1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
|
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 | 31 +++++++++++++++++++++++++++++++
audio/channel_send.h | 1 +
call/audio_send_stream.h | 2 ++
4 files changed, 35 insertions(+)
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 59b0ea5b5e..64bc15ab4e 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -427,6 +427,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
stats.target_bitrate_bps = channel_send_->GetTargetBitrate();
webrtc::CallSendStatistics call_stats = channel_send_->GetRTCPStatistics();
+ stats.rtcp_packet_type_counts = call_stats.rtcp_packet_type_counts;
stats.payload_bytes_sent = call_stats.payload_bytes_sent;
stats.header_and_padding_bytes_sent =
call_stats.header_and_padding_bytes_sent;
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index ab1b72a2f3..db632b3aa8 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -56,6 +56,31 @@ constexpr int64_t kMinRetransmissionWindowMs = 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 ChannelSend : public ChannelSendInterface,
public AudioPacketizationCallback, // receive encoded
// packets from the ACM
@@ -210,6 +235,8 @@ 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_;
@@ -397,6 +424,7 @@ ChannelSend::ChannelSend(
const FieldTrialsView& field_trials)
: ssrc_(ssrc),
event_log_(rtc_event_log),
+ rtcp_counter_observer_(new RtcpCounterObserver(ssrc)),
rtp_packet_pacer_proxy_(new RtpPacketSenderProxy()),
retransmission_rate_limiter_(
new RateLimiter(clock, kMaxRetransmissionWindowMs)),
@@ -423,6 +451,8 @@ ChannelSend::ChannelSend(
configuration.event_log = event_log_;
configuration.rtt_stats = rtcp_rtt_stats;
+ configuration.rtcp_packet_type_counter_observer =
+ rtcp_counter_observer_.get();
if (field_trials.IsDisabled("WebRTC-DisableRtxRateLimiter")) {
configuration.retransmission_rate_limiter =
retransmission_rate_limiter_.get();
@@ -691,6 +721,7 @@ CallSendStatistics ChannelSend::GetRTCPStatistics() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
CallSendStatistics stats = {0};
stats.rttMs = GetRTT();
+ 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 b6a6a37bf5..f36085c1fa 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -43,6 +43,7 @@ struct CallSendStatistics {
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 f9e49db574..5f4f871bf0 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -31,6 +31,7 @@
#include "call/rtp_config.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
namespace webrtc {
@@ -65,6 +66,7 @@ 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
|
