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webrtc/src/7040fbcb53/./video/receive_statistics_proxy.cc
blob: 2d1a283a56b86c4784fc28c3b79ce130116a653c [file]
/*
* Copyright (c) 2020 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 "video/receive_statistics_proxy.h"
#include <algorithm>
#include <cmath>
#include <cstddef>
#include <cstdint>
#include <optional>
#include <string>
#include <utility>
#include "absl/strings/string_view.h"
#include "api/rtp_packet_info.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "api/video/video_codec_type.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame.h"
#include "api/video/video_frame_type.h"
#include "api/video/video_timing.h"
#include "api/video_codecs/video_decoder.h"
#include "call/video_receive_stream.h"
#include "modules/rtp_rtcp/include/rtcp_statistics.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/thread.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/metrics.h"
#include "video/stats_counter.h"
#include "video/video_quality_observer2.h"
#include "video/video_receive_stream2.h"
namespace webrtc {
namespace internal {
namespace {
// Periodic time interval for processing samples for `freq_offset_counter_`.
const int64_t kFreqOffsetProcessIntervalMs = 40000;
// Some metrics are reported as a maximum over this period.
// This should be synchronized with a typical getStats polling interval in
// the clients.
const int kMovingMaxWindowMs = 1000;
// How large window we use to calculate the framerate/bitrate.
const int kRateStatisticsWindowSizeMs = 1000;
// Some sane ballpark estimate for maximum common value of inter-frame delay.
// Values below that will be stored explicitly in the array,
// values above - in the map.
const int kMaxCommonInterframeDelayMs = 500;
const char* UmaPrefixForContentType(VideoContentType content_type) {
if (videocontenttypehelpers::IsScreenshare(content_type))
return "WebRTC.Video.Screenshare";
return "WebRTC.Video";
}
// TODO(https://bugs.webrtc.org/11572): Workaround for an issue with some
// webrtc::Thread instances and/or implementations that don't register as the
// current task queue.
bool IsCurrentTaskQueueOrThread(TaskQueueBase* task_queue) {
if (task_queue->IsCurrent())
return true;
Thread* current_thread = ThreadManager::Instance()->CurrentThread();
if (!current_thread)
return false;
return static_cast<TaskQueueBase*>(current_thread) == task_queue;
}
} // namespace
ReceiveStatisticsProxy::ReceiveStatisticsProxy(uint32_t remote_ssrc,
Clock* clock,
TaskQueueBase* worker_thread)
: clock_(clock),
start_(clock->CurrentTime()),
remote_ssrc_(remote_ssrc),
// 1000ms window, scale 1000 for ms to s.
decode_fps_estimator_(1000, 1000),
renders_fps_estimator_(1000, 1000),
first_frame_rendered_(Timestamp::MinusInfinity()),
total_render_sqrt_pixels_(0),
video_quality_observer_(new VideoQualityObserver()),
interframe_delay_max_moving_(kMovingMaxWindowMs),
freq_offset_counter_(clock, nullptr, kFreqOffsetProcessIntervalMs),
last_content_type_(VideoContentType::UNSPECIFIED),
last_codec_type_(kVideoCodecVP8),
num_delayed_frames_rendered_(0),
timing_frame_info_counter_(kMovingMaxWindowMs),
worker_thread_(worker_thread) {
RTC_DCHECK(worker_thread);
decode_queue_.Detach();
incoming_render_queue_.Detach();
stats_.ssrc = remote_ssrc_;
}
ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
RTC_DCHECK_RUN_ON(&main_thread_);
}
void ReceiveStatisticsProxy::UpdateHistograms(
std::optional<int> fraction_lost,
const StreamDataCounters& rtp_stats,
const StreamDataCounters* rtx_stats) {
RTC_DCHECK_RUN_ON(&main_thread_);
char log_stream_buf[8 * 1024];
SimpleStringBuilder log_stream(log_stream_buf);
Timestamp now = clock_->CurrentTime();
TimeDelta stream_duration = now - start_;
if (stats_.frame_counts.key_frames > 0 ||
stats_.frame_counts.delta_frames > 0) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.ReceiveStreamLifetimeInSeconds",
