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webrtc_m130/pc/rtc_stats_collector.cc
Philipp Hancke 0359ba2225 stats: add frame assembly time stats 
implements a total frame assembly time statistic that measures the
cumulative time between the arrival of the first packet of a frame
(the lowest reception time) and the time all packets of the frame have
been received (i.e. the highest reception time)

This is similar to totalProcessingDelay
  https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame.

This statistic is useful for evaluating mechanisms like NACK and FEC
and gives some insight into the behavior of the pacer sending the
packets.
Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added.

Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as

totalAssemblyTime of type double
	Only exists for video. 	The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received.
    Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible.

    This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket.

framesAssembledFromMultiplePacket of type unsigned long
	Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet.
	For such frames the totalAssemblyTime is incremented.

BUG=webrtc:13986

Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36922}
2022-05-18 09:16:10 +00:00

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/*
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "pc/rtc_stats_collector.h"
#include <stdint.h>
#include <stdio.h>
#include <cstdint>
#include <map>
#include <memory>
#include <string>
#include <type_traits>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "api/candidate.h"
#include "api/dtls_transport_interface.h"
#include "api/media_stream_interface.h"
#include "api/rtp_parameters.h"
#include "api/sequence_checker.h"
#include "api/stats/rtc_stats.h"
#include "api/stats/rtcstats_objects.h"
#include "api/task_queue/queued_task.h"
#include "api/units/time_delta.h"
#include "api/video/video_content_type.h"
#include "common_video/include/quality_limitation_reason.h"
#include "media/base/media_channel.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "p2p/base/connection_info.h"
#include "p2p/base/ice_transport_internal.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/port.h"
#include "pc/channel_interface.h"
#include "pc/data_channel_utils.h"
#include "pc/rtc_stats_traversal.h"
#include "pc/rtp_receiver_proxy.h"
#include "pc/rtp_sender_proxy.h"
#include "pc/webrtc_sdp.h"
#include "rtc_base/checks.h"
#include "rtc_base/ip_address.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/network_constants.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/string_encode.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
namespace {
// TODO(https://crbug.com/webrtc/10656): Consider making IDs less predictable.
std::string RTCCertificateIDFromFingerprint(const std::string& fingerprint) {
return "RTCCertificate_" + fingerprint;
}
std::string RTCCodecStatsIDFromMidDirectionAndPayload(const std::string& mid,
bool inbound,
uint32_t payload_type) {
char buf[1024];
rtc::SimpleStringBuilder sb(buf);
sb << "RTCCodec_" << mid << (inbound ? "_Inbound_" : "_Outbound_")
<< payload_type;
return sb.str();
}
std::string RTCIceCandidatePairStatsIDFromConnectionInfo(
const cricket::ConnectionInfo& info) {
char buf[4096];
rtc::SimpleStringBuilder sb(buf);
sb << "RTCIceCandidatePair_" << info.local_candidate.id() << "_"
<< info.remote_candidate.id();
return sb.str();
}
const char kSender[] = "sender";
const char kReceiver[] = "receiver";
std::string RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
const char* direction,
int attachment_id) {
char buf[1024];
rtc::SimpleStringBuilder sb(buf);
sb << "RTCMediaStreamTrack_" << direction << "_" << attachment_id;
return sb.str();
}
std::string RTCTransportStatsIDFromTransportChannel(
const std::string& transport_name,
int channel_component) {
char buf[1024];
rtc::SimpleStringBuilder sb(buf);
sb << "RTCTransport_" << transport_name << "_" << channel_component;
return sb.str();
}
std::string RTCInboundRTPStreamStatsIDFromSSRC(cricket::MediaType media_type,
uint32_t ssrc) {
char buf[1024];
rtc::SimpleStringBuilder sb(buf);
sb << "RTCInboundRTP"
<< (media_type == cricket::MEDIA_TYPE_AUDIO ? "Audio" : "Video")
<< "Stream_" << ssrc;
return sb.str();
}
std::string RTCOutboundRTPStreamStatsIDFromSSRC(cricket::MediaType media_type,
uint32_t ssrc) {
char buf[1024];
rtc::SimpleStringBuilder sb(buf);
sb << "RTCOutboundRTP"
<< (media_type == cricket::MEDIA_TYPE_AUDIO ? "Audio" : "Video")
<< "Stream_" << ssrc;
return sb.str();
}
std::string RTCRemoteInboundRtpStreamStatsIdFromSourceSsrc(
cricket::MediaType media_type,
uint32_t source_ssrc) {
char buf[1024];
rtc::SimpleStringBuilder sb(buf);
sb << "RTCRemoteInboundRtp"
<< (media_type == cricket::MEDIA_TYPE_AUDIO ? "Audio" : "Video")
<< "Stream_" << source_ssrc;
return sb.str();
}
std::string RTCRemoteOutboundRTPStreamStatsIDFromSSRC(
cricket::MediaType media_type,
uint32_t source_ssrc) {
char buf[1024];
rtc::SimpleStringBuilder sb(buf);
sb << "RTCRemoteOutboundRTP"
<< (media_type == cricket::MEDIA_TYPE_AUDIO ? "Audio" : "Video")
<< "Stream_" << source_ssrc;
return sb.str();
}
std::string RTCMediaSourceStatsIDFromKindAndAttachment(
cricket::MediaType media_type,
int attachment_id) {
char buf[1024];
rtc::SimpleStringBuilder sb(buf);
sb << "RTC" << (media_type == cricket::MEDIA_TYPE_AUDIO ? "Audio" : "Video")
<< "Source_" << attachment_id;
return sb.str();
}
const char* CandidateTypeToRTCIceCandidateType(const std::string& type) {
if (type == cricket::LOCAL_PORT_TYPE)
return RTCIceCandidateType::kHost;
if (type == cricket::STUN_PORT_TYPE)
return RTCIceCandidateType::kSrflx;
if (type == cricket::PRFLX_PORT_TYPE)
return RTCIceCandidateType::kPrflx;
if (type == cricket::RELAY_PORT_TYPE)
return RTCIceCandidateType::kRelay;
RTC_DCHECK_NOTREACHED();
return nullptr;
}
const char* DataStateToRTCDataChannelState(
DataChannelInterface::DataState state) {
switch (state) {
case DataChannelInterface::kConnecting:
return RTCDataChannelState::kConnecting;
case DataChannelInterface::kOpen:
return RTCDataChannelState::kOpen;
case DataChannelInterface::kClosing:
return RTCDataChannelState::kClosing;
case DataChannelInterface::kClosed:
return RTCDataChannelState::kClosed;
default:
RTC_DCHECK_NOTREACHED();
return nullptr;
}
}
const char* IceCandidatePairStateToRTCStatsIceCandidatePairState(
cricket::IceCandidatePairState state) {
switch (state) {
case cricket::IceCandidatePairState::WAITING:
return RTCStatsIceCandidatePairState::kWaiting;
case cricket::IceCandidatePairState::IN_PROGRESS:
return RTCStatsIceCandidatePairState::kInProgress;
case cricket::IceCandidatePairState::SUCCEEDED:
return RTCStatsIceCandidatePairState::kSucceeded;
case cricket::IceCandidatePairState::FAILED:
return RTCStatsIceCandidatePairState::kFailed;
default:
RTC_DCHECK_NOTREACHED();
return nullptr;
}
}
const char* IceRoleToRTCIceRole(cricket::IceRole role) {
switch (role) {
case cricket::IceRole::ICEROLE_UNKNOWN:
return RTCIceRole::kUnknown;
case cricket::IceRole::ICEROLE_CONTROLLED:
return RTCIceRole::kControlled;
case cricket::IceRole::ICEROLE_CONTROLLING:
return RTCIceRole::kControlling;
default:
RTC_DCHECK_NOTREACHED();
return nullptr;
}
}
const char* DtlsTransportStateToRTCDtlsTransportState(
DtlsTransportState state) {
switch (state) {
case DtlsTransportState::kNew:
return RTCDtlsTransportState::kNew;
case DtlsTransportState::kConnecting:
return RTCDtlsTransportState::kConnecting;
case DtlsTransportState::kConnected:
return RTCDtlsTransportState::kConnected;
case DtlsTransportState::kClosed:
return RTCDtlsTransportState::kClosed;
case DtlsTransportState::kFailed:
return RTCDtlsTransportState::kFailed;
default:
RTC_CHECK_NOTREACHED();
return nullptr;
}
}
const char* IceTransportStateToRTCIceTransportState(IceTransportState state) {
switch (state) {
case IceTransportState::kNew:
return RTCIceTransportState::kNew;
case IceTransportState::kChecking:
return RTCIceTransportState::kChecking;
case IceTransportState::kConnected:
return RTCIceTransportState::kConnected;
case IceTransportState::kCompleted:
return RTCIceTransportState::kCompleted;
case IceTransportState::kFailed:
return RTCIceTransportState::kFailed;
case IceTransportState::kDisconnected:
return RTCIceTransportState::kDisconnected;
case IceTransportState::kClosed:
return RTCIceTransportState::kClosed;
default:
RTC_CHECK_NOTREACHED();
return nullptr;
}
}
const char* NetworkTypeToStatsType(rtc::AdapterType type) {
switch (type) {
case rtc::ADAPTER_TYPE_CELLULAR:
case rtc::ADAPTER_TYPE_CELLULAR_2G:
case rtc::ADAPTER_TYPE_CELLULAR_3G:
case rtc::ADAPTER_TYPE_CELLULAR_4G:
case rtc::ADAPTER_TYPE_CELLULAR_5G:
return RTCNetworkType::kCellular;
case rtc::ADAPTER_TYPE_ETHERNET:
return RTCNetworkType::kEthernet;
case rtc::ADAPTER_TYPE_WIFI:
return RTCNetworkType::kWifi;
case rtc::ADAPTER_TYPE_VPN:
return RTCNetworkType::kVpn;
case rtc::ADAPTER_TYPE_UNKNOWN:
case rtc::ADAPTER_TYPE_LOOPBACK:
case rtc::ADAPTER_TYPE_ANY:
return RTCNetworkType::kUnknown;
}
RTC_DCHECK_NOTREACHED();
return nullptr;
}
absl::string_view NetworkTypeToStatsNetworkAdapterType(rtc::AdapterType type) {
switch (type) {
case rtc::ADAPTER_TYPE_CELLULAR:
return RTCNetworkAdapterType::kCellular;
case rtc::ADAPTER_TYPE_CELLULAR_2G:
return RTCNetworkAdapterType::kCellular2g;
case rtc::ADAPTER_TYPE_CELLULAR_3G:
return RTCNetworkAdapterType::kCellular3g;
case rtc::ADAPTER_TYPE_CELLULAR_4G:
return RTCNetworkAdapterType::kCellular4g;
case rtc::ADAPTER_TYPE_CELLULAR_5G:
return RTCNetworkAdapterType::kCellular5g;
case rtc::ADAPTER_TYPE_ETHERNET:
return RTCNetworkAdapterType::kEthernet;
case rtc::ADAPTER_TYPE_WIFI:
return RTCNetworkAdapterType::kWifi;
case rtc::ADAPTER_TYPE_UNKNOWN:
return RTCNetworkAdapterType::kUnknown;
case rtc::ADAPTER_TYPE_LOOPBACK:
return RTCNetworkAdapterType::kLoopback;
case rtc::ADAPTER_TYPE_ANY:
return RTCNetworkAdapterType::kAny;
case rtc::ADAPTER_TYPE_VPN:
/* should not be handled here. Vpn is modelled as a bool */
break;
}
RTC_DCHECK_NOTREACHED();
return {};
}
const char* QualityLimitationReasonToRTCQualityLimitationReason(
QualityLimitationReason reason) {
switch (reason) {
case QualityLimitationReason::kNone:
return RTCQualityLimitationReason::kNone;
case QualityLimitationReason::kCpu:
return RTCQualityLimitationReason::kCpu;
case QualityLimitationReason::kBandwidth:
return RTCQualityLimitationReason::kBandwidth;
case QualityLimitationReason::kOther:
return RTCQualityLimitationReason::kOther;
}
RTC_CHECK_NOTREACHED();
}
std::map<std::string, double>
QualityLimitationDurationToRTCQualityLimitationDuration(
std::map<webrtc::QualityLimitationReason, int64_t> durations_ms) {
std::map<std::string, double> result;
// The internal duration is defined in milliseconds while the spec defines
// the value in seconds:
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationdurations
for (const auto& elem : durations_ms) {
result[QualityLimitationReasonToRTCQualityLimitationReason(elem.first)] =
elem.second / static_cast<double>(rtc::kNumMillisecsPerSec);
}
return result;
}
double DoubleAudioLevelFromIntAudioLevel(int audio_level) {
RTC_DCHECK_GE(audio_level, 0);
RTC_DCHECK_LE(audio_level, 32767);
return audio_level / 32767.0;
}
std::unique_ptr<RTCCodecStats> CodecStatsFromRtpCodecParameters(
uint64_t timestamp_us,
const std::string& mid,
const std::string& transport_id,
bool inbound,
const RtpCodecParameters& codec_params) {
RTC_DCHECK_GE(codec_params.payload_type, 0);
RTC_DCHECK_LE(codec_params.payload_type, 127);
RTC_DCHECK(codec_params.clock_rate);
uint32_t payload_type = static_cast<uint32_t>(codec_params.payload_type);
std::unique_ptr<RTCCodecStats> codec_stats(new RTCCodecStats(
RTCCodecStatsIDFromMidDirectionAndPayload(mid, inbound, payload_type),
timestamp_us));
codec_stats->payload_type = payload_type;
codec_stats->mime_type = codec_params.mime_type();
if (codec_params.clock_rate) {
codec_stats->clock_rate = static_cast<uint32_t>(*codec_params.clock_rate);
}
if (codec_params.num_channels) {
codec_stats->channels = *codec_params.num_channels;
}
rtc::StringBuilder fmtp;
if (WriteFmtpParameters(codec_params.parameters, &fmtp)) {
codec_stats->sdp_fmtp_line = fmtp.Release();
}
codec_stats->transport_id = transport_id;
return codec_stats;
}
void SetMediaStreamTrackStatsFromMediaStreamTrackInterface(
const MediaStreamTrackInterface& track,
RTCMediaStreamTrackStats* track_stats) {
track_stats->track_identifier = track.id();
track_stats->ended = (track.state() == MediaStreamTrackInterface::kEnded);
}
// Provides the media independent counters (both audio and video).
