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From: Andreas Pehrson <apehrson@mozilla.com>
Date: Mon, 18 Jan 2021 11:04:00 +0100
Subject: Bug 1654112 - Include RtcpPacketTypeCounter in audio send stats, to
not regress nackCount. r=ng
This is similar to how it's already included for video send.
Differential Revision: https://phabricator.services.mozilla.com/D102273
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d380a43d59f4f7cbc001f4eab9b63ee993b32cd8
---
audio/audio_send_stream.cc | 1 +
audio/channel_send.cc | 32 ++++++++++++++++++++++++++++++++
audio/channel_send.h | 1 +
call/audio_send_stream.h | 2 ++
4 files changed, 36 insertions(+)
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 4710362715..a56b45d405 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -437,6 +437,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
webrtc::ChannelSendStatistics channel_stats =
channel_send_->GetRTCPStatistics();
+ stats.rtcp_packet_type_counts = channel_stats.rtcp_packet_type_counts;
stats.payload_bytes_sent = channel_stats.payload_bytes_sent;
stats.header_and_padding_bytes_sent =
channel_stats.header_and_padding_bytes_sent;
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index fbda6e852e..7536645e4e 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -81,6 +81,32 @@ constexpr TimeDelta kMinRetransmissionWindow = TimeDelta::Millis(30);
class RtpPacketSenderProxy;
class TransportSequenceNumberProxy;
+class RtcpCounterObserver : public RtcpPacketTypeCounterObserver {
+ public:
+ explicit RtcpCounterObserver(uint32_t ssrc) : ssrc_(ssrc) {}
+
+ void RtcpPacketTypesCounterUpdated(
+ uint32_t ssrc,
+ const RtcpPacketTypeCounter& packet_counter) override {
+ if (ssrc_ != ssrc) {
+ return;
+ }
+
+ MutexLock lock(&mutex_);
+ packet_counter_ = packet_counter;
+ }
+
+ RtcpPacketTypeCounter GetCounts() {
+ MutexLock lock(&mutex_);
+ return packet_counter_;
+ }
+
+ private:
+ Mutex mutex_;
+ const uint32_t ssrc_;
+ RtcpPacketTypeCounter packet_counter_;
+};
+
class AudioBitrateAccountant {
public:
void RegisterPacketOverhead(int packet_byte_overhead) {
@@ -288,6 +314,8 @@ class ChannelSend : public ChannelSendInterface,
bool input_mute_ RTC_GUARDED_BY(volume_settings_mutex_) = false;
bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_checker_) = false;
+ const std::unique_ptr<RtcpCounterObserver> rtcp_counter_observer_;
+
PacketRouter* packet_router_ RTC_GUARDED_BY(&worker_thread_checker_) =
nullptr;
const std::unique_ptr<RtpPacketSenderProxy> rtp_packet_pacer_proxy_;
@@ -500,6 +528,7 @@ ChannelSend::ChannelSend(
RtpTransportControllerSendInterface* transport_controller)
: env_(env),
ssrc_(ssrc),
+ rtcp_counter_observer_(new RtcpCounterObserver(ssrc)),
rtp_packet_pacer_proxy_(new RtpPacketSenderProxy()),
retransmission_rate_limiter_(
new RateLimiter(&env_.clock(), kMaxRetransmissionWindow.ms())),
@@ -521,6 +550,8 @@ ChannelSend::ChannelSend(
configuration.paced_sender = rtp_packet_pacer_proxy_.get();
configuration.rtt_stats = rtcp_rtt_stats;
+ configuration.rtcp_packet_type_counter_observer =
+ rtcp_counter_observer_.get();
if (env_.field_trials().IsDisabled("WebRTC-DisableRtxRateLimiter")) {
configuration.retransmission_rate_limiter =
retransmission_rate_limiter_.get();
@@ -804,6 +835,7 @@ ChannelSendStatistics ChannelSend::GetRTCPStatistics() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
ChannelSendStatistics stats = {
.round_trip_time = rtp_rtcp_->LastRtt().value_or(TimeDelta::Zero())};
+ stats.rtcp_packet_type_counts = rtcp_counter_observer_->GetCounts();
StreamDataCounters rtp_stats;
StreamDataCounters rtx_stats;
diff --git a/audio/channel_send.h b/audio/channel_send.h
index 185f3e9975..5a2c547673 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -52,6 +52,7 @@ struct ChannelSendStatistics {
TimeDelta total_packet_send_delay = TimeDelta::Zero();
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent
uint64_t retransmitted_packets_sent;
+ RtcpPacketTypeCounter rtcp_packet_type_counts;
// A snapshot of Report Blocks with additional data of interest to statistics.
// Within this list, the sender-source SSRC pair is unique and per-pair the
// ReportBlockData represents the latest Report Block that was received for
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index 2ae1742f91..21ae21b66f 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -32,6 +32,7 @@
#include "api/units/time_delta.h"
#include "call/audio_sender.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
namespace webrtc {
@@ -68,6 +69,7 @@ class AudioSendStream : public AudioSender {
ANAStats ana_statistics;
AudioProcessingStats apm_statistics;
+ RtcpPacketTypeCounter rtcp_packet_type_counts;
int64_t target_bitrate_bps = 0;
// A snapshot of Report Blocks with additional data of interest to
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