stream_duration.seconds());
log_stream << "WebRTC.Video.ReceiveStreamLifetimeInSeconds "
<< stream_duration.seconds() << '\n';
}
log_stream << "Frames decoded " << stats_.frames_decoded << '\n';
if (num_unique_frames_) {
int num_dropped_frames = *num_unique_frames_ - stats_.frames_decoded;
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DroppedFrames.Receiver",
num_dropped_frames);
log_stream << "WebRTC.Video.DroppedFrames.Receiver " << num_dropped_frames
<< '\n';
}
if (fraction_lost && stream_duration >= metrics::kMinRunTime) {
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
*fraction_lost);
log_stream << "WebRTC.Video.ReceivedPacketsLostInPercent " << *fraction_lost
<< '\n';
}
if (first_decoded_frame_time_ms_) {
const int64_t elapsed_ms =
(clock_->TimeInMilliseconds() - *first_decoded_frame_time_ms_);
if (elapsed_ms >= metrics::kMinRunTime.ms()) {
int decoded_fps = static_cast<int>(
(stats_.frames_decoded * 1000.0f / elapsed_ms) + 0.5f);
RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.DecodedFramesPerSecond",
decoded_fps);
log_stream << "WebRTC.Video.DecodedFramesPerSecond " << decoded_fps
<< '\n';
const uint32_t frames_rendered = stats_.frames_rendered;
if (frames_rendered > 0) {
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DelayedFramesToRenderer",
static_cast<int>(num_delayed_frames_rendered_ *
100 / frames_rendered));
if (num_delayed_frames_rendered_ > 0) {
RTC_HISTOGRAM_COUNTS_1000(
"WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs",
(sum_missed_render_deadline_ / num_delayed_frames_rendered_)
.ms());
}
}
}
}
const int kMinRequiredSamples = 200;
TimeDelta render_duration = now - first_frame_rendered_;
if (stats_.frames_rendered >= kMinRequiredSamples &&
render_duration > TimeDelta::Zero()) {
int rendered_fps = (stats_.frames_rendered / render_duration).hertz();
RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.RenderFramesPerSecond",
rendered_fps);
log_stream << "WebRTC.Video.RenderFramesPerSecond " << rendered_fps << '\n';
RTC_HISTOGRAM_COUNTS_100000(
"WebRTC.Video.RenderSqrtPixelsPerSecond",
(total_render_sqrt_pixels_ / render_duration).hertz());
}
std::optional<int> sync_offset_ms =
sync_offset_counter_.Avg(kMinRequiredSamples);
if (sync_offset_ms) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.AVSyncOffsetInMs",
*sync_offset_ms);
log_stream << "WebRTC.Video.AVSyncOffsetInMs " << *sync_offset_ms << '\n';
}
AggregatedStats freq_offset_stats = freq_offset_counter_.GetStats();
if (freq_offset_stats.num_samples > 0) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtpToNtpFreqOffsetInKhz",
freq_offset_stats.average);
log_stream << "WebRTC.Video.RtpToNtpFreqOffsetInKhz "
<< freq_offset_stats.ToString() << '\n';
}
int num_total_frames =
stats_.frame_counts.key_frames + stats_.frame_counts.delta_frames;
if (num_total_frames >= kMinRequiredSamples) {
int num_key_frames = stats_.frame_counts.key_frames;
int key_frames_permille =
(num_key_frames * 1000 + num_total_frames / 2) / num_total_frames;
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesReceivedInPermille",
key_frames_permille);
log_stream << "WebRTC.Video.KeyFramesReceivedInPermille "
<< key_frames_permille << '\n';
}
std::optional<int> vp8_qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
if (vp8_qp) {
RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", *vp8_qp);
log_stream << "WebRTC.Video.Decoded.Vp8.Qp " << *vp8_qp << '\n';
}
std::optional<int> decode_ms = decode_time_counter_.Avg(kMinRequiredSamples);
if (decode_ms) {
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", *decode_ms);
log_stream << "WebRTC.Video.DecodeTimeInMs " << *decode_ms << '\n';
}
std::optional<int> jb_delay_ms =
jitter_delay_counter_.Avg(kMinRequiredSamples);
if (jb_delay_ms) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