void SetInboundRTPStreamStatsFromMediaReceiverInfo(
const cricket::MediaReceiverInfo& media_receiver_info,
RTCInboundRTPStreamStats* inbound_stats) {
RTC_DCHECK(inbound_stats);
inbound_stats->ssrc = media_receiver_info.ssrc();
inbound_stats->packets_received =
static_cast<uint32_t>(media_receiver_info.packets_rcvd);
inbound_stats->bytes_received =
static_cast<uint64_t>(media_receiver_info.payload_bytes_rcvd);
inbound_stats->header_bytes_received =
static_cast<uint64_t>(media_receiver_info.header_and_padding_bytes_rcvd);
inbound_stats->packets_lost =
static_cast<int32_t>(media_receiver_info.packets_lost);
inbound_stats->jitter_buffer_delay =
media_receiver_info.jitter_buffer_delay_seconds;
inbound_stats->jitter_buffer_emitted_count =
media_receiver_info.jitter_buffer_emitted_count;
if (media_receiver_info.nacks_sent) {
inbound_stats->nack_count = *media_receiver_info.nacks_sent;
}
}
std::unique_ptr<RTCInboundRTPStreamStats> CreateInboundAudioStreamStats(
const cricket::VoiceReceiverInfo& voice_receiver_info,
const std::string& mid,
int64_t timestamp_us) {
auto inbound_audio = std::make_unique<RTCInboundRTPStreamStats>(
/*id=*/RTCInboundRTPStreamStatsIDFromSSRC(cricket::MEDIA_TYPE_AUDIO,
voice_receiver_info.ssrc()),
timestamp_us);
SetInboundRTPStreamStatsFromMediaReceiverInfo(voice_receiver_info,
inbound_audio.get());
inbound_audio->media_type = "audio";
inbound_audio->kind = "audio";
if (voice_receiver_info.codec_payload_type) {
inbound_audio->codec_id = RTCCodecStatsIDFromMidDirectionAndPayload(
mid, /*inbound=*/true, *voice_receiver_info.codec_payload_type);
}
inbound_audio->jitter = static_cast<double>(voice_receiver_info.jitter_ms) /
rtc::kNumMillisecsPerSec;
inbound_audio->total_samples_received =
voice_receiver_info.total_samples_received;
inbound_audio->concealed_samples = voice_receiver_info.concealed_samples;
inbound_audio->silent_concealed_samples =
voice_receiver_info.silent_concealed_samples;
inbound_audio->concealment_events = voice_receiver_info.concealment_events;
inbound_audio->inserted_samples_for_deceleration =
voice_receiver_info.inserted_samples_for_deceleration;
inbound_audio->removed_samples_for_acceleration =
voice_receiver_info.removed_samples_for_acceleration;
if (voice_receiver_info.audio_level >= 0) {
inbound_audio->audio_level =
DoubleAudioLevelFromIntAudioLevel(voice_receiver_info.audio_level);
}
inbound_audio->total_audio_energy = voice_receiver_info.total_output_energy;
inbound_audio->total_samples_duration =
voice_receiver_info.total_output_duration;
// `fir_count`, `pli_count` and `sli_count` are only valid for video and are
// purposefully left undefined for audio.
if (voice_receiver_info.last_packet_received_timestamp_ms) {
inbound_audio->last_packet_received_timestamp = static_cast<double>(
*voice_receiver_info.last_packet_received_timestamp_ms);
}
if (voice_receiver_info.estimated_playout_ntp_timestamp_ms) {
// TODO(bugs.webrtc.org/10529): Fix time origin.
inbound_audio->estimated_playout_timestamp = static_cast<double>(
*voice_receiver_info.estimated_playout_ntp_timestamp_ms);
}
inbound_audio->fec_packets_received =
voice_receiver_info.fec_packets_received;
inbound_audio->fec_packets_discarded =
voice_receiver_info.fec_packets_discarded;
inbound_audio->packets_discarded = voice_receiver_info.packets_discarded;
return inbound_audio;
}
std::unique_ptr<RTCRemoteOutboundRtpStreamStats>
CreateRemoteOutboundAudioStreamStats(
const cricket::VoiceReceiverInfo& voice_receiver_info,
const std::string& mid,
const std::string& inbound_audio_id,
const std::string& transport_id) {
if (!voice_receiver_info.last_sender_report_timestamp_ms.has_value()) {
// Cannot create `RTCRemoteOutboundRtpStreamStats` when the RTCP SR arrival
// timestamp is not available - i.e., until the first sender report is
// received.
return nullptr;
}
RTC_DCHECK_GT(voice_receiver_info.sender_reports_reports_count, 0);
// Create.
auto stats = std::make_unique<RTCRemoteOutboundRtpStreamStats>(
/*id=*/RTCRemoteOutboundRTPStreamStatsIDFromSSRC(
cricket::MEDIA_TYPE_AUDIO, voice_receiver_info.ssrc()),
/*timestamp_us=*/rtc::kNumMicrosecsPerMillisec *
voice_receiver_info.last_sender_report_timestamp_ms.value());
// Populate.
// - RTCRtpStreamStats.
stats->ssrc = voice_receiver_info.ssrc();
stats->kind = "audio";
stats->transport_id = transport_id;
stats->codec_id = RTCCodecStatsIDFromMidDirectionAndPayload(
mid,
/*inbound=*/true, // Remote-outbound same as local-inbound.
*voice_receiver_info.codec_payload_type);
// - RTCSentRtpStreamStats.
stats->packets_sent = voice_receiver_info.sender_reports_packets_sent;
stats->bytes_sent = voice_receiver_info.sender_reports_bytes_sent;
// - RTCRemoteOutboundRtpStreamStats.
stats->local_id = inbound_audio_id;
RTC_DCHECK(
voice_receiver_info.last_sender_report_remote_timestamp_ms.has_value());
stats->remote_timestamp = static_cast<double>(
voice_receiver_info.last_sender_report_remote_timestamp_ms.value());
stats->reports_sent = voice_receiver_info.sender_reports_reports_count;
if (voice_receiver_info.round_trip_time) {
stats->round_trip_time =
voice_receiver_info.round_trip_time->seconds<double>();
}
stats->round_trip_time_measurements =
voice_receiver_info.round_trip_time_measurements;
stats->total_round_trip_time =
voice_receiver_info.total_round_trip_time.seconds<double>();
return stats;
}
void SetInboundRTPStreamStatsFromVideoReceiverInfo(
const std::string& mid,
const cricket::VideoReceiverInfo& video_receiver_info,
RTCInboundRTPStreamStats* inbound_video) {
SetInboundRTPStreamStatsFromMediaReceiverInfo(video_receiver_info,
inbound_video);
inbound_video->media_type = "video";
inbound_video->kind = "video";
if (video_receiver_info.codec_payload_type) {
inbound_video->codec_id = RTCCodecStatsIDFromMidDirectionAndPayload(
mid, /*inbound=*/true, *video_receiver_info.codec_payload_type);
}
inbound_video->jitter = static_cast<double>(video_receiver_info.jitter_ms) /
rtc::kNumMillisecsPerSec;
inbound_video->fir_count =
static_cast<uint32_t>(video_receiver_info.firs_sent);
inbound_video->pli_count =
static_cast<uint32_t>(video_receiver_info.plis_sent);
inbound_video->frames_received = video_receiver_info.frames_received;
inbound_video->frames_decoded = video_receiver_info.frames_decoded;
inbound_video->frames_dropped = video_receiver_info.frames_dropped;
inbound_video->key_frames_decoded = video_receiver_info.key_frames_decoded;
if (video_receiver_info.frame_width > 0) {
inbound_video->frame_width =
static_cast<uint32_t>(video_receiver_info.frame_width);
}
if (video_receiver_info.frame_height > 0) {
inbound_video->frame_height =
static_cast<uint32_t>(video_receiver_info.frame_height);
}
if (video_receiver_info.framerate_rcvd > 0) {
inbound_video->frames_per_second = video_receiver_info.framerate_rcvd;
}
if (video_receiver_info.qp_sum)
inbound_video->qp_sum = *video_receiver_info.qp_sum;
inbound_video->total_decode_time =
static_cast<double>(video_receiver_info.total_decode_time_ms) /
rtc::kNumMillisecsPerSec;
inbound_video->total_processing_delay =
static_cast<double>(video_receiver_info.total_processing_delay.ms()) /
rtc::kNumMillisecsPerSec;
inbound_video->total_assembly_time =
static_cast<double>(video_receiver_info.total_assembly_time.ms()) /
rtc::kNumMillisecsPerSec;
inbound_video->frames_assembled_from_multiple_packets =
video_receiver_info.frames_assembled_from_multiple_packets;
inbound_video->total_inter_frame_delay =
video_receiver_info.total_inter_frame_delay;
inbound_video->total_squared_inter_frame_delay =
video_receiver_info.total_squared_inter_frame_delay;
if (video_receiver_info.last_packet_received_timestamp_ms) {
inbound_video->last_packet_received_timestamp = static_cast<double>(
*video_receiver_info.last_packet_received_timestamp_ms);
}
if (video_receiver_info.estimated_playout_ntp_timestamp_ms) {
// TODO(bugs.webrtc.org/10529): Fix time origin if needed.
inbound_video->estimated_playout_timestamp = static_cast<double>(
*video_receiver_info.estimated_playout_ntp_timestamp_ms);
}
// TODO(bugs.webrtc.org/10529): When info's `content_info` is optional
// support the "unspecified" value.
if (video_receiver_info.content_type == VideoContentType::SCREENSHARE)
inbound_video->content_type = RTCContentType::kScreenshare;
if (!video_receiver_info.decoder_implementation_name.empty()) {
inbound_video->decoder_implementation =
video_receiver_info.decoder_implementation_name;
}
}
// Provides the media independent counters (both audio and video).
void SetOutboundRTPStreamStatsFromMediaSenderInfo(
const cricket::MediaSenderInfo& media_sender_info,
RTCOutboundRTPStreamStats* outbound_stats) {
RTC_DCHECK(outbound_stats);
outbound_stats->ssrc = media_sender_info.ssrc();
outbound_stats->packets_sent =
static_cast<uint32_t>(media_sender_info.packets_sent);
outbound_stats->retransmitted_packets_sent =
media_sender_info.retransmitted_packets_sent;
outbound_stats->bytes_sent =
static_cast<uint64_t>(media_sender_info.payload_bytes_sent);
outbound_stats->header_bytes_sent =
static_cast<uint64_t>(media_sender_info.header_and_padding_bytes_sent);
outbound_stats->retransmitted_bytes_sent =
media_sender_info.retransmitted_bytes_sent;
outbound_stats->nack_count = media_sender_info.nacks_rcvd;
}
void SetOutboundRTPStreamStatsFromVoiceSenderInfo(
const std::string& mid,
const cricket::VoiceSenderInfo& voice_sender_info,
RTCOutboundRTPStreamStats* outbound_audio) {
SetOutboundRTPStreamStatsFromMediaSenderInfo(voice_sender_info,
outbound_audio);
outbound_audio->media_type = "audio";
outbound_audio->kind = "audio";
if (voice_sender_info.target_bitrate > 0) {
outbound_audio->target_bitrate = voice_sender_info.target_bitrate;
}
if (voice_sender_info.codec_payload_type) {
outbound_audio->codec_id = RTCCodecStatsIDFromMidDirectionAndPayload(
mid, /*inbound=*/false, *voice_sender_info.codec_payload_type);
}
// `fir_count`, `pli_count` and `sli_count` are only valid for video and are
// purposefully left undefined for audio.
