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Call: prepare receive stats for thread switch.

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This change collects the receive stats that Call
maintains into a new thread-compatible
internal class which can easily be switched to
the network thread.

Bug: webrtc:11993
Change-Id: I9fa9a7f057149789aa327e5ba8a8cb3379762272
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220760
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34158}
This commit is contained in:
Markus Handell 2021-05-31 09:02:01 +02:00 committed by WebRTC LUCI CQ
parent 22fead3041
commit c81afe365d
1 changed files with 128 additions and 83 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

211
call/call.cc
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@ -321,6 +321,37 @@ class Call final : public webrtc::Call,
void SetClientBitratePreferences(const BitrateSettings& preferences) override;
private:
// Thread-compatible class that collects received packet stats and exposes
// them as UMA histograms on destruction.
class ReceiveStats {
public:
explicit ReceiveStats(Clock* clock);
~ReceiveStats();
void AddReceivedRtcpBytes(int bytes);
void AddReceivedAudioBytes(int bytes, webrtc::Timestamp arrival_time);
void AddReceivedVideoBytes(int bytes, webrtc::Timestamp arrival_time);
private:
SequenceChecker sequence_checker_;
RateCounter received_bytes_per_second_counter_
RTC_GUARDED_BY(sequence_checker_);
RateCounter received_audio_bytes_per_second_counter_
RTC_GUARDED_BY(sequence_checker_);
RateCounter received_video_bytes_per_second_counter_
RTC_GUARDED_BY(sequence_checker_);
RateCounter received_rtcp_bytes_per_second_counter_
RTC_GUARDED_BY(sequence_checker_);
absl::optional<Timestamp> first_received_rtp_audio_timestamp_
RTC_GUARDED_BY(sequence_checker_);
absl::optional<Timestamp> last_received_rtp_audio_timestamp_
RTC_GUARDED_BY(sequence_checker_);
absl::optional<Timestamp> first_received_rtp_video_timestamp_
RTC_GUARDED_BY(sequence_checker_);
absl::optional<Timestamp> last_received_rtp_video_timestamp_
RTC_GUARDED_BY(sequence_checker_);
};
DeliveryStatus DeliverRtcp(MediaType media_type,
const uint8_t* packet,
size_t length)
@ -336,7 +367,6 @@ class Call final : public webrtc::Call,
MediaType media_type)
RTC_SHARED_LOCKS_REQUIRED(worker_thread_);
void UpdateReceiveHistograms();
void UpdateAggregateNetworkState();
// Ensure that necessary process threads are started, and any required
@ -433,17 +463,9 @@ class Call final : public webrtc::Call,
webrtc::RtcEventLog* event_log_;
// The following members are only accessed (exclusively) from one thread and
// from the destructor, and therefore doesn't need any explicit
// synchronization.
RateCounter received_bytes_per_second_counter_;
RateCounter received_audio_bytes_per_second_counter_;
RateCounter received_video_bytes_per_second_counter_;
RateCounter received_rtcp_bytes_per_second_counter_;
absl::optional<int64_t> first_received_rtp_audio_ms_;
absl::optional<int64_t> last_received_rtp_audio_ms_;
absl::optional<int64_t> first_received_rtp_video_ms_;
absl::optional<int64_t> last_received_rtp_video_ms_;
// TODO(bugs.webrtc.org/11993) ready to move receive stats access to the
// network thread.
ReceiveStats receive_stats_ RTC_GUARDED_BY(worker_thread_);
uint32_t last_bandwidth_bps_ RTC_GUARDED_BY(worker_thread_);
// TODO(holmer): Remove this lock once BitrateController no longer calls
@ -625,6 +647,94 @@ VideoSendStream* Call::CreateVideoSendStream(
namespace internal {
Call::ReceiveStats::ReceiveStats(Clock* clock)
: received_bytes_per_second_counter_(clock, nullptr, false),
received_audio_bytes_per_second_counter_(clock, nullptr, false),
received_video_bytes_per_second_counter_(clock, nullptr, false),
received_rtcp_bytes_per_second_counter_(clock, nullptr, false) {
sequence_checker_.Detach();
}
void Call::ReceiveStats::AddReceivedRtcpBytes(int bytes) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
if (received_bytes_per_second_counter_.HasSample()) {
// First RTP packet has been received.