*jb_delay_ms);
log_stream << "WebRTC.Video.JitterBufferDelayInMs " << *jb_delay_ms << '\n';
}
std::optional<int> target_delay_ms =
target_delay_counter_.Avg(kMinRequiredSamples);
if (target_delay_ms) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs",
*target_delay_ms);
log_stream << "WebRTC.Video.TargetDelayInMs " << *target_delay_ms << '\n';
}
std::optional<int> current_delay_ms =
current_delay_counter_.Avg(kMinRequiredSamples);
if (current_delay_ms) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
*current_delay_ms);
log_stream << "WebRTC.Video.CurrentDelayInMs " << *current_delay_ms << '\n';
}
std::optional<int> delay_ms = oneway_delay_counter_.Avg(kMinRequiredSamples);
if (delay_ms)
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", *delay_ms);
for (const auto& it : content_specific_stats_) {
// For the metric Foo we report the following slices:
// WebRTC.Video.Foo,
// WebRTC.Video.Screenshare.Foo,
auto content_type = it.first;
auto stats = it.second;
std::string uma_prefix = UmaPrefixForContentType(content_type);
std::optional<int> e2e_delay_ms =
stats.e2e_delay_counter.Avg(kMinRequiredSamples);
if (e2e_delay_ms) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix + ".EndToEndDelayInMs",
*e2e_delay_ms);
log_stream << uma_prefix << ".EndToEndDelayInMs" << " " << *e2e_delay_ms
<< '\n';
}
std::optional<int> e2e_delay_max_ms = stats.e2e_delay_counter.Max();
if (e2e_delay_max_ms && e2e_delay_ms) {
RTC_HISTOGRAM_COUNTS_SPARSE_100000(uma_prefix + ".EndToEndDelayMaxInMs",
*e2e_delay_max_ms);
log_stream << uma_prefix << ".EndToEndDelayMaxInMs" << " "
<< *e2e_delay_max_ms << '\n';
}
std::optional<int> interframe_delay_ms =
stats.interframe_delay_counter.Avg(kMinRequiredSamples);
if (interframe_delay_ms) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix + ".InterframeDelayInMs",
*interframe_delay_ms);
log_stream << uma_prefix << ".InterframeDelayInMs" << " "
<< *interframe_delay_ms << '\n';
}
std::optional<int> interframe_delay_max_ms =
stats.interframe_delay_counter.Max();
if (interframe_delay_max_ms && interframe_delay_ms) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix + ".InterframeDelayMaxInMs",
*interframe_delay_max_ms);
log_stream << uma_prefix << ".InterframeDelayMaxInMs" << " "
<< *interframe_delay_max_ms << '\n';
}
std::optional<uint32_t> interframe_delay_95p_ms =
stats.interframe_delay_percentiles.GetPercentile(0.95f);
if (interframe_delay_95p_ms && interframe_delay_ms != -1) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(
uma_prefix + ".InterframeDelay95PercentileInMs",
*interframe_delay_95p_ms);
log_stream << uma_prefix << ".InterframeDelay95PercentileInMs" << " "
<< *interframe_delay_95p_ms << '\n';
}
std::optional<int> width = stats.received_width.Avg(kMinRequiredSamples);
if (width) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix + ".ReceivedWidthInPixels",
*width);
log_stream << uma_prefix << ".ReceivedWidthInPixels" << " " << *width
<< '\n';
}
std::optional<int> height = stats.received_height.Avg(kMinRequiredSamples);
if (height) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix + ".ReceivedHeightInPixels",
*height);
log_stream << uma_prefix << ".ReceivedHeightInPixels" << " " << *height
<< '\n';
}
std::optional<double> corruption_score = stats.corruption_score.GetMean();
if (corruption_score) {
// Granularity level: 2e-3.
RTC_HISTOGRAM_COUNTS_SPARSE(uma_prefix + ".CorruptionLikelihoodPermille",
static_cast<int>(*corruption_score * 1000),
/*min=*/0, /*max=*/1000,
/*bucket_count=*/500);
log_stream << uma_prefix << ".CorruptionLikelihoodPermille" << " "
<< static_cast<int>(*corruption_score * 1000) << '\n';
}
if (content_type != VideoContentType::UNSPECIFIED) {
// Don't report these 3 metrics unsliced, as more precise variants
// are reported separately in this method.