}
void SetOutboundRTPStreamStatsFromVideoSenderInfo(
const std::string& mid,
const cricket::VideoSenderInfo& video_sender_info,
RTCOutboundRTPStreamStats* outbound_video) {
SetOutboundRTPStreamStatsFromMediaSenderInfo(video_sender_info,
outbound_video);
outbound_video->media_type = "video";
outbound_video->kind = "video";
if (video_sender_info.codec_payload_type) {
outbound_video->codec_id = RTCCodecStatsIDFromMidDirectionAndPayload(
mid, /*inbound=*/false, *video_sender_info.codec_payload_type);
}
outbound_video->fir_count =
static_cast<uint32_t>(video_sender_info.firs_rcvd);
outbound_video->pli_count =
static_cast<uint32_t>(video_sender_info.plis_rcvd);
if (video_sender_info.qp_sum)
outbound_video->qp_sum = *video_sender_info.qp_sum;
outbound_video->frames_encoded = video_sender_info.frames_encoded;
outbound_video->key_frames_encoded = video_sender_info.key_frames_encoded;
outbound_video->total_encode_time =
static_cast<double>(video_sender_info.total_encode_time_ms) /
rtc::kNumMillisecsPerSec;
outbound_video->total_encoded_bytes_target =
video_sender_info.total_encoded_bytes_target;
if (video_sender_info.send_frame_width > 0) {
outbound_video->frame_width =
static_cast<uint32_t>(video_sender_info.send_frame_width);
}
if (video_sender_info.send_frame_height > 0) {
outbound_video->frame_height =
static_cast<uint32_t>(video_sender_info.send_frame_height);
}
if (video_sender_info.framerate_sent > 0) {
outbound_video->frames_per_second = video_sender_info.framerate_sent;
}
outbound_video->frames_sent = video_sender_info.frames_sent;
outbound_video->huge_frames_sent = video_sender_info.huge_frames_sent;
outbound_video->total_packet_send_delay =
static_cast<double>(video_sender_info.total_packet_send_delay_ms) /
rtc::kNumMillisecsPerSec;
outbound_video->quality_limitation_reason =
QualityLimitationReasonToRTCQualityLimitationReason(
video_sender_info.quality_limitation_reason);
outbound_video->quality_limitation_durations =
QualityLimitationDurationToRTCQualityLimitationDuration(
video_sender_info.quality_limitation_durations_ms);
outbound_video->quality_limitation_resolution_changes =
video_sender_info.quality_limitation_resolution_changes;
// TODO(https://crbug.com/webrtc/10529): When info's `content_info` is
// optional, support the "unspecified" value.
if (video_sender_info.content_type == VideoContentType::SCREENSHARE)
outbound_video->content_type = RTCContentType::kScreenshare;
if (!video_sender_info.encoder_implementation_name.empty()) {
outbound_video->encoder_implementation =
video_sender_info.encoder_implementation_name;
}
if (video_sender_info.rid) {
outbound_video->rid = *video_sender_info.rid;
}
}
std::unique_ptr<RTCRemoteInboundRtpStreamStats>
ProduceRemoteInboundRtpStreamStatsFromReportBlockData(
const ReportBlockData& report_block_data,
cricket::MediaType media_type,
const std::map<std::string, RTCOutboundRTPStreamStats*>& outbound_rtps,
const RTCStatsReport& report) {
const auto& report_block = report_block_data.report_block();
// RTCStats' timestamp generally refers to when the metric was sampled, but
// for "remote-[outbound/inbound]-rtp" it refers to the local time when the
// Report Block was received.
auto remote_inbound = std::make_unique<RTCRemoteInboundRtpStreamStats>(
RTCRemoteInboundRtpStreamStatsIdFromSourceSsrc(media_type,
report_block.source_ssrc),
/*timestamp=*/report_block_data.report_block_timestamp_utc_us());
remote_inbound->ssrc = report_block.source_ssrc;
remote_inbound->kind =
media_type == cricket::MEDIA_TYPE_AUDIO ? "audio" : "video";
remote_inbound->packets_lost = report_block.packets_lost;
remote_inbound->fraction_lost =
static_cast<double>(report_block.fraction_lost) / (1 << 8);
remote_inbound->round_trip_time =
static_cast<double>(report_block_data.last_rtt_ms()) /
rtc::kNumMillisecsPerSec;
remote_inbound->total_round_trip_time =
static_cast<double>(report_block_data.sum_rtt_ms()) /
rtc::kNumMillisecsPerSec;
remote_inbound->round_trip_time_measurements =
report_block_data.num_rtts();
std::string local_id =
RTCOutboundRTPStreamStatsIDFromSSRC(media_type, report_block.source_ssrc);
// Look up local stat from `outbound_rtps` where the pointers are non-const.
auto local_id_it = outbound_rtps.find(local_id);
if (local_id_it != outbound_rtps.end()) {
remote_inbound->local_id = local_id;
auto& outbound_rtp = *local_id_it->second;
outbound_rtp.remote_id = remote_inbound->id();
// The RTP/RTCP transport is obtained from the
// RTCOutboundRtpStreamStats's transport.
const auto* transport_from_id = outbound_rtp.transport_id.is_defined()
? report.Get(*outbound_rtp.transport_id)
: nullptr;
if (transport_from_id) {
const auto& transport = transport_from_id->cast_to<RTCTransportStats>();
// If RTP and RTCP are not multiplexed, there is a separate RTCP
// transport paired with the RTP transport, otherwise the same
// transport is used for RTCP and RTP.
remote_inbound->transport_id =
transport.rtcp_transport_stats_id.is_defined()
? *transport.rtcp_transport_stats_id
: *outbound_rtp.transport_id;
}
// We're assuming the same codec is used on both ends. However if the
// codec is switched out on the fly we may have received a Report Block
// based on the previous codec and there is no way to tell which point in
// time the codec changed for the remote end.
const auto* codec_from_id = outbound_rtp.codec_id.is_defined()
? report.Get(*outbound_rtp.codec_id)
: nullptr;
if (codec_from_id) {
remote_inbound->codec_id = *outbound_rtp.codec_id;
const auto& codec = codec_from_id->cast_to<RTCCodecStats>();
if (codec.clock_rate.is_defined()) {
// The Report Block jitter is expressed in RTP timestamp units
// (https://tools.ietf.org/html/rfc3550#section-6.4.1). To convert this
// to seconds we divide by the codec's clock rate.
remote_inbound->jitter =
static_cast<double>(report_block.jitter) / *codec.clock_rate;
}
}
}
return remote_inbound;
}
void ProduceCertificateStatsFromSSLCertificateStats(
int64_t timestamp_us,
const rtc::SSLCertificateStats& certificate_stats,
RTCStatsReport* report) {
RTCCertificateStats* prev_certificate_stats = nullptr;
for (const rtc::SSLCertificateStats* s = &certificate_stats; s;
s = s->issuer.get()) {
std::string certificate_stats_id =
RTCCertificateIDFromFingerprint(s->fingerprint);
// It is possible for the same certificate to show up multiple times, e.g.
// if local and remote side use the same certificate in a loopback call.
// If the report already contains stats for this certificate, skip it.
if (report->Get(certificate_stats_id)) {
RTC_DCHECK_EQ(s, &certificate_stats);
break;
}
RTCCertificateStats* certificate_stats =
new RTCCertificateStats(certificate_stats_id, timestamp_us);
certificate_stats->fingerprint = s->fingerprint;
certificate_stats->fingerprint_algorithm = s->fingerprint_algorithm;
certificate_stats->base64_certificate = s->base64_certificate;
if (prev_certificate_stats)
prev_certificate_stats->issuer_certificate_id = certificate_stats->id();
report->AddStats(std::unique_ptr<RTCCertificateStats>(certificate_stats));
prev_certificate_stats = certificate_stats;
}
}
const std::string& ProduceIceCandidateStats(int64_t timestamp_us,
const cricket::Candidate& candidate,
bool is_local,
const std::string& transport_id,
RTCStatsReport* report) {
const std::string& id = "RTCIceCandidate_" + candidate.id();
const RTCStats* stats = report->Get(id);
if (!stats) {
std::unique_ptr<RTCIceCandidateStats> candidate_stats;
if (is_local)
candidate_stats.reset(new RTCLocalIceCandidateStats(id, timestamp_us));
else
candidate_stats.reset(new RTCRemoteIceCandidateStats(id, timestamp_us));
candidate_stats->transport_id = transport_id;
if (is_local) {
candidate_stats->network_type =
NetworkTypeToStatsType(candidate.network_type());
const std::string& candidate_type = candidate.type();
const std::string& relay_protocol = candidate.relay_protocol();
const std::string& url = candidate.url();
if (candidate_type == cricket::RELAY_PORT_TYPE ||
(candidate_type == cricket::PRFLX_PORT_TYPE &&
!relay_protocol.empty())) {
RTC_DCHECK(relay_protocol.compare("udp") == 0 ||
relay_protocol.compare("tcp") == 0 ||
relay_protocol.compare("tls") == 0);
candidate_stats->relay_protocol = relay_protocol;
if (!url.empty()) {
candidate_stats->url = url;
}
} else if (candidate_type == cricket::STUN_PORT_TYPE) {
if (!url.empty()) {
candidate_stats->url = url;
}
}
if (candidate.network_type() == rtc::ADAPTER_TYPE_VPN) {
candidate_stats->vpn = true;
candidate_stats->network_adapter_type =
std::string(NetworkTypeToStatsNetworkAdapterType(
candidate.underlying_type_for_vpn()));
} else {
candidate_stats->vpn = false;
candidate_stats->network_adapter_type = std::string(
NetworkTypeToStatsNetworkAdapterType(candidate.network_type()));
}
} else {
// We don't expect to know the adapter type of remote candidates.
RTC_DCHECK_EQ(rtc::ADAPTER_TYPE_UNKNOWN, candidate.network_type());
}
candidate_stats->ip = candidate.address().ipaddr().ToString();
candidate_stats->address = candidate.address().ipaddr().ToString();
candidate_stats->port = static_cast<int32_t>(candidate.address().port());
candidate_stats->protocol = candidate.protocol();
candidate_stats->candidate_type =
CandidateTypeToRTCIceCandidateType(candidate.type());
candidate_stats->priority = static_cast<int32_t>(candidate.priority());
stats = candidate_stats.get();
report->AddStats(std::move(candidate_stats));
}
RTC_DCHECK_EQ(stats->type(), is_local ? RTCLocalIceCandidateStats::kType
: RTCRemoteIceCandidateStats::kType);
return stats->id();
}
template <typename StatsType>
void SetAudioProcessingStats(StatsType* stats,
const AudioProcessingStats& apm_stats) {
if (apm_stats.echo_return_loss) {
stats->echo_return_loss = *apm_stats.echo_return_loss;
}
if (apm_stats.echo_return_loss_enhancement) {
stats->echo_return_loss_enhancement =
*apm_stats.echo_return_loss_enhancement;
}
}
std::unique_ptr<RTCMediaStreamTrackStats>
ProduceMediaStreamTrackStatsFromVoiceSenderInfo(
int64_t timestamp_us,
AudioTrackInterface& audio_track,
const cricket::VoiceSenderInfo& voice_sender_info,
int attachment_id) {
std::unique_ptr<RTCMediaStreamTrackStats> audio_track_stats(
new RTCMediaStreamTrackStats(
RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(kSender,
attachment_id),
timestamp_us, RTCMediaStreamTrackKind::kAudio));
SetMediaStreamTrackStatsFromMediaStreamTrackInterface(
audio_track, audio_track_stats.get());
audio_track_stats->media_source_id =
RTCMediaSourceStatsIDFromKindAndAttachment(cricket::MEDIA_TYPE_AUDIO,
attachment_id);
audio_track_stats->remote_source = false;
audio_track_stats->detached = false;
// Audio processor may be attached to either the track or the send
// stream, so look in both places.
