +0100
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Subject: Bug 1654112 - Include RtcpPacketTypeCounter in audio send stats, to
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not regress nackCount. r=ng
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This is similar to how it's already included for video send.
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Differential Revision: https://phabricator.services.mozilla.com/D102273
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Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d380a43d59f4f7cbc001f4eab9b63ee993b32cd8
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---
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audio/audio_send_stream.cc | 1 +
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audio/channel_send.cc | 31 +++++++++++++++++++++++++++++++
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audio/channel_send.h | 1 +
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call/audio_send_stream.h | 2 ++
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4 files changed, 35 insertions(+)
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diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
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index 7d6ec794d4..20af3f7722 100644
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--- a/audio/audio_send_stream.cc
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+++ b/audio/audio_send_stream.cc
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@@ -442,6 +442,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
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stats.target_bitrate_bps = channel_send_->GetTargetBitrate();
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webrtc::CallSendStatistics call_stats = channel_send_->GetRTCPStatistics();
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+ stats.rtcp_packet_type_counts = call_stats.rtcp_packet_type_counts;
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stats.payload_bytes_sent = call_stats.payload_bytes_sent;
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stats.header_and_padding_bytes_sent =
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call_stats.header_and_padding_bytes_sent;
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diff --git a/audio/channel_send.cc b/audio/channel_send.cc
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index 0bca328846..c84a91770e 100644
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--- a/audio/channel_send.cc
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+++ b/audio/channel_send.cc
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@@ -56,6 +56,31 @@ class RtpPacketSenderProxy;
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class TransportSequenceNumberProxy;
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class VoERtcpObserver;
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+class RtcpCounterObserver : public RtcpPacketTypeCounterObserver {
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+ public:
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+ explicit RtcpCounterObserver(uint32_t ssrc) : ssrc_(ssrc) {}
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+
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+ void RtcpPacketTypesCounterUpdated(
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+ uint32_t ssrc, const RtcpPacketTypeCounter& packet_counter) override {
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+ if (ssrc_ != ssrc) {
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+ return;
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+ }
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+
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+ MutexLock lock(&mutex_);
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+ packet_counter_ = packet_counter;
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+ }
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+
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+ RtcpPacketTypeCounter GetCounts() {
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+ MutexLock lock(&mutex_);
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+ return packet_counter_;
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+ }
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+
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+ private:
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+ Mutex mutex_;
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+ const uint32_t ssrc_;
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+ RtcpPacketTypeCounter packet_counter_;
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+};
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+
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class ChannelSend : public ChannelSendInterface,
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public AudioPacketizationCallback, // receive encoded
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// packets from the ACM
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@@ -208,6 +233,8 @@ class ChannelSend : public ChannelSendInterface,
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// RtcpBandwidthObserver
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const std::unique_ptr<VoERtcpObserver> rtcp_observer_;
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+ const std::unique_ptr<RtcpCounterObserver> rtcp_counter_observer_;
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+
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PacketRouter* packet_router_ RTC_GUARDED_BY(&worker_thread_checker_) =
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nullptr;
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TransportFeedbackObserver* const feedback_observer_;
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@@ -460,6 +487,7 @@ ChannelSend::ChannelSend(
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: ssrc_(ssrc),
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event_log_(rtc_event_log),
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rtcp_observer_(new VoERtcpObserver(this)),
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+ rtcp_counter_observer_(new RtcpCounterObserver(ssrc)),
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feedback_observer_(feedback_observer),
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rtp_packet_pacer_proxy_(new RtpPacketSenderProxy()),
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retransmission_rate_limiter_(
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@@ -482,6 +510,8 @@ ChannelSend::ChannelSend(
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configuration.event_log = event_log_;
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configuration.rtt_stats = rtcp_rtt_stats;
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+ configuration.rtcp_packet_type_counter_observer =
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+ rtcp_counter_observer_.get();
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configuration.retransmission_rate_limiter =
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retransmission_rate_limiter_.get();
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configuration.extmap_allow_mixed = extmap_allow_mixed;
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@@ -759,6 +789,7 @@ CallSendStatistics ChannelSend::GetRTCPStatistics() const {
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RTC_DCHECK_RUN_ON(&worker_thread_checker_);
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CallSendStatistics stats = {0};
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stats.rttMs = GetRTT();
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+ stats.rtcp_packet_type_counts = rtcp_counter_observer_->GetCounts();
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StreamDataCounters rtp_stats;
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StreamDataCounters rtx_stats;
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diff --git a/audio/channel_send.h b/audio/channel_send.h
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index cf9a273f70..9b3969161c 100644
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--- a/audio/channel_send.h
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+++ b/audio/channel_send.h
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@@ -43,6 +43,7 @@ struct CallSendStatistics {
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TimeDelta total_packet_send_delay = TimeDelta::Zero();
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// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent
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uint64_t retransmitted_packets_sent;
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+ RtcpPacketTypeCounter rtcp_packet_type_counts;
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// A snapshot of Report Blocks with additional data of interest to statistics.
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// Within this list, the sender-source SSRC pair is unique and per-pair the
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// ReportBlockData represents the latest Report Block that was received for
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diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
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index 0f42d0fb82..bafa22d312 100644
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--- a/call/audio_send_stream.h
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+++ b/call/audio_send_stream.h
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@@ -31,6 +31,7 @@
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#include "call/rtp_config.h"
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#include "modules/audio_processing/include/audio_processing_statistics.h"
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#include "modules/rtp_rtcp/include/report_block_data.h"
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+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
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namespace webrtc {
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@@ -65,6 +66,7 @@ class AudioSendStream : public AudioSender {
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ANAStats ana_statistics;
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AudioProcessingStats apm_statistics;
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+ RtcpPacketTypeCounter rtcp_packet_type_counts;
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int64_t target_bitrate_bps = 0;
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// A snapshot of Report Blocks with additional data of interest to
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--
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2.34.1
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