float flow_duration_sec = stats.flow_duration_ms / 1000.0;
if (flow_duration_sec >= metrics::kMinRunTime.seconds()) {
int media_bitrate_kbps = static_cast<int>(stats.total_media_bytes * 8 /
flow_duration_sec / 1000);
RTC_HISTOGRAM_COUNTS_SPARSE_10000(
uma_prefix + ".MediaBitrateReceivedInKbps", media_bitrate_kbps);
log_stream << uma_prefix << ".MediaBitrateReceivedInKbps" << " "
<< media_bitrate_kbps << '\n';
}
num_total_frames =
stats.frame_counts.key_frames + stats.frame_counts.delta_frames;
if (num_total_frames >= kMinRequiredSamples) {
int num_key_frames = stats.frame_counts.key_frames;
int key_frames_permille =
(num_key_frames * 1000 + num_total_frames / 2) / num_total_frames;
RTC_HISTOGRAM_COUNTS_SPARSE_1000(
uma_prefix + ".KeyFramesReceivedInPermille", key_frames_permille);
log_stream << uma_prefix << ".KeyFramesReceivedInPermille" << " "
<< key_frames_permille << '\n';
}
std::optional<int> qp = stats.qp_counter.Avg(kMinRequiredSamples);
if (qp) {
RTC_HISTOGRAM_COUNTS_SPARSE_200(uma_prefix + ".Decoded.Vp8.Qp", *qp);
log_stream << uma_prefix << ".Decoded.Vp8.Qp" << " " << *qp << '\n';
}
}
}
StreamDataCounters rtp_rtx_stats = rtp_stats;
if (rtx_stats)
rtp_rtx_stats.Add(*rtx_stats);
TimeDelta elapsed = rtp_rtx_stats.TimeSinceFirstPacket(clock_->CurrentTime());
if (elapsed >= metrics::kMinRunTime) {
int64_t elapsed_sec = elapsed.seconds();
RTC_HISTOGRAM_COUNTS_10000(
"WebRTC.Video.BitrateReceivedInKbps",
static_cast<int>(rtp_rtx_stats.transmitted.TotalBytes() * 8 /
elapsed_sec / 1000));
int media_bitrate_kbs = static_cast<int>(rtp_stats.MediaPayloadBytes() * 8 /
elapsed_sec / 1000);
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateReceivedInKbps",
media_bitrate_kbs);
log_stream << "WebRTC.Video.MediaBitrateReceivedInKbps "
<< media_bitrate_kbs << '\n';
RTC_HISTOGRAM_COUNTS_10000(
"WebRTC.Video.PaddingBitrateReceivedInKbps",
static_cast<int>(rtp_rtx_stats.transmitted.padding_bytes * 8 /
elapsed_sec / 1000));
RTC_HISTOGRAM_COUNTS_10000(
"WebRTC.Video.RetransmittedBitrateReceivedInKbps",
static_cast<int>(rtp_rtx_stats.retransmitted.TotalBytes() * 8 /
elapsed_sec / 1000));
if (rtx_stats) {
RTC_HISTOGRAM_COUNTS_10000(
"WebRTC.Video.RtxBitrateReceivedInKbps",
static_cast<int>(rtx_stats->transmitted.TotalBytes() * 8 /
elapsed_sec / 1000));
}
const RtcpPacketTypeCounter& counters = stats_.rtcp_packet_type_counts;
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute",
counters.nack_packets * 60 / elapsed_sec);
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute",
counters.fir_packets * 60 / elapsed_sec);
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute",
counters.pli_packets * 60 / elapsed_sec);
if (counters.nack_requests > 0) {
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent",
counters.UniqueNackRequestsInPercent());
}
}
RTC_LOG(LS_INFO) << log_stream.str();
video_quality_observer_->UpdateHistograms(
videocontenttypehelpers::IsScreenshare(last_content_type_));
}
void ReceiveStatisticsProxy::UpdateFramerate(int64_t now_ms) const {
RTC_DCHECK_RUN_ON(&main_thread_);
int64_t old_frames_ms = now_ms - kRateStatisticsWindowSizeMs;
while (!frame_window_.empty() &&
frame_window_.begin()->first < old_frames_ms) {
frame_window_.erase(frame_window_.begin());
}
size_t framerate =
(frame_window_.size() * 1000 + 500) / kRateStatisticsWindowSizeMs;
stats_.network_frame_rate = static_cast<int>(framerate);
}
std::optional<int64_t>
ReceiveStatisticsProxy::GetCurrentEstimatedPlayoutNtpTimestampMs(
int64_t now_ms) const {
RTC_DCHECK_RUN_ON(&main_thread_);
if (!last_estimated_playout_ntp_timestamp_ms_ ||
!last_estimated_playout_time_ms_) {
return std::nullopt;
}
int64_t elapsed_ms = now_ms - *last_estimated_playout_time_ms_;
return *last_estimated_playout_ntp_timestamp_ms_ + elapsed_ms;
}
VideoReceiveStreamInterface::Stats ReceiveStatisticsProxy::GetStats() const {
RTC_DCHECK_RUN_ON(&main_thread_);
// Like VideoReceiveStreamInterface::GetStats, called on the worker thread
// from StatsCollector::ExtractMediaInfo via worker_thread()->BlockingCall().
// WebRtcVideoChannel::GetStats(), GetVideoReceiverInfo.
// Get current frame rates here, as only updating them on new frames prevents
// us from ever correctly displaying frame rate of 0.
int64_t now_ms = clock_->TimeInMilliseconds();
UpdateFramerate(now_ms);
stats_.render_frame_rate = renders_fps_estimator_.Rate(now_ms).value_or(0);
stats_.decode_frame_rate = decode_fps_estimator_.Rate(now_ms).value_or(0);
if (last_decoded_frame_time_ms_) {
// Avoid using a newer timestamp than might be pending for decoded frames.