SetAudioProcessingStats(audio_track_stats.get(),
voice_sender_info.apm_statistics);
auto audio_processor(audio_track.GetAudioProcessor());
if (audio_processor.get()) {
// The `has_remote_tracks` argument is obsolete; makes no difference if it's
// set to true or false.
AudioProcessorInterface::AudioProcessorStatistics ap_stats =
audio_processor->GetStats(/*has_remote_tracks=*/false);
SetAudioProcessingStats(audio_track_stats.get(), ap_stats.apm_statistics);
}
return audio_track_stats;
}
std::unique_ptr<RTCMediaStreamTrackStats>
ProduceMediaStreamTrackStatsFromVoiceReceiverInfo(
int64_t timestamp_us,
const AudioTrackInterface& audio_track,
const cricket::VoiceReceiverInfo& voice_receiver_info,
int attachment_id) {
// Since receiver tracks can't be reattached, we use the SSRC as
// an attachment identifier.
std::unique_ptr<RTCMediaStreamTrackStats> audio_track_stats(
new RTCMediaStreamTrackStats(
RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(kReceiver,
attachment_id),
timestamp_us, RTCMediaStreamTrackKind::kAudio));
SetMediaStreamTrackStatsFromMediaStreamTrackInterface(
audio_track, audio_track_stats.get());
audio_track_stats->remote_source = true;
audio_track_stats->detached = false;
if (voice_receiver_info.audio_level >= 0) {
audio_track_stats->audio_level =
DoubleAudioLevelFromIntAudioLevel(voice_receiver_info.audio_level);
}
audio_track_stats->jitter_buffer_delay =
voice_receiver_info.jitter_buffer_delay_seconds;
audio_track_stats->jitter_buffer_emitted_count =
voice_receiver_info.jitter_buffer_emitted_count;
audio_track_stats->inserted_samples_for_deceleration =
voice_receiver_info.inserted_samples_for_deceleration;
audio_track_stats->removed_samples_for_acceleration =
voice_receiver_info.removed_samples_for_acceleration;
audio_track_stats->total_audio_energy =
voice_receiver_info.total_output_energy;
audio_track_stats->total_samples_received =
voice_receiver_info.total_samples_received;
audio_track_stats->total_samples_duration =
voice_receiver_info.total_output_duration;
audio_track_stats->concealed_samples = voice_receiver_info.concealed_samples;
audio_track_stats->silent_concealed_samples =
voice_receiver_info.silent_concealed_samples;
audio_track_stats->concealment_events =
voice_receiver_info.concealment_events;
audio_track_stats->jitter_buffer_flushes =
voice_receiver_info.jitter_buffer_flushes;
audio_track_stats->delayed_packet_outage_samples =
voice_receiver_info.delayed_packet_outage_samples;
audio_track_stats->relative_packet_arrival_delay =
voice_receiver_info.relative_packet_arrival_delay_seconds;
audio_track_stats->jitter_buffer_target_delay =
voice_receiver_info.jitter_buffer_target_delay_seconds;
audio_track_stats->interruption_count =
voice_receiver_info.interruption_count >= 0
? voice_receiver_info.interruption_count
: 0;
audio_track_stats->total_interruption_duration =
static_cast<double>(voice_receiver_info.total_interruption_duration_ms) /
rtc::kNumMillisecsPerSec;
return audio_track_stats;
}
std::unique_ptr<RTCMediaStreamTrackStats>
ProduceMediaStreamTrackStatsFromVideoSenderInfo(
int64_t timestamp_us,
const VideoTrackInterface& video_track,
const cricket::VideoSenderInfo& video_sender_info,
int attachment_id) {
std::unique_ptr<RTCMediaStreamTrackStats> video_track_stats(
new RTCMediaStreamTrackStats(
RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(kSender,
attachment_id),
timestamp_us, RTCMediaStreamTrackKind::kVideo));
SetMediaStreamTrackStatsFromMediaStreamTrackInterface(
video_track, video_track_stats.get());
video_track_stats->media_source_id =
RTCMediaSourceStatsIDFromKindAndAttachment(cricket::MEDIA_TYPE_VIDEO,
attachment_id);
video_track_stats->remote_source = false;
video_track_stats->detached = false;
video_track_stats->frame_width =
static_cast<uint32_t>(video_sender_info.send_frame_width);
video_track_stats->frame_height =
static_cast<uint32_t>(video_sender_info.send_frame_height);
// TODO(hbos): Will reduce this by frames dropped due to congestion control
// when available. https://crbug.com/659137
video_track_stats->frames_sent = video_sender_info.frames_encoded;
video_track_stats->huge_frames_sent = video_sender_info.huge_frames_sent;
return video_track_stats;
}
std::unique_ptr<RTCMediaStreamTrackStats>
ProduceMediaStreamTrackStatsFromVideoReceiverInfo(
int64_t timestamp_us,
const VideoTrackInterface& video_track,
const cricket::VideoReceiverInfo& video_receiver_info,
int attachment_id) {
std::unique_ptr<RTCMediaStreamTrackStats> video_track_stats(
new RTCMediaStreamTrackStats(
RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(kReceiver,
attachment_id),
timestamp_us, RTCMediaStreamTrackKind::kVideo));
SetMediaStreamTrackStatsFromMediaStreamTrackInterface(
video_track, video_track_stats.get());
video_track_stats->remote_source = true;
video_track_stats->detached = false;
if (video_receiver_info.frame_width > 0 &&
video_receiver_info.frame_height > 0) {
video_track_stats->frame_width =
static_cast<uint32_t>(video_receiver_info.frame_width);
video_track_stats->frame_height =
static_cast<uint32_t>(video_receiver_info.frame_height);
}
video_track_stats->jitter_buffer_delay =
video_receiver_info.jitter_buffer_delay_seconds;
video_track_stats->jitter_buffer_emitted_count =
video_receiver_info.jitter_buffer_emitted_count;
video_track_stats->frames_received = video_receiver_info.frames_received;
// TODO(hbos): When we support receiving simulcast, this should be the total
// number of frames correctly decoded, independent of which SSRC it was
// received from. Since we don't support that, this is correct and is the same
// value as "RTCInboundRTPStreamStats.framesDecoded". https://crbug.com/659137
video_track_stats->frames_decoded = video_receiver_info.frames_decoded;
video_track_stats->frames_dropped = video_receiver_info.frames_dropped;
video_track_stats->freeze_count = video_receiver_info.freeze_count;
video_track_stats->pause_count = video_receiver_info.pause_count;
video_track_stats->total_freezes_duration =
static_cast<double>(video_receiver_info.total_freezes_duration_ms) /
rtc::kNumMillisecsPerSec;
video_track_stats->total_pauses_duration =
static_cast<double>(video_receiver_info.total_pauses_duration_ms) /
rtc::kNumMillisecsPerSec;
video_track_stats->total_frames_duration =
static_cast<double>(video_receiver_info.total_frames_duration_ms) /
rtc::kNumMillisecsPerSec;
video_track_stats->sum_squared_frame_durations =
video_receiver_info.sum_squared_frame_durations;
return video_track_stats;
}
void ProduceSenderMediaTrackStats(
int64_t timestamp_us,
const TrackMediaInfoMap& track_media_info_map,
std::vector<rtc::scoped_refptr<RtpSenderInternal>> senders,
RTCStatsReport* report) {
// This function iterates over the senders to generate outgoing track stats.
// TODO(hbos): Return stats of detached tracks. We have to perform stats
// gathering at the time of detachment to get accurate stats and timestamps.
// https://crbug.com/659137
for (const auto& sender : senders) {
if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
AudioTrackInterface* track =
static_cast<AudioTrackInterface*>(sender->track().get());
if (!track)
continue;
cricket::VoiceSenderInfo null_sender_info;
const cricket::VoiceSenderInfo* voice_sender_info = &null_sender_info;
// TODO(hta): Checking on ssrc is not proper. There should be a way
// to see from a sender whether it's connected or not.
// Related to https://crbug.com/8694 (using ssrc 0 to indicate "none")
if (sender->ssrc() != 0) {
// When pc.close is called, sender info is discarded, so
// we generate zeroes instead. Bug: It should be retained.
// https://crbug.com/807174
const cricket::VoiceSenderInfo* sender_info =
track_media_info_map.GetVoiceSenderInfoBySsrc(sender->ssrc());
if (sender_info) {
voice_sender_info = sender_info;
} else {
RTC_DLOG(LS_INFO)
<< "RTCStatsCollector: No voice sender info for sender with ssrc "
<< sender->ssrc();
}
}
std::unique_ptr<RTCMediaStreamTrackStats> audio_track_stats =
ProduceMediaStreamTrackStatsFromVoiceSenderInfo(
timestamp_us, *track, *voice_sender_info, sender->AttachmentId());
report->AddStats(std::move(audio_track_stats));
} else if (sender->media_type() == cricket::MEDIA_TYPE_VIDEO) {
VideoTrackInterface* track =
static_cast<VideoTrackInterface*>(sender->track().get());
if (!track)
continue;
cricket::VideoSenderInfo null_sender_info;
const cricket::VideoSenderInfo* video_sender_info = &null_sender_info;
// TODO(hta): Check on state not ssrc when state is available
// Related to https://bugs.webrtc.org/8694 (using ssrc 0 to indicate
// "none")
if (sender->ssrc() != 0) {
// When pc.close is called, sender info is discarded, so
// we generate zeroes instead. Bug: It should be retained.
// https://crbug.com/807174
const cricket::VideoSenderInfo* sender_info =
track_media_info_map.GetVideoSenderInfoBySsrc(sender->ssrc());
if (sender_info) {
video_sender_info = sender_info;
} else {
RTC_DLOG(LS_INFO)
<< "No video sender info for sender with ssrc " << sender->ssrc();
}
}
std::unique_ptr<RTCMediaStreamTrackStats> video_track_stats =
ProduceMediaStreamTrackStatsFromVideoSenderInfo(
timestamp_us, *track, *video_sender_info, sender->AttachmentId());
report->AddStats(std::move(video_track_stats));
} else {
RTC_DCHECK_NOTREACHED();
}
}
}
void ProduceReceiverMediaTrackStats(
int64_t timestamp_us,
const TrackMediaInfoMap& track_media_info_map,
std::vector<rtc::scoped_refptr<RtpReceiverInternal>> receivers,
RTCStatsReport* report) {
// This function iterates over the receivers to find the remote tracks.
for (const auto& receiver : receivers) {
if (receiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
AudioTrackInterface* track =
static_cast<AudioTrackInterface*>(receiver->track().get());
const cricket::VoiceReceiverInfo* voice_receiver_info =
track_media_info_map.GetVoiceReceiverInfo(*track);
if (!voice_receiver_info) {
continue;
}
std::unique_ptr<RTCMediaStreamTrackStats> audio_track_stats =
ProduceMediaStreamTrackStatsFromVoiceReceiverInfo(
timestamp_us, *track, *voice_receiver_info,
receiver->AttachmentId());
report->AddStats(std::move(audio_track_stats));
} else if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
VideoTrackInterface* track =
static_cast<VideoTrackInterface*>(receiver->track().get());
const cricket::VideoReceiverInfo* video_receiver_info =
track_media_info_map.GetVideoReceiverInfo(*track);
if (!video_receiver_info) {
continue;
}
std::unique_ptr<RTCMediaStreamTrackStats> video_track_stats =
ProduceMediaStreamTrackStatsFromVideoReceiverInfo(
timestamp_us, *track, *video_receiver_info,
receiver->AttachmentId());
report->AddStats(std::move(video_track_stats));
} else {
RTC_DCHECK_NOTREACHED();
}
}
}
rtc::scoped_refptr<RTCStatsReport> CreateReportFilteredBySelector(
bool filter_by_sender_selector,
rtc::scoped_refptr<const RTCStatsReport> report,
rtc::scoped_refptr<RtpSenderInternal> sender_selector,
rtc::scoped_refptr<RtpReceiverInternal> receiver_selector) {
std::vector<std::string> rtpstream_ids;
if (filter_by_sender_selector) {
// Filter mode: RTCStatsCollector::RequestInfo::kSenderSelector
if (sender_selector) {
// Find outbound-rtp(s) of the sender, i.e. the outbound-rtp(s) that
// reference the sender stats.