received_bytes_per_second_counter_.Add(static_cast<int>(bytes));
received_rtcp_bytes_per_second_counter_.Add(static_cast<int>(bytes));
}
}
void Call::ReceiveStats::AddReceivedAudioBytes(int bytes,
webrtc::Timestamp arrival_time) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
received_bytes_per_second_counter_.Add(bytes);
received_audio_bytes_per_second_counter_.Add(bytes);
if (!first_received_rtp_audio_timestamp_)
first_received_rtp_audio_timestamp_ = arrival_time;
last_received_rtp_audio_timestamp_ = arrival_time;
}
void Call::ReceiveStats::AddReceivedVideoBytes(int bytes,
webrtc::Timestamp arrival_time) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
received_bytes_per_second_counter_.Add(bytes);
received_video_bytes_per_second_counter_.Add(bytes);
if (!first_received_rtp_video_timestamp_)
first_received_rtp_video_timestamp_ = arrival_time;
last_received_rtp_video_timestamp_ = arrival_time;
}
Call::ReceiveStats::~ReceiveStats() {
RTC_DCHECK_RUN_ON(&sequence_checker_);
if (first_received_rtp_audio_timestamp_) {
RTC_HISTOGRAM_COUNTS_100000(
"WebRTC.Call.TimeReceivingAudioRtpPacketsInSeconds",
(*last_received_rtp_audio_timestamp_ -
*first_received_rtp_audio_timestamp_)
.seconds());
}
if (first_received_rtp_video_timestamp_) {
RTC_HISTOGRAM_COUNTS_100000(
"WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds",
(*last_received_rtp_video_timestamp_ -
*first_received_rtp_video_timestamp_)
.seconds());
}
const int kMinRequiredPeriodicSamples = 5;
AggregatedStats video_bytes_per_sec =
received_video_bytes_per_second_counter_.GetStats();
if (video_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
video_bytes_per_sec.average * 8 / 1000);
RTC_LOG(LS_INFO) << "WebRTC.Call.VideoBitrateReceivedInBps, "
<< video_bytes_per_sec.ToStringWithMultiplier(8);
}
AggregatedStats audio_bytes_per_sec =
received_audio_bytes_per_second_counter_.GetStats();
if (audio_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
audio_bytes_per_sec.average * 8 / 1000);
RTC_LOG(LS_INFO) << "WebRTC.Call.AudioBitrateReceivedInBps, "
<< audio_bytes_per_sec.ToStringWithMultiplier(8);
}
AggregatedStats rtcp_bytes_per_sec =
received_rtcp_bytes_per_second_counter_.GetStats();
if (rtcp_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
rtcp_bytes_per_sec.average * 8);
RTC_LOG(LS_INFO) << "WebRTC.Call.RtcpBitrateReceivedInBps, "
<< rtcp_bytes_per_sec.ToStringWithMultiplier(8);
}
AggregatedStats recv_bytes_per_sec =
received_bytes_per_second_counter_.GetStats();
if (recv_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps",
recv_bytes_per_sec.average * 8 / 1000);
RTC_LOG(LS_INFO) << "WebRTC.Call.BitrateReceivedInBps, "
<< recv_bytes_per_sec.ToStringWithMultiplier(8);
}
}
Call::Call(Clock* clock,
const Call::Config& config,
std::unique_ptr<RtpTransportControllerSendInterface> transport_send,
@ -646,10 +756,7 @@ Call::Call(Clock* clock,
video_network_state_(kNetworkDown),
aggregate_network_up_(false),
event_log_(config.event_log),
received_bytes_per_second_counter_(clock_, nullptr, true),
received_audio_bytes_per_second_counter_(clock_, nullptr, true),
received_video_bytes_per_second_counter_(clock_, nullptr, true),
received_rtcp_bytes_per_second_counter_(clock_, nullptr, true),
receive_stats_(clock_),
last_bandwidth_bps_(0),
min_allocated_send_bitrate_bps_(0),
configured_max_padding_bitrate_bps_(0),
@ -710,8 +817,6 @@ Call::~Call() {
pacer_bitrate_kbps_counter_);
}
UpdateReceiveHistograms();
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.LifetimeInSeconds",
(now.ms() - start_ms_) / 1000);
}
@ -737,52 +842,6 @@ void Call::SetClientBitratePreferences(const BitrateSettings& preferences) {
GetTransportControllerSend()->SetClientBitratePreferences(preferences);
}
void Call::UpdateReceiveHistograms() {