// If we do use now_ms, we might roll the max window to a value that is
// higher than that of a decoded frame timestamp that we haven't yet
// captured the data for (i.e. pending call to OnDecodedFrame).
stats_.interframe_delay_max_ms =
interframe_delay_max_moving_.Max(*last_decoded_frame_time_ms_)
.value_or(-1);
} else {
// We're paused. Avoid changing the state of `interframe_delay_max_moving_`.
stats_.interframe_delay_max_ms = -1;
}
stats_.freeze_count = video_quality_observer_->NumFreezes();
stats_.pause_count = video_quality_observer_->NumPauses();
stats_.total_freezes_duration_ms =
video_quality_observer_->TotalFreezesDurationMs();
stats_.total_pauses_duration_ms =
video_quality_observer_->TotalPausesDurationMs();
stats_.total_inter_frame_delay =
static_cast<double>(video_quality_observer_->TotalFramesDurationMs()) /
kNumMillisecsPerSec;
stats_.total_squared_inter_frame_delay =
video_quality_observer_->SumSquaredFrameDurationsSec();
stats_.content_type = last_content_type_;
stats_.timing_frame_info = timing_frame_info_counter_.Max(now_ms);
stats_.estimated_playout_ntp_timestamp_ms =
GetCurrentEstimatedPlayoutNtpTimestampMs(now_ms);
return stats_;
}
void ReceiveStatisticsProxy::OnIncomingPayloadType(int payload_type) {
RTC_DCHECK_RUN_ON(&decode_queue_);
worker_thread_->PostTask(SafeTask(task_safety_.flag(), [payload_type, this] {
RTC_DCHECK_RUN_ON(&main_thread_);
stats_.current_payload_type = payload_type;
}));
}
void ReceiveStatisticsProxy::OnDecoderInfo(
const VideoDecoder::DecoderInfo& decoder_info) {
RTC_DCHECK_RUN_ON(&decode_queue_);
worker_thread_->PostTask(SafeTask(
task_safety_.flag(),
[this, name = decoder_info.implementation_name,
is_hardware_accelerated = decoder_info.is_hardware_accelerated]() {
RTC_DCHECK_RUN_ON(&main_thread_);
stats_.decoder_implementation_name = name;
stats_.power_efficient_decoder = is_hardware_accelerated;
}));
}
void ReceiveStatisticsProxy::OnDecodableFrame(TimeDelta jitter_buffer_delay,
TimeDelta target_delay,
TimeDelta minimum_delay) {
RTC_DCHECK_RUN_ON(&main_thread_);
// Cumulative stats exposed through standardized GetStats.
stats_.jitter_buffer_delay += jitter_buffer_delay;
stats_.jitter_buffer_target_delay += target_delay;
++stats_.jitter_buffer_emitted_count;
stats_.jitter_buffer_minimum_delay += minimum_delay;
}
void ReceiveStatisticsProxy::OnFrameBufferTimingsUpdated(
int estimated_max_decode_time_ms,
int current_delay_ms,
int target_delay_ms,
int jitter_delay_ms,
int min_playout_delay_ms,
int render_delay_ms) {
RTC_DCHECK_RUN_ON(&main_thread_);
// Instantaneous stats exposed through legacy GetStats.
stats_.max_decode_ms = estimated_max_decode_time_ms;
stats_.current_delay_ms = current_delay_ms;
stats_.target_delay_ms = target_delay_ms;
stats_.jitter_buffer_ms = jitter_delay_ms;
stats_.min_playout_delay_ms = min_playout_delay_ms;
stats_.render_delay_ms = render_delay_ms;
// UMA stats.
jitter_delay_counter_.Add(jitter_delay_ms);
target_delay_counter_.Add(target_delay_ms);
current_delay_counter_.Add(current_delay_ms);
// Estimated one-way delay: network delay (rtt/2) + target_delay_ms (jitter
// delay + decode time + render delay).
oneway_delay_counter_.Add(target_delay_ms + avg_rtt_ms_ / 2);
}
void ReceiveStatisticsProxy::OnUniqueFramesCounted(int num_unique_frames) {
RTC_DCHECK_RUN_ON(&main_thread_);
num_unique_frames_.emplace(num_unique_frames);
}
void ReceiveStatisticsProxy::OnTimingFrameInfoUpdated(
const TimingFrameInfo& info) {
RTC_DCHECK_RUN_ON(&main_thread_);
if (info.flags != VideoSendTiming::kInvalid) {
int64_t now_ms = clock_->TimeInMilliseconds();
timing_frame_info_counter_.Add(info, now_ms);
}
// Measure initial decoding latency between the first frame arriving and
// the first frame being decoded.