// Because we do not implement sender stats, we look at outbound-rtp(s)
// that reference the track attachment stats for the sender instead.
std::string track_id =
RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
kSender, sender_selector->AttachmentId());
for (const auto& stats : *report) {
if (stats.type() != RTCOutboundRTPStreamStats::kType)
continue;
const auto& outbound_rtp = stats.cast_to<RTCOutboundRTPStreamStats>();
if (outbound_rtp.track_id.is_defined() &&
*outbound_rtp.track_id == track_id) {
rtpstream_ids.push_back(outbound_rtp.id());
}
}
}
} else {
// Filter mode: RTCStatsCollector::RequestInfo::kReceiverSelector
if (receiver_selector) {
// Find inbound-rtp(s) of the receiver, i.e. the inbound-rtp(s) that
// reference the receiver stats.
// Because we do not implement receiver stats, we look at inbound-rtp(s)
// that reference the track attachment stats for the receiver instead.
std::string track_id =
RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
kReceiver, receiver_selector->AttachmentId());
for (const auto& stats : *report) {
if (stats.type() != RTCInboundRTPStreamStats::kType)
continue;
const auto& inbound_rtp = stats.cast_to<RTCInboundRTPStreamStats>();
if (inbound_rtp.track_id.is_defined() &&
*inbound_rtp.track_id == track_id) {
rtpstream_ids.push_back(inbound_rtp.id());
}
}
}
}
if (rtpstream_ids.empty())
return RTCStatsReport::Create(report->timestamp_us());
return TakeReferencedStats(report->Copy(), rtpstream_ids);
}
} // namespace
RTCStatsCollector::RequestInfo::RequestInfo(
rtc::scoped_refptr<RTCStatsCollectorCallback> callback)
: RequestInfo(FilterMode::kAll, std::move(callback), nullptr, nullptr) {}
RTCStatsCollector::RequestInfo::RequestInfo(
rtc::scoped_refptr<RtpSenderInternal> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback)
: RequestInfo(FilterMode::kSenderSelector,
std::move(callback),
std::move(selector),
nullptr) {}
RTCStatsCollector::RequestInfo::RequestInfo(
rtc::scoped_refptr<RtpReceiverInternal> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback)
: RequestInfo(FilterMode::kReceiverSelector,
std::move(callback),
nullptr,
std::move(selector)) {}
RTCStatsCollector::RequestInfo::RequestInfo(
RTCStatsCollector::RequestInfo::FilterMode filter_mode,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback,
rtc::scoped_refptr<RtpSenderInternal> sender_selector,
rtc::scoped_refptr<RtpReceiverInternal> receiver_selector)
: filter_mode_(filter_mode),
callback_(std::move(callback)),
sender_selector_(std::move(sender_selector)),
receiver_selector_(std::move(receiver_selector)) {
RTC_DCHECK(callback_);
RTC_DCHECK(!sender_selector_ || !receiver_selector_);
}
rtc::scoped_refptr<RTCStatsCollector> RTCStatsCollector::Create(
PeerConnectionInternal* pc,
int64_t cache_lifetime_us) {
return rtc::make_ref_counted<RTCStatsCollector>(pc, cache_lifetime_us);
}
RTCStatsCollector::RTCStatsCollector(PeerConnectionInternal* pc,
int64_t cache_lifetime_us)
: pc_(pc),
signaling_thread_(pc->signaling_thread()),
worker_thread_(pc->worker_thread()),
network_thread_(pc->network_thread()),
num_pending_partial_reports_(0),
partial_report_timestamp_us_(0),
network_report_event_(true /* manual_reset */,
true /* initially_signaled */),
cache_timestamp_us_(0),
cache_lifetime_us_(cache_lifetime_us) {
RTC_DCHECK(pc_);
RTC_DCHECK(signaling_thread_);
RTC_DCHECK(worker_thread_);
RTC_DCHECK(network_thread_);
RTC_DCHECK_GE(cache_lifetime_us_, 0);
pc_->SignalSctpDataChannelCreated().connect(
this, &RTCStatsCollector::OnSctpDataChannelCreated);
}
RTCStatsCollector::~RTCStatsCollector() {
RTC_DCHECK_EQ(num_pending_partial_reports_, 0);
}
void RTCStatsCollector::GetStatsReport(
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
GetStatsReportInternal(RequestInfo(std::move(callback)));
}
void RTCStatsCollector::GetStatsReport(
rtc::scoped_refptr<RtpSenderInternal> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
GetStatsReportInternal(RequestInfo(std::move(selector), std::move(callback)));
}
void RTCStatsCollector::GetStatsReport(
rtc::scoped_refptr<RtpReceiverInternal> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
GetStatsReportInternal(RequestInfo(std::move(selector), std::move(callback)));
}
void RTCStatsCollector::GetStatsReportInternal(
RTCStatsCollector::RequestInfo request) {
RTC_DCHECK_RUN_ON(signaling_thread_);
requests_.push_back(std::move(request));
// "Now" using a monotonically increasing timer.
int64_t cache_now_us = rtc::TimeMicros();
if (cached_report_ &&
cache_now_us - cache_timestamp_us_ <= cache_lifetime_us_) {
// We have a fresh cached report to deliver. Deliver asynchronously, since
// the caller may not be expecting a synchronous callback, and it avoids
// reentrancy problems.
std::vector<RequestInfo> requests;
requests.swap(requests_);
// Task subclass to take ownership of the requests.
// TODO(nisse): Delete when we can use C++14, and do lambda capture with
// std::move.
class DeliveryTask : public QueuedTask {
public:
DeliveryTask(rtc::scoped_refptr<RTCStatsCollector> collector,
rtc::scoped_refptr<const RTCStatsReport> cached_report,
std::vector<RequestInfo> requests)
: collector_(collector),
cached_report_(cached_report),
requests_(std::move(requests)) {}
bool Run() override {
collector_->DeliverCachedReport(cached_report_, std::move(requests_));
return true;
}
private:
rtc::scoped_refptr<RTCStatsCollector> collector_;
rtc::scoped_refptr<const RTCStatsReport> cached_report_;
std::vector<RequestInfo> requests_;
};
signaling_thread_->PostTask(std::make_unique<DeliveryTask>(
rtc::scoped_refptr<RTCStatsCollector>(this), cached_report_,
std::move(requests)));
} else if (!num_pending_partial_reports_) {
// Only start gathering stats if we're not already gathering stats. In the
// case of already gathering stats, `callback_` will be invoked when there
// are no more pending partial reports.
// "Now" using a system clock, relative to the UNIX epoch (Jan 1, 1970,
// UTC), in microseconds. The system clock could be modified and is not
// necessarily monotonically increasing.
int64_t timestamp_us = rtc::TimeUTCMicros();
num_pending_partial_reports_ = 2;
partial_report_timestamp_us_ = cache_now_us;
// Prepare `transceiver_stats_infos_` and `call_stats_` for use in
// `ProducePartialResultsOnNetworkThread` and
// `ProducePartialResultsOnSignalingThread`.
PrepareTransceiverStatsInfosAndCallStats_s_w_n();
// Don't touch `network_report_` on the signaling thread until
// ProducePartialResultsOnNetworkThread() has signaled the
// `network_report_event_`.
network_report_event_.Reset();
rtc::scoped_refptr<RTCStatsCollector> collector(this);
network_thread_->PostTask(
[collector, sctp_transport_name = pc_->sctp_transport_name(),
timestamp_us]() mutable {
collector->ProducePartialResultsOnNetworkThread(
timestamp_us, std::move(sctp_transport_name));
});
ProducePartialResultsOnSignalingThread(timestamp_us);
}
}
void RTCStatsCollector::ClearCachedStatsReport() {
RTC_DCHECK_RUN_ON(signaling_thread_);
cached_report_ = nullptr;
}
void RTCStatsCollector::WaitForPendingRequest() {
RTC_DCHECK_RUN_ON(signaling_thread_);
// If a request is pending, blocks until the `network_report_event_` is
// signaled and then delivers the result. Otherwise this is a NO-OP.
MergeNetworkReport_s();
}
void RTCStatsCollector::ProducePartialResultsOnSignalingThread(
int64_t timestamp_us) {
RTC_DCHECK_RUN_ON(signaling_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
partial_report_ = RTCStatsReport::Create(timestamp_us);
ProducePartialResultsOnSignalingThreadImpl(timestamp_us,
partial_report_.get());
// ProducePartialResultsOnSignalingThread() is running synchronously on the
// signaling thread, so it is always the first partial result delivered on the
// signaling thread. The request is not complete until MergeNetworkReport_s()
// happens; we don't have to do anything here.
RTC_DCHECK_GT(num_pending_partial_reports_, 1);
--num_pending_partial_reports_;
}
void RTCStatsCollector::ProducePartialResultsOnSignalingThreadImpl(
int64_t timestamp_us,
RTCStatsReport* partial_report) {
RTC_DCHECK_RUN_ON(signaling_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
ProduceDataChannelStats_s(timestamp_us, partial_report);
ProduceMediaStreamStats_s(timestamp_us, partial_report);
ProduceMediaStreamTrackStats_s(timestamp_us, partial_report);
ProduceMediaSourceStats_s(timestamp_us, partial_report);
ProducePeerConnectionStats_s(timestamp_us, partial_report);
}
void RTCStatsCollector::ProducePartialResultsOnNetworkThread(
int64_t timestamp_us,
absl::optional<std::string> sctp_transport_name) {
TRACE_EVENT0("webrtc",
"RTCStatsCollector::ProducePartialResultsOnNetworkThread");
RTC_DCHECK_RUN_ON(network_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
// Touching `network_report_` on this thread is safe by this method because
// `network_report_event_` is reset before this method is invoked.
network_report_ = RTCStatsReport::Create(timestamp_us);
std::set<std::string> transport_names;
if (sctp_transport_name) {
transport_names.emplace(std::move(*sctp_transport_name));
}
for (const auto& info : transceiver_stats_infos_) {
if (info.transport_name)
transport_names.insert(*info.transport_name);
}
std::map<std::string, cricket::TransportStats> transport_stats_by_name =
pc_->GetTransportStatsByNames(transport_names);
std::map<std::string, CertificateStatsPair> transport_cert_stats =
PrepareTransportCertificateStats_n(transport_stats_by_name);
ProducePartialResultsOnNetworkThreadImpl(
timestamp_us, transport_stats_by_name, transport_cert_stats,
network_report_.get());
// Signal that it is now safe to touch `network_report_` on the signaling
// thread, and post a task to merge it into the final results.
network_report_event_.Set();
rtc::scoped_refptr<RTCStatsCollector> collector(this);
signaling_thread_->PostTask(
[collector] { collector->MergeNetworkReport_s(); });
}
void RTCStatsCollector::ProducePartialResultsOnNetworkThreadImpl(
int64_t timestamp_us,
const std::map<std::string, cricket::TransportStats>&
transport_stats_by_name,
const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
RTCStatsReport* partial_report) {
RTC_DCHECK_RUN_ON(network_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
ProduceCertificateStats_n(timestamp_us, transport_cert_stats, partial_report);
ProduceCodecStats_n(timestamp_us, transceiver_stats_infos_, partial_report);
ProduceIceCandidateAndPairStats_n(timestamp_us, transport_stats_by_name,
call_stats_, partial_report);
ProduceTransportStats_n(timestamp_us, transport_stats_by_name,
transport_cert_stats, partial_report);
ProduceRTPStreamStats_n(timestamp_us, transceiver_stats_infos_,
partial_report);
}
void RTCStatsCollector::MergeNetworkReport_s() {
RTC_DCHECK_RUN_ON(signaling_thread_);
// The `network_report_event_` must be signaled for it to be safe to touch
// `network_report_`. This is normally not blocking, but if
// WaitForPendingRequest() is called while a request is pending, we might have
// to wait until the network thread is done touching `network_report_`.
network_report_event_.Wait(rtc::Event::kForever);
if (!network_report_) {
// Normally, MergeNetworkReport_s() is executed because it is posted from
// the network thread. But if WaitForPendingRequest() is called while a
// request is pending, an early call to MergeNetworkReport_s() is made,
// merging the report and setting `network_report_` to null. If so, when the
// previously posted MergeNetworkReport_s() is later executed, the report is
// already null and nothing needs to be done here.
return;
}
RTC_DCHECK_GT(num_pending_partial_reports_, 0);
RTC_DCHECK(partial_report_);
partial_report_->TakeMembersFrom(network_report_);
network_report_ = nullptr;
--num_pending_partial_reports_;
// `network_report_` is currently the only partial report collected
// asynchronously, so `num_pending_partial_reports_` must now be 0 and we are
// ready to deliver the result.