if (first_received_rtp_audio_ms_) {
RTC_HISTOGRAM_COUNTS_100000(
"WebRTC.Call.TimeReceivingAudioRtpPacketsInSeconds",
(*last_received_rtp_audio_ms_ - *first_received_rtp_audio_ms_) / 1000);
}
if (first_received_rtp_video_ms_) {
RTC_HISTOGRAM_COUNTS_100000(
"WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds",
(*last_received_rtp_video_ms_ - *first_received_rtp_video_ms_) / 1000);
}
const int kMinRequiredPeriodicSamples = 5;
AggregatedStats video_bytes_per_sec =
received_video_bytes_per_second_counter_.GetStats();
if (video_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
video_bytes_per_sec.average * 8 / 1000);
RTC_LOG(LS_INFO) << "WebRTC.Call.VideoBitrateReceivedInBps, "
<< video_bytes_per_sec.ToStringWithMultiplier(8);
}
AggregatedStats audio_bytes_per_sec =
received_audio_bytes_per_second_counter_.GetStats();
if (audio_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
audio_bytes_per_sec.average * 8 / 1000);
RTC_LOG(LS_INFO) << "WebRTC.Call.AudioBitrateReceivedInBps, "
<< audio_bytes_per_sec.ToStringWithMultiplier(8);
}
AggregatedStats rtcp_bytes_per_sec =
received_rtcp_bytes_per_second_counter_.GetStats();
if (rtcp_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
rtcp_bytes_per_sec.average * 8);
RTC_LOG(LS_INFO) << "WebRTC.Call.RtcpBitrateReceivedInBps, "
<< rtcp_bytes_per_sec.ToStringWithMultiplier(8);
}
AggregatedStats recv_bytes_per_sec =
received_bytes_per_second_counter_.GetStats();
if (recv_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps",
recv_bytes_per_sec.average * 8 / 1000);
RTC_LOG(LS_INFO) << "WebRTC.Call.BitrateReceivedInBps, "
<< recv_bytes_per_sec.ToStringWithMultiplier(8);
}
}
PacketReceiver* Call::Receiver() {
return this;
}
@ -1381,11 +1440,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
// TODO(pbos): Make sure it's a valid packet.
// Return DELIVERY_UNKNOWN_SSRC if it can be determined that
// there's no receiver of the packet.
if (received_bytes_per_second_counter_.HasSample()) {
// First RTP packet has been received.
received_bytes_per_second_counter_.Add(static_cast<int>(length));
received_rtcp_bytes_per_second_counter_.Add(static_cast<int>(length));
}
receive_stats_.AddReceivedRtcpBytes(static_cast<int>(length));
bool rtcp_delivered = false;
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
for (VideoReceiveStream2* stream : video_receive_streams_) {
@ -1471,29 +1526,19 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
int length = static_cast<int>(parsed_packet.size());
if (media_type == MediaType::AUDIO) {
if (audio_receiver_controller_.OnRtpPacket(parsed_packet)) {
received_bytes_per_second_counter_.Add(length);
received_audio_bytes_per_second_counter_.Add(length);
receive_stats_.AddReceivedAudioBytes(length,
parsed_packet.arrival_time());
event_log_->Log(
std::make_unique<RtcEventRtpPacketIncoming>(parsed_packet));
const int64_t arrival_time_ms = parsed_packet.arrival_time().ms();
if (!first_received_rtp_audio_ms_) {
first_received_rtp_audio_ms_.emplace(arrival_time_ms);
}
last_received_rtp_audio_ms_.emplace(arrival_time_ms);
return DELIVERY_OK;
}
} else if (media_type == MediaType::VIDEO) {
parsed_packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
if (video_receiver_controller_.OnRtpPacket(parsed_packet)) {
received_bytes_per_second_counter_.Add(length);
received_video_bytes_per_second_counter_.Add(length);
receive_stats_.AddReceivedVideoBytes(length,
parsed_packet.arrival_time());
event_log_->Log(
std::make_unique<RtcEventRtpPacketIncoming>(parsed_packet));
const int64_t arrival_time_ms = parsed_packet.arrival_time().ms();
if (!first_received_rtp_video_ms_) {
first_received_rtp_video_ms_.emplace(arrival_time_ms);
}
last_received_rtp_video_ms_.emplace(arrival_time_ms);
return DELIVERY_OK;
}
}