if (!first_frame_received_time_ms_.has_value()) {
first_frame_received_time_ms_ = info.receive_finish_ms;
}
if (stats_.first_frame_received_to_decoded_ms == -1 &&
first_decoded_frame_time_ms_) {
stats_.first_frame_received_to_decoded_ms =
*first_decoded_frame_time_ms_ - *first_frame_received_time_ms_;
}
}
void ReceiveStatisticsProxy::RtcpPacketTypesCounterUpdated(
uint32_t ssrc,
const RtcpPacketTypeCounter& packet_counter) {
if (ssrc != remote_ssrc_)
return;
if (!IsCurrentTaskQueueOrThread(worker_thread_)) {
// RtpRtcpInterface::Configuration has a single
// RtcpPacketTypeCounterObserver and that same configuration may be used for
// both receiver and sender (see ModuleRtpRtcpImpl::ModuleRtpRtcpImpl). The
// RTCPSender implementation currently makes calls to this function on a
// process thread whereas the RTCPReceiver implementation calls back on the
// [main] worker thread.
// So until the sender implementation has been updated, we work around this
// here by posting the update to the expected thread. We make a by value
// copy of the `task_safety_` to handle the case if the queued task
// runs after the `ReceiveStatisticsProxy` has been deleted. In such a
// case the packet_counter update won't be recorded.
worker_thread_->PostTask(
SafeTask(task_safety_.flag(), [ssrc, packet_counter, this]() {
RtcpPacketTypesCounterUpdated(ssrc, packet_counter);
}));
return;
}
RTC_DCHECK_RUN_ON(&main_thread_);
stats_.rtcp_packet_type_counts = packet_counter;
}
void ReceiveStatisticsProxy::OnCname(uint32_t ssrc, absl::string_view cname) {
RTC_DCHECK_RUN_ON(&main_thread_);
// TODO(pbos): Handle both local and remote ssrcs here and RTC_DCHECK that we
// receive stats from one of them.
if (remote_ssrc_ != ssrc)
return;
stats_.c_name = std::string(cname);
}
void ReceiveStatisticsProxy::OnDecodedFrame(const VideoFrame& frame,
std::optional<uint8_t> qp,
TimeDelta decode_time,
VideoContentType content_type,
VideoFrameType frame_type) {
TimeDelta processing_delay = TimeDelta::Zero();
Timestamp current_time = clock_->CurrentTime();
// TODO(bugs.webrtc.org/13984): some tests do not fill packet_infos().
TimeDelta assembly_time = TimeDelta::Zero();
if (!frame.packet_infos().empty()) {
const auto [first_packet, last_packet] = std::minmax_element(
frame.packet_infos().cbegin(), frame.packet_infos().cend(),
[](const webrtc::RtpPacketInfo& a, const webrtc::RtpPacketInfo& b) {
return a.receive_time() < b.receive_time();
});
if (first_packet->receive_time().IsFinite()) {
processing_delay = current_time - first_packet->receive_time();
// Extract frame assembly time (i.e. time between earliest and latest
// packet arrival). Note: for single-packet frames this will be 0.
assembly_time =
last_packet->receive_time() - first_packet->receive_time();
}
}
// See VCMDecodedFrameCallback::Decoded for more info on what thread/queue we
// may be on. E.g. on iOS this gets called on
// "com.apple.coremedia.decompressionsession.clientcallback"
VideoFrameMetaData meta(frame, current_time);
worker_thread_->PostTask(SafeTask(
task_safety_.flag(), [meta, qp, decode_time, processing_delay,
assembly_time, content_type, frame_type, this]() {
OnDecodedFrame(meta, qp, decode_time, processing_delay, assembly_time,
content_type, frame_type);
}));
}
void ReceiveStatisticsProxy::OnDecodedFrame(
const VideoFrameMetaData& frame_meta,
std::optional<uint8_t> qp,
TimeDelta decode_time,
TimeDelta processing_delay,
TimeDelta assembly_time,
VideoContentType content_type,
VideoFrameType frame_type) {
RTC_DCHECK_RUN_ON(&main_thread_);
const bool is_screenshare =
videocontenttypehelpers::IsScreenshare(content_type);
const bool was_screenshare =
videocontenttypehelpers::IsScreenshare(last_content_type_);
if (is_screenshare != was_screenshare) {
// Reset the quality observer if content type is switched. But first report
// stats for the previous part of the call.