RTC_DCHECK_EQ(num_pending_partial_reports_, 0);
cache_timestamp_us_ = partial_report_timestamp_us_;
cached_report_ = partial_report_;
partial_report_ = nullptr;
transceiver_stats_infos_.clear();
// Trace WebRTC Stats when getStats is called on Javascript.
// This allows access to WebRTC stats from trace logs. To enable them,
// select the "webrtc_stats" category when recording traces.
TRACE_EVENT_INSTANT1("webrtc_stats", "webrtc_stats", "report",
cached_report_->ToJson());
// Deliver report and clear `requests_`.
std::vector<RequestInfo> requests;
requests.swap(requests_);
DeliverCachedReport(cached_report_, std::move(requests));
}
void RTCStatsCollector::DeliverCachedReport(
rtc::scoped_refptr<const RTCStatsReport> cached_report,
std::vector<RTCStatsCollector::RequestInfo> requests) {
RTC_DCHECK_RUN_ON(signaling_thread_);
RTC_DCHECK(!requests.empty());
RTC_DCHECK(cached_report);
for (const RequestInfo& request : requests) {
if (request.filter_mode() == RequestInfo::FilterMode::kAll) {
request.callback()->OnStatsDelivered(cached_report);
} else {
bool filter_by_sender_selector;
rtc::scoped_refptr<RtpSenderInternal> sender_selector;
rtc::scoped_refptr<RtpReceiverInternal> receiver_selector;
if (request.filter_mode() == RequestInfo::FilterMode::kSenderSelector) {
filter_by_sender_selector = true;
sender_selector = request.sender_selector();
} else {
RTC_DCHECK(request.filter_mode() ==
RequestInfo::FilterMode::kReceiverSelector);
filter_by_sender_selector = false;
receiver_selector = request.receiver_selector();
}
request.callback()->OnStatsDelivered(CreateReportFilteredBySelector(
filter_by_sender_selector, cached_report, sender_selector,
receiver_selector));
}
}
}
void RTCStatsCollector::ProduceCertificateStats_n(
int64_t timestamp_us,
const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
RTCStatsReport* report) const {
RTC_DCHECK_RUN_ON(network_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
for (const auto& transport_cert_stats_pair : transport_cert_stats) {
if (transport_cert_stats_pair.second.local) {
ProduceCertificateStatsFromSSLCertificateStats(
timestamp_us, *transport_cert_stats_pair.second.local.get(), report);
}
if (transport_cert_stats_pair.second.remote) {
ProduceCertificateStatsFromSSLCertificateStats(
timestamp_us, *transport_cert_stats_pair.second.remote.get(), report);
}
}
}
void RTCStatsCollector::ProduceCodecStats_n(
int64_t timestamp_us,
const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
RTCStatsReport* report) const {
RTC_DCHECK_RUN_ON(network_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
for (const auto& stats : transceiver_stats_infos) {
if (!stats.mid) {
continue;
}
std::string transport_id = RTCTransportStatsIDFromTransportChannel(
*stats.transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
const cricket::VoiceMediaInfo* voice_media_info =
stats.track_media_info_map->voice_media_info();
const cricket::VideoMediaInfo* video_media_info =
stats.track_media_info_map->video_media_info();
// Audio
if (voice_media_info) {
// Inbound
for (const auto& pair : voice_media_info->receive_codecs) {
report->AddStats(CodecStatsFromRtpCodecParameters(
timestamp_us, *stats.mid, transport_id, true, pair.second));
}
// Outbound
for (const auto& pair : voice_media_info->send_codecs) {
report->AddStats(CodecStatsFromRtpCodecParameters(
timestamp_us, *stats.mid, transport_id, false, pair.second));
}
}
// Video
if (video_media_info) {
// Inbound
for (const auto& pair : video_media_info->receive_codecs) {
report->AddStats(CodecStatsFromRtpCodecParameters(
timestamp_us, *stats.mid, transport_id, true, pair.second));
}
// Outbound
for (const auto& pair : video_media_info->send_codecs) {
report->AddStats(CodecStatsFromRtpCodecParameters(
timestamp_us, *stats.mid, transport_id, false, pair.second));
}
}
}
}
void RTCStatsCollector::ProduceDataChannelStats_s(
int64_t timestamp_us,
RTCStatsReport* report) const {
RTC_DCHECK_RUN_ON(signaling_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
std::vector<DataChannelStats> data_stats = pc_->GetDataChannelStats();
for (const auto& stats : data_stats) {
std::unique_ptr<RTCDataChannelStats> data_channel_stats(
new RTCDataChannelStats(
"RTCDataChannel_" + rtc::ToString(stats.internal_id),
timestamp_us));
data_channel_stats->label = std::move(stats.label);
data_channel_stats->protocol = std::move(stats.protocol);
data_channel_stats->data_channel_identifier = stats.id;
data_channel_stats->state = DataStateToRTCDataChannelState(stats.state);
data_channel_stats->messages_sent = stats.messages_sent;
data_channel_stats->bytes_sent = stats.bytes_sent;
data_channel_stats->messages_received = stats.messages_received;
data_channel_stats->bytes_received = stats.bytes_received;
report->AddStats(std::move(data_channel_stats));
}
}
void RTCStatsCollector::ProduceIceCandidateAndPairStats_n(
int64_t timestamp_us,
const std::map<std::string, cricket::TransportStats>&
transport_stats_by_name,
const Call::Stats& call_stats,
RTCStatsReport* report) const {
RTC_DCHECK_RUN_ON(network_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
for (const auto& entry : transport_stats_by_name) {
const std::string& transport_name = entry.first;
const cricket::TransportStats& transport_stats = entry.second;
for (const auto& channel_stats : transport_stats.channel_stats) {
std::string transport_id = RTCTransportStatsIDFromTransportChannel(
transport_name, channel_stats.component);
for (const cricket::ConnectionInfo& info :
channel_stats.ice_transport_stats.connection_infos) {
std::unique_ptr<RTCIceCandidatePairStats> candidate_pair_stats(
new RTCIceCandidatePairStats(
RTCIceCandidatePairStatsIDFromConnectionInfo(info),
timestamp_us));
candidate_pair_stats->transport_id = transport_id;
// TODO(hbos): There could be other candidates that are not paired with
// anything. We don't have a complete list. Local candidates come from
// Port objects, and prflx candidates (both local and remote) are only
// stored in candidate pairs. https://crbug.com/632723
candidate_pair_stats->local_candidate_id = ProduceIceCandidateStats(
timestamp_us, info.local_candidate, true, transport_id, report);
candidate_pair_stats->remote_candidate_id = ProduceIceCandidateStats(
timestamp_us, info.remote_candidate, false, transport_id, report);
candidate_pair_stats->state =
IceCandidatePairStateToRTCStatsIceCandidatePairState(info.state);
candidate_pair_stats->priority = info.priority;
candidate_pair_stats->nominated = info.nominated;
// TODO(hbos): This writable is different than the spec. It goes to
// false after a certain amount of time without a response passes.
// https://crbug.com/633550
candidate_pair_stats->writable = info.writable;
// Note that sent_total_packets includes discarded packets but
// sent_total_bytes does not.
candidate_pair_stats->packets_sent = static_cast<uint64_t>(
info.sent_total_packets - info.sent_discarded_packets);
candidate_pair_stats->packets_discarded_on_send =
static_cast<uint64_t>(info.sent_discarded_packets);
candidate_pair_stats->packets_received =
static_cast<uint64_t>(info.packets_received);
candidate_pair_stats->bytes_sent =
static_cast<uint64_t>(info.sent_total_bytes);
candidate_pair_stats->bytes_discarded_on_send =
static_cast<uint64_t>(info.sent_discarded_bytes);
candidate_pair_stats->bytes_received =
static_cast<uint64_t>(info.recv_total_bytes);
candidate_pair_stats->total_round_trip_time =
static_cast<double>(info.total_round_trip_time_ms) /
rtc::kNumMillisecsPerSec;
if (info.current_round_trip_time_ms) {
candidate_pair_stats->current_round_trip_time =
static_cast<double>(*info.current_round_trip_time_ms) /
rtc::kNumMillisecsPerSec;
}
if (info.best_connection) {
// The bandwidth estimations we have are for the selected candidate
// pair ("info.best_connection").
RTC_DCHECK_GE(call_stats.send_bandwidth_bps, 0);
RTC_DCHECK_GE(call_stats.recv_bandwidth_bps, 0);
if (call_stats.send_bandwidth_bps > 0) {
candidate_pair_stats->available_outgoing_bitrate =
static_cast<double>(call_stats.send_bandwidth_bps);
}
if (call_stats.recv_bandwidth_bps > 0) {
candidate_pair_stats->available_incoming_bitrate =
static_cast<double>(call_stats.recv_bandwidth_bps);
}
}
candidate_pair_stats->requests_received =
static_cast<uint64_t>(info.recv_ping_requests);
candidate_pair_stats->requests_sent = static_cast<uint64_t>(
info.sent_ping_requests_before_first_response);
candidate_pair_stats->responses_received =
static_cast<uint64_t>(info.recv_ping_responses);
candidate_pair_stats->responses_sent =
static_cast<uint64_t>(info.sent_ping_responses);
RTC_DCHECK_GE(info.sent_ping_requests_total,
info.sent_ping_requests_before_first_response);
candidate_pair_stats->consent_requests_sent = static_cast<uint64_t>(
info.sent_ping_requests_total -
info.sent_ping_requests_before_first_response);
report->AddStats(std::move(candidate_pair_stats));
}
}
}
}
void RTCStatsCollector::ProduceMediaStreamStats_s(
int64_t timestamp_us,
RTCStatsReport* report) const {
RTC_DCHECK_RUN_ON(signaling_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
std::map<std::string, std::vector<std::string>> track_ids;
for (const auto& stats : transceiver_stats_infos_) {
for (const auto& sender : stats.transceiver->senders()) {
std::string track_id =
RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
kSender, sender->internal()->AttachmentId());
for (auto& stream_id : sender->stream_ids()) {
track_ids[stream_id].push_back(track_id);
}
}
for (const auto& receiver : stats.transceiver->receivers()) {
std::string track_id =
RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
kReceiver, receiver->internal()->AttachmentId());
for (auto& stream : receiver->streams()) {
track_ids[stream->id()].push_back(track_id);
}
}
}
// Build stats for each stream ID known.
for (auto& it : track_ids) {
std::unique_ptr<RTCMediaStreamStats> stream_stats(
new RTCMediaStreamStats("RTCMediaStream_" + it.first, timestamp_us));
stream_stats->stream_identifier = it.first;
stream_stats->track_ids = it.second;
report->AddStats(std::move(stream_stats));
}
}
void RTCStatsCollector::ProduceMediaStreamTrackStats_s(
int64_t timestamp_us,
RTCStatsReport* report) const {
RTC_DCHECK_RUN_ON(signaling_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
for (const RtpTransceiverStatsInfo& stats : transceiver_stats_infos_) {
std::vector<rtc::scoped_refptr<RtpSenderInternal>> senders;
for (const auto& sender : stats.transceiver->senders()) {
senders.push_back(
rtc::scoped_refptr<RtpSenderInternal>(sender->internal()));
}
ProduceSenderMediaTrackStats(timestamp_us, *stats.track_media_info_map,
senders, report);
std::vector<rtc::scoped_refptr<RtpReceiverInternal>> receivers;
for (const auto& receiver : stats.transceiver->receivers()) {
receivers.push_back(
rtc::scoped_refptr<RtpReceiverInternal>(receiver->internal()));
}
ProduceReceiverMediaTrackStats(timestamp_us, *stats.track_media_info_map,
receivers, report);
}
}
void RTCStatsCollector::ProduceMediaSourceStats_s(
int64_t timestamp_us,
RTCStatsReport* report) const {
RTC_DCHECK_RUN_ON(signaling_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
for (const RtpTransceiverStatsInfo& transceiver_stats_info :
transceiver_stats_infos_) {
const auto& track_media_info_map =
transceiver_stats_info.track_media_info_map;
for (const auto& sender : transceiver_stats_info.transceiver->senders()) {
const auto& sender_internal = sender->internal();
const auto& track = sender_internal->track();
if (!track)
continue;
// TODO(https://crbug.com/webrtc/10771): The same track could be attached
// to multiple senders which should result in multiple senders referencing
// the same media-source stats. When all media source related metrics are
// moved to the track's source (e.g. input frame rate is moved from
// cricket::VideoSenderInfo to VideoTrackSourceInterface::Stats and audio
// levels are moved to the corresponding audio track/source object), don't
// create separate media source stats objects on a per-attachment basis.
std::unique_ptr<RTCMediaSourceStats> media_source_stats;
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
AudioTrackInterface* audio_track =
static_cast<AudioTrackInterface*>(track.get());
auto audio_source_stats = std::make_unique<RTCAudioSourceStats>(
RTCMediaSourceStatsIDFromKindAndAttachment(
cricket::MEDIA_TYPE_AUDIO, sender_internal->AttachmentId()),
timestamp_us);
// TODO(https://crbug.com/webrtc/10771): We shouldn't need to have an
// SSRC assigned (there shouldn't need to exist a send-stream, created
// by an O/A exchange) in order to read audio media-source stats.
// TODO(https://crbug.com/webrtc/8694): SSRC 0 shouldn't be a magic
// value indicating no SSRC.
if (sender_internal->ssrc() != 0) {
auto* voice_sender_info =
track_media_info_map->GetVoiceSenderInfoBySsrc(
sender_internal->ssrc());
if (voice_sender_info) {
audio_source_stats->audio_level = DoubleAudioLevelFromIntAudioLevel(
voice_sender_info->audio_level);
audio_source_stats->total_audio_energy =
voice_sender_info->total_input_energy;
audio_source_stats->total_samples_duration =
voice_sender_info->total_input_duration;
SetAudioProcessingStats(audio_source_stats.get(),
voice_sender_info->apm_statistics);
}
}
// Audio processor may be attached to either the track or the send
// stream, so look in both places.
auto audio_processor(audio_track->GetAudioProcessor());
if (audio_processor.get()) {
// The `has_remote_tracks` argument is obsolete; makes no difference
// if it's set to true or false.
AudioProcessorInterface::AudioProcessorStatistics ap_stats =
audio_processor->GetStats(/*has_remote_tracks=*/false);
SetAudioProcessingStats(audio_source_stats.get(),
ap_stats.apm_statistics);
}
media_source_stats = std::move(audio_source_stats);
} else {
RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());
auto video_source_stats = std::make_unique<RTCVideoSourceStats>(
RTCMediaSourceStatsIDFromKindAndAttachment(
cricket::MEDIA_TYPE_VIDEO, sender_internal->AttachmentId()),
timestamp_us);
auto* video_track = static_cast<VideoTrackInterface*>(track.get());
auto* video_source = video_track->GetSource();
VideoTrackSourceInterface::Stats source_stats;
if (video_source && video_source->GetStats(&source_stats)) {
video_source_stats->width = source_stats.input_width;
video_source_stats->height = source_stats.input_height;
}
// TODO(https://crbug.com/webrtc/10771): We shouldn't need to have an
// SSRC assigned (there shouldn't need to exist a send-stream, created
// by an O/A exchange) in order to get framesPerSecond.
// TODO(https://crbug.com/webrtc/8694): SSRC 0 shouldn't be a magic
// value indicating no SSRC.
if (sender_internal->ssrc() != 0) {
auto* video_sender_info =
track_media_info_map->GetVideoSenderInfoBySsrc(
sender_internal->ssrc());
if (video_sender_info) {
video_source_stats->frames_per_second =
video_sender_info->framerate_input;
video_source_stats->frames = video_sender_info->frames;
}
}
media_source_stats = std::move(video_source_stats);
}
media_source_stats->track_identifier = track->id();
media_source_stats->kind = track->kind();
report->AddStats(std::move(media_source_stats));
}
}
}
void RTCStatsCollector::ProducePeerConnectionStats_s(
int64_t timestamp_us,
RTCStatsReport* report) const {
RTC_DCHECK_RUN_ON(signaling_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
std::unique_ptr<RTCPeerConnectionStats> stats(
new RTCPeerConnectionStats("RTCPeerConnection", timestamp_us));
stats->data_channels_opened = internal_record_.data_channels_opened;
stats->data_channels_closed = internal_record_.data_channels_closed;
report->AddStats(std::move(stats));
}
void RTCStatsCollector::ProduceRTPStreamStats_n(
int64_t timestamp_us,
const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
RTCStatsReport* report) const {
RTC_DCHECK_RUN_ON(network_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
for (const RtpTransceiverStatsInfo& stats : transceiver_stats_infos) {
if (stats.media_type == cricket::MEDIA_TYPE_AUDIO) {
ProduceAudioRTPStreamStats_n(timestamp_us, stats, report);
} else if (stats.media_type == cricket::MEDIA_TYPE_VIDEO) {
ProduceVideoRTPStreamStats_n(timestamp_us, stats, report);
} else {
RTC_DCHECK_NOTREACHED();
}
}
}
void RTCStatsCollector::ProduceAudioRTPStreamStats_n(
int64_t timestamp_us,
const RtpTransceiverStatsInfo& stats,
RTCStatsReport* report) const {
RTC_DCHECK_RUN_ON(network_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
if (!stats.mid || !stats.transport_name) {
return;
}
RTC_DCHECK(stats.track_media_info_map);
const TrackMediaInfoMap& track_media_info_map = *stats.track_media_info_map;
RTC_DCHECK(track_media_info_map.voice_media_info());
std::string mid = *stats.mid;
std::string transport_id = RTCTransportStatsIDFromTransportChannel(
*stats.transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
// Inbound and remote-outbound.
// The remote-outbound stats are based on RTCP sender reports sent from the
// remote endpoint providing metrics about the remote outbound streams.
for (const cricket::VoiceReceiverInfo& voice_receiver_info :
track_media_info_map.voice_media_info()->receivers) {
if (!voice_receiver_info.connected())
continue;
// Inbound.
auto inbound_audio =
CreateInboundAudioStreamStats(voice_receiver_info, mid, timestamp_us);
// TODO(hta): This lookup should look for the sender, not the track.
rtc::scoped_refptr<AudioTrackInterface> audio_track =
track_media_info_map.GetAudioTrack(voice_receiver_info);
if (audio_track) {
inbound_audio->track_id =
RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
kReceiver,
track_media_info_map.GetAttachmentIdByTrack(audio_track.get())
.value());
}
inbound_audio->transport_id = transport_id;
// Remote-outbound.
auto remote_outbound_audio = CreateRemoteOutboundAudioStreamStats(
voice_receiver_info, mid, inbound_audio->id(), transport_id);
// Add stats.
if (remote_outbound_audio) {
// When the remote outbound stats are available, the remote ID for the
// local inbound stats is set.
inbound_audio->remote_id = remote_outbound_audio->id();
report->AddStats(std::move(remote_outbound_audio));
}
report->AddStats(std::move(inbound_audio));
}
// Outbound.
std::map<std::string, RTCOutboundRTPStreamStats*> audio_outbound_rtps;
for (const cricket::VoiceSenderInfo& voice_sender_info :
track_media_info_map.voice_media_info()->senders) {
if (!voice_sender_info.connected())
continue;
auto outbound_audio = std::make_unique<RTCOutboundRTPStreamStats>(
RTCOutboundRTPStreamStatsIDFromSSRC(cricket::MEDIA_TYPE_AUDIO,
voice_sender_info.ssrc()),
timestamp_us);
SetOutboundRTPStreamStatsFromVoiceSenderInfo(mid, voice_sender_info,
outbound_audio.get());
rtc::scoped_refptr<AudioTrackInterface> audio_track =
track_media_info_map.GetAudioTrack(voice_sender_info);
if (audio_track) {
int attachment_id =
track_media_info_map.GetAttachmentIdByTrack(audio_track.get())
.value();
outbound_audio->track_id =
RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(kSender,
attachment_id);
outbound_audio->media_source_id =
RTCMediaSourceStatsIDFromKindAndAttachment(cricket::MEDIA_TYPE_AUDIO,
attachment_id);
}
outbound_audio->transport_id = transport_id;
audio_outbound_rtps.insert(
std::make_pair(outbound_audio->id(), outbound_audio.get()));
report->AddStats(std::move(outbound_audio));
}
// Remote-inbound.
// These are Report Block-based, information sent from the remote endpoint,
// providing metrics about our Outbound streams. We take advantage of the fact
// that RTCOutboundRtpStreamStats, RTCCodecStats and RTCTransport have already
// been added to the report.
for (const cricket::VoiceSenderInfo& voice_sender_info :
track_media_info_map.voice_media_info()->senders) {
for (const auto& report_block_data : voice_sender_info.report_block_datas) {
report->AddStats(ProduceRemoteInboundRtpStreamStatsFromReportBlockData(
report_block_data, cricket::MEDIA_TYPE_AUDIO, audio_outbound_rtps,
*report));
}
}
}
void RTCStatsCollector::ProduceVideoRTPStreamStats_n(
int64_t timestamp_us,
const RtpTransceiverStatsInfo& stats,
RTCStatsReport* report) const {
RTC_DCHECK_RUN_ON(network_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
if (!stats.mid || !stats.transport_name) {
return;
}
RTC_DCHECK(stats.track_media_info_map);
const TrackMediaInfoMap& track_media_info_map = *stats.track_media_info_map;
RTC_DCHECK(track_media_info_map.video_media_info());
std::string mid = *stats.mid;
std::string transport_id = RTCTransportStatsIDFromTransportChannel(
*stats.transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
// Inbound
for (const cricket::VideoReceiverInfo& video_receiver_info :
track_media_info_map.video_media_info()->receivers) {
if (!video_receiver_info.connected())
continue;
auto inbound_video = std::make_unique<RTCInboundRTPStreamStats>(
RTCInboundRTPStreamStatsIDFromSSRC(cricket::MEDIA_TYPE_VIDEO,
video_receiver_info.ssrc()),
timestamp_us);
SetInboundRTPStreamStatsFromVideoReceiverInfo(mid, video_receiver_info,
inbound_video.get());
rtc::scoped_refptr<VideoTrackInterface> video_track =
track_media_info_map.GetVideoTrack(video_receiver_info);
if (video_track) {
inbound_video->track_id =
RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
kReceiver,
track_media_info_map.GetAttachmentIdByTrack(video_track.get())
.value());
}
inbound_video->transport_id = transport_id;
report->AddStats(std::move(inbound_video));
// TODO(crbug.com/webrtc/12529): Add remote-outbound stats.