video_quality_observer_->UpdateHistograms(was_screenshare);
video_quality_observer_.reset(new VideoQualityObserver());
}
video_quality_observer_->OnDecodedFrame(frame_meta.rtp_timestamp, qp,
last_codec_type_);
ContentSpecificStats* content_specific_stats =
&content_specific_stats_[content_type];
++stats_.frames_decoded;
if (qp) {
if (!stats_.qp_sum) {
if (stats_.frames_decoded != 1) {
RTC_LOG(LS_WARNING)
<< "Frames decoded was not 1 when first qp value was received.";
}
stats_.qp_sum = 0;
}
*stats_.qp_sum += *qp;
content_specific_stats->qp_counter.Add(*qp);
} else if (stats_.qp_sum) {
RTC_LOG(LS_WARNING)
<< "QP sum was already set and no QP was given for a frame.";
stats_.qp_sum.reset();
}
decode_time_counter_.Add(decode_time.ms());
stats_.decode_ms = decode_time.ms();
stats_.total_decode_time += decode_time;
stats_.total_processing_delay += processing_delay;
stats_.total_assembly_time += assembly_time;
if (!assembly_time.IsZero()) {
++stats_.frames_assembled_from_multiple_packets;
}
last_content_type_ = content_type;
decode_fps_estimator_.Update(1, frame_meta.decode_timestamp.ms());
if (last_decoded_frame_time_ms_) {
int64_t interframe_delay_ms =
frame_meta.decode_timestamp.ms() - *last_decoded_frame_time_ms_;
RTC_DCHECK_GE(interframe_delay_ms, 0);
interframe_delay_max_moving_.Add(interframe_delay_ms,
frame_meta.decode_timestamp.ms());
content_specific_stats->interframe_delay_counter.Add(interframe_delay_ms);
content_specific_stats->interframe_delay_percentiles.Add(
interframe_delay_ms);
content_specific_stats->flow_duration_ms += interframe_delay_ms;
}
if (stats_.frames_decoded == 1) {
first_decoded_frame_time_ms_.emplace(frame_meta.decode_timestamp.ms());
}
last_decoded_frame_time_ms_.emplace(frame_meta.decode_timestamp.ms());
}
void ReceiveStatisticsProxy::OnRenderedFrame(
const VideoFrameMetaData& frame_meta) {
RTC_DCHECK_RUN_ON(&main_thread_);
// Called from VideoReceiveStream2::OnFrame.
RTC_DCHECK_GT(frame_meta.width, 0);
RTC_DCHECK_GT(frame_meta.height, 0);
video_quality_observer_->OnRenderedFrame(frame_meta);
ContentSpecificStats* content_specific_stats =
&content_specific_stats_[last_content_type_];
renders_fps_estimator_.Update(1, frame_meta.decode_timestamp.ms());
if (stats_.frames_rendered == 0) {
first_frame_rendered_ = clock_->CurrentTime();
}
++stats_.frames_rendered;
stats_.width = frame_meta.width;
stats_.height = frame_meta.height;
total_render_sqrt_pixels_ += sqrt(frame_meta.width * frame_meta.height);
content_specific_stats->received_width.Add(frame_meta.width);
content_specific_stats->received_height.Add(frame_meta.height);
// Consider taking stats_.render_delay_ms into account.
const TimeDelta time_until_rendering =
frame_meta.render_time - frame_meta.decode_timestamp;
if (time_until_rendering < TimeDelta::Zero()) {
sum_missed_render_deadline_ += -time_until_rendering;
++num_delayed_frames_rendered_;
}
if (frame_meta.ntp_time_ms > 0) {
int64_t delay_ms =
clock_->CurrentNtpInMilliseconds() - frame_meta.ntp_time_ms;
if (delay_ms >= 0) {
content_specific_stats->e2e_delay_counter.Add(delay_ms);
}
}
}
void ReceiveStatisticsProxy::OnSyncOffsetUpdated(int64_t video_playout_ntp_ms,
int64_t sync_offset_ms,
double estimated_freq_khz) {
RTC_DCHECK_RUN_ON(&main_thread_);
const int64_t now_ms = clock_->TimeInMilliseconds();
sync_offset_counter_.Add(std::abs(sync_offset_ms));
stats_.sync_offset_ms = sync_offset_ms;
last_estimated_playout_ntp_timestamp_ms_ = video_playout_ntp_ms;
last_estimated_playout_time_ms_ = now_ms;
const double kMaxFreqKhz = 10000.0;
int offset_khz = kMaxFreqKhz;
// Should not be zero or negative. If so, report max.
if (estimated_freq_khz < kMaxFreqKhz && estimated_freq_khz > 0.0)
offset_khz = static_cast<int>(std::fabs(estimated_freq_khz - 90.0) + 0.5);
freq_offset_counter_.Add(offset_khz);
}
void ReceiveStatisticsProxy::OnCompleteFrame(bool is_keyframe,
size_t size_bytes,
VideoContentType content_type) {
RTC_DCHECK_RUN_ON(&main_thread_);
if (is_keyframe) {
++stats_.frame_counts.key_frames;
} else {
++stats_.frame_counts.delta_frames;
}
// Content type extension is set only for keyframes and should be propagated
// for all the following delta frames. Here we may receive frames out of order
// and miscategorise some delta frames near the layer switch.