}
// Outbound
std::map<std::string, RTCOutboundRTPStreamStats*> video_outbound_rtps;
for (const cricket::VideoSenderInfo& video_sender_info :
track_media_info_map.video_media_info()->senders) {
if (!video_sender_info.connected())
continue;
auto outbound_video = std::make_unique<RTCOutboundRTPStreamStats>(
RTCOutboundRTPStreamStatsIDFromSSRC(cricket::MEDIA_TYPE_VIDEO,
video_sender_info.ssrc()),
timestamp_us);
SetOutboundRTPStreamStatsFromVideoSenderInfo(mid, video_sender_info,
outbound_video.get());
rtc::scoped_refptr<VideoTrackInterface> video_track =
track_media_info_map.GetVideoTrack(video_sender_info);
if (video_track) {
int attachment_id =
track_media_info_map.GetAttachmentIdByTrack(video_track.get())
.value();
outbound_video->track_id =
RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(kSender,
attachment_id);
outbound_video->media_source_id =
RTCMediaSourceStatsIDFromKindAndAttachment(cricket::MEDIA_TYPE_VIDEO,
attachment_id);
}
outbound_video->transport_id = transport_id;
video_outbound_rtps.insert(
std::make_pair(outbound_video->id(), outbound_video.get()));
report->AddStats(std::move(outbound_video));
}
// Remote-inbound
// These are Report Block-based, information sent from the remote endpoint,
// providing metrics about our Outbound streams. We take advantage of the fact
// that RTCOutboundRtpStreamStats, RTCCodecStats and RTCTransport have already
// been added to the report.
for (const cricket::VideoSenderInfo& video_sender_info :
track_media_info_map.video_media_info()->senders) {
for (const auto& report_block_data : video_sender_info.report_block_datas) {
report->AddStats(ProduceRemoteInboundRtpStreamStatsFromReportBlockData(
report_block_data, cricket::MEDIA_TYPE_VIDEO, video_outbound_rtps,
*report));
}
}
}
void RTCStatsCollector::ProduceTransportStats_n(
int64_t timestamp_us,
const std::map<std::string, cricket::TransportStats>&
transport_stats_by_name,
const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
RTCStatsReport* report) const {
RTC_DCHECK_RUN_ON(network_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
for (const auto& entry : transport_stats_by_name) {
const std::string& transport_name = entry.first;
const cricket::TransportStats& transport_stats = entry.second;
// Get reference to RTCP channel, if it exists.
std::string rtcp_transport_stats_id;
for (const cricket::TransportChannelStats& channel_stats :
transport_stats.channel_stats) {
if (channel_stats.component == cricket::ICE_CANDIDATE_COMPONENT_RTCP) {
rtcp_transport_stats_id = RTCTransportStatsIDFromTransportChannel(
transport_name, channel_stats.component);
break;
}
}
// Get reference to local and remote certificates of this transport, if they
// exist.
const auto& certificate_stats_it =
transport_cert_stats.find(transport_name);
RTC_DCHECK(certificate_stats_it != transport_cert_stats.cend());
std::string local_certificate_id;
if (certificate_stats_it->second.local) {
local_certificate_id = RTCCertificateIDFromFingerprint(
certificate_stats_it->second.local->fingerprint);
}
std::string remote_certificate_id;
if (certificate_stats_it->second.remote) {
remote_certificate_id = RTCCertificateIDFromFingerprint(
certificate_stats_it->second.remote->fingerprint);
}
// There is one transport stats for each channel.
for (const cricket::TransportChannelStats& channel_stats :
transport_stats.channel_stats) {
std::unique_ptr<RTCTransportStats> transport_stats(
new RTCTransportStats(RTCTransportStatsIDFromTransportChannel(
transport_name, channel_stats.component),
timestamp_us));
transport_stats->packets_sent =
channel_stats.ice_transport_stats.packets_sent;
transport_stats->packets_received =
channel_stats.ice_transport_stats.packets_received;
transport_stats->bytes_sent =
channel_stats.ice_transport_stats.bytes_sent;
transport_stats->bytes_received =
channel_stats.ice_transport_stats.bytes_received;
transport_stats->dtls_state =
DtlsTransportStateToRTCDtlsTransportState(channel_stats.dtls_state);
transport_stats->selected_candidate_pair_changes =
channel_stats.ice_transport_stats.selected_candidate_pair_changes;
transport_stats->ice_role =
IceRoleToRTCIceRole(channel_stats.ice_transport_stats.ice_role);
transport_stats->ice_local_username_fragment =
channel_stats.ice_transport_stats.ice_local_username_fragment;
transport_stats->ice_state = IceTransportStateToRTCIceTransportState(
channel_stats.ice_transport_stats.ice_state);
for (const cricket::ConnectionInfo& info :
channel_stats.ice_transport_stats.connection_infos) {
if (info.best_connection) {
transport_stats->selected_candidate_pair_id =
RTCIceCandidatePairStatsIDFromConnectionInfo(info);
}
}
if (channel_stats.component != cricket::ICE_CANDIDATE_COMPONENT_RTCP &&
!rtcp_transport_stats_id.empty()) {
transport_stats->rtcp_transport_stats_id = rtcp_transport_stats_id;
}
if (!local_certificate_id.empty())
transport_stats->local_certificate_id = local_certificate_id;
if (!remote_certificate_id.empty())
transport_stats->remote_certificate_id = remote_certificate_id;
// Crypto information
if (channel_stats.ssl_version_bytes) {
char bytes[5];
snprintf(bytes, sizeof(bytes), "%04X", channel_stats.ssl_version_bytes);
transport_stats->tls_version = bytes;
}
if (channel_stats.dtls_role) {
transport_stats->dtls_role = *channel_stats.dtls_role == rtc::SSL_CLIENT
? webrtc::RTCDtlsRole::kClient
: webrtc::RTCDtlsRole::kServer;
} else {
transport_stats->dtls_role = webrtc::RTCDtlsRole::kUnknown;
}
if (channel_stats.ssl_cipher_suite != rtc::kTlsNullWithNullNull &&
rtc::SSLStreamAdapter::SslCipherSuiteToName(
channel_stats.ssl_cipher_suite)
.length()) {
transport_stats->dtls_cipher =
rtc::SSLStreamAdapter::SslCipherSuiteToName(
channel_stats.ssl_cipher_suite);
}
if (channel_stats.srtp_crypto_suite != rtc::kSrtpInvalidCryptoSuite &&
rtc::SrtpCryptoSuiteToName(channel_stats.srtp_crypto_suite)
.length()) {
transport_stats->srtp_cipher =
rtc::SrtpCryptoSuiteToName(channel_stats.srtp_crypto_suite);
}
report->AddStats(std::move(transport_stats));
}
}
}
std::map<std::string, RTCStatsCollector::CertificateStatsPair>
RTCStatsCollector::PrepareTransportCertificateStats_n(
const std::map<std::string, cricket::TransportStats>&
transport_stats_by_name) const {
RTC_DCHECK_RUN_ON(network_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
std::map<std::string, CertificateStatsPair> transport_cert_stats;
for (const auto& entry : transport_stats_by_name) {
const std::string& transport_name = entry.first;
CertificateStatsPair certificate_stats_pair;
rtc::scoped_refptr<rtc::RTCCertificate> local_certificate;
if (pc_->GetLocalCertificate(transport_name, &local_certificate)) {
certificate_stats_pair.local =
local_certificate->GetSSLCertificateChain().GetStats();
}
std::unique_ptr<rtc::SSLCertChain> remote_cert_chain =
pc_->GetRemoteSSLCertChain(transport_name);
if (remote_cert_chain) {
certificate_stats_pair.remote = remote_cert_chain->GetStats();
}
transport_cert_stats.insert(
std::make_pair(transport_name, std::move(certificate_stats_pair)));
}
return transport_cert_stats;
}
void RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w_n() {
RTC_DCHECK_RUN_ON(signaling_thread_);
transceiver_stats_infos_.clear();
// These are used to invoke GetStats for all the media channels together in
// one worker thread hop.
std::map<cricket::MediaChannel*, std::unique_ptr<cricket::VoiceMediaInfo>>
voice_stats;
std::map<cricket::MediaChannel*, std::unique_ptr<cricket::VideoMediaInfo>>
video_stats;
auto transceivers = pc_->GetTransceiversInternal();
// TODO(tommi): See if we can avoid synchronously blocking the signaling
// thread while we do this (or avoid the Invoke at all).
network_thread_->Invoke<void>(RTC_FROM_HERE, [this, &transceivers,
&voice_stats, &video_stats] {
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
for (const auto& transceiver_proxy : transceivers) {
RtpTransceiver* transceiver = transceiver_proxy->internal();
cricket::MediaType media_type = transceiver->media_type();
// Prepare stats entry. The TrackMediaInfoMap will be filled in after the
// stats have been fetched on the worker thread.
transceiver_stats_infos_.emplace_back();
RtpTransceiverStatsInfo& stats = transceiver_stats_infos_.back();
stats.transceiver = transceiver;
stats.media_type = media_type;
cricket::ChannelInterface* channel = transceiver->channel();
if (!channel) {
// The remaining fields require a BaseChannel.
continue;
}
stats.mid = channel->mid();
stats.transport_name = std::string(channel->transport_name());
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
RTC_DCHECK(voice_stats.find(channel->media_channel()) ==
voice_stats.end());
voice_stats[channel->media_channel()] =
std::make_unique<cricket::VoiceMediaInfo>();
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
RTC_DCHECK(video_stats.find(channel->media_channel()) ==
video_stats.end());
video_stats[channel->media_channel()] =
std::make_unique<cricket::VideoMediaInfo>();
} else {
RTC_DCHECK_NOTREACHED();
}
}
});
// We jump to the worker thread and call GetStats() on each media channel as
// well as GetCallStats(). At the same time we construct the
// TrackMediaInfoMaps, which also needs info from the worker thread. This
// minimizes the number of thread jumps.
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
for (const auto& entry : voice_stats) {
if (!static_cast<cricket::VoiceMediaChannel*>(entry.first)
->GetStats(entry.second.get(),
/*get_and_clear_legacy_stats=*/false)) {
RTC_LOG(LS_WARNING) << "Failed to get voice stats.";
}
}
for (const auto& entry : video_stats) {
if (!static_cast<cricket::VideoMediaChannel*>(entry.first)
->GetStats(entry.second.get())) {
RTC_LOG(LS_WARNING) << "Failed to get video stats.";
}
}
// Create the TrackMediaInfoMap for each transceiver stats object.
for (auto& stats : transceiver_stats_infos_) {
auto transceiver = stats.transceiver;
std::unique_ptr<cricket::VoiceMediaInfo> voice_media_info;
std::unique_ptr<cricket::VideoMediaInfo> video_media_info;
auto channel = transceiver->channel();
if (channel) {
cricket::MediaType media_type = transceiver->media_type();
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
RTC_DCHECK(voice_stats[channel->media_channel()]);
voice_media_info = std::move(voice_stats[channel->media_channel()]);
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
RTC_DCHECK(video_stats[channel->media_channel()]);
video_media_info = std::move(video_stats[channel->media_channel()]);
}
}
std::vector<rtc::scoped_refptr<RtpSenderInternal>> senders;
for (const auto& sender : transceiver->senders()) {
senders.push_back(
rtc::scoped_refptr<RtpSenderInternal>(sender->internal()));
}
std::vector<rtc::scoped_refptr<RtpReceiverInternal>> receivers;
for (const auto& receiver : transceiver->receivers()) {
receivers.push_back(
rtc::scoped_refptr<RtpReceiverInternal>(receiver->internal()));
}
stats.track_media_info_map = std::make_unique<TrackMediaInfoMap>(
std::move(voice_media_info), std::move(video_media_info), senders,
receivers);
}
call_stats_ = pc_->GetCallStats();
});
}
void RTCStatsCollector::OnSctpDataChannelCreated(SctpDataChannel* channel) {
channel->SignalOpened.connect(this, &RTCStatsCollector::OnDataChannelOpened);
channel->SignalClosed.connect(this, &RTCStatsCollector::OnDataChannelClosed);
}
void RTCStatsCollector::OnDataChannelOpened(DataChannelInterface* channel) {
RTC_DCHECK_RUN_ON(signaling_thread_);
bool result = internal_record_.opened_data_channels
.insert(reinterpret_cast<uintptr_t>(channel))
.second;
++internal_record_.data_channels_opened;
RTC_DCHECK(result);
}
void RTCStatsCollector::OnDataChannelClosed(DataChannelInterface* channel) {
RTC_DCHECK_RUN_ON(signaling_thread_);
// Only channels that have been fully opened (and have increased the
// `data_channels_opened_` counter) increase the closed counter.
if (internal_record_.opened_data_channels.erase(
reinterpret_cast<uintptr_t>(channel))) {
++internal_record_.data_channels_closed;
}
}
const char* CandidateTypeToRTCIceCandidateTypeForTesting(
const std::string& type) {
return CandidateTypeToRTCIceCandidateType(type);
}
const char* DataStateToRTCDataChannelStateForTesting(
DataChannelInterface::DataState state) {
return DataStateToRTCDataChannelState(state);
}
} // namespace webrtc
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