// This may slightly offset calculated bitrate and keyframes permille metrics.
VideoContentType propagated_content_type =
is_keyframe ? content_type : last_content_type_;
ContentSpecificStats* content_specific_stats =
&content_specific_stats_[propagated_content_type];
content_specific_stats->total_media_bytes += size_bytes;
if (is_keyframe) {
++content_specific_stats->frame_counts.key_frames;
} else {
++content_specific_stats->frame_counts.delta_frames;
}
int64_t now_ms = clock_->TimeInMilliseconds();
frame_window_.insert(std::make_pair(now_ms, size_bytes));
UpdateFramerate(now_ms);
}
void ReceiveStatisticsProxy::OnDroppedFrames(uint32_t frames_dropped) {
// Can be called on either the decode queue or the worker thread
// See FrameBuffer2 for more details.
worker_thread_->PostTask(
SafeTask(task_safety_.flag(), [frames_dropped, this]() {
RTC_DCHECK_RUN_ON(&main_thread_);
stats_.frames_dropped += frames_dropped;
}));
}
void ReceiveStatisticsProxy::OnPreDecode(VideoCodecType codec_type, int qp) {
RTC_DCHECK_RUN_ON(&main_thread_);
last_codec_type_ = codec_type;
if (last_codec_type_ == kVideoCodecVP8 && qp != -1) {
qp_counters_.vp8.Add(qp);
}
}
void ReceiveStatisticsProxy::OnStreamInactive() {
RTC_DCHECK_RUN_ON(&main_thread_);
// TODO(sprang): Figure out any other state that should be reset.
// Don't report inter-frame delay if stream was paused.
last_decoded_frame_time_ms_.reset();
video_quality_observer_->OnStreamInactive();
}
void ReceiveStatisticsProxy::OnRttUpdate(int64_t avg_rtt_ms) {
RTC_DCHECK_RUN_ON(&main_thread_);
avg_rtt_ms_ = avg_rtt_ms;
}
void ReceiveStatisticsProxy::OnCorruptionScore(double corruption_score,
VideoContentType content_type) {
worker_thread_->PostTask(SafeTask(task_safety_.flag(), [corruption_score,
content_type, this] {
RTC_DCHECK_RUN_ON(&main_thread_);
if (!stats_.corruption_score_sum.has_value()) {
RTC_DCHECK(!stats_.corruption_score_squared_sum.has_value());
RTC_DCHECK_EQ(stats_.corruption_score_count, 0);
stats_.corruption_score_sum = 0;
stats_.corruption_score_squared_sum = 0;
}
*stats_.corruption_score_sum += corruption_score;
*stats_.corruption_score_squared_sum += corruption_score * corruption_score;
++stats_.corruption_score_count;
ContentSpecificStats* content_specific_stats =
&content_specific_stats_[content_type];
content_specific_stats->corruption_score.AddSample(corruption_score);
}));
}
void ReceiveStatisticsProxy::DecoderThreadStarting() {
RTC_DCHECK_RUN_ON(&main_thread_);
}
void ReceiveStatisticsProxy::DecoderThreadStopped() {
RTC_DCHECK_RUN_ON(&main_thread_);
decode_queue_.Detach();
}
ReceiveStatisticsProxy::ContentSpecificStats::ContentSpecificStats()
: interframe_delay_percentiles(kMaxCommonInterframeDelayMs) {}
ReceiveStatisticsProxy::ContentSpecificStats::~ContentSpecificStats() = default;
void ReceiveStatisticsProxy::ContentSpecificStats::Add(
const ContentSpecificStats& other) {
e2e_delay_counter.Add(other.e2e_delay_counter);
interframe_delay_counter.Add(other.interframe_delay_counter);
flow_duration_ms += other.flow_duration_ms;
total_media_bytes += other.total_media_bytes;
received_height.Add(other.received_height);
received_width.Add(other.received_width);
qp_counter.Add(other.qp_counter);
frame_counts.key_frames += other.frame_counts.key_frames;
frame_counts.delta_frames += other.frame_counts.delta_frames;
interframe_delay_percentiles.Add(other.interframe_delay_percentiles);
corruption_score.MergeStatistics(other.corruption_score);
}
} // namespace internal
} // namespace webrtc