Files
Michael Froman 06e90addc7 Bug 1952339 - Vendor libwebrtc from 995688c8e8
We already cherry-picked this when we vendored f30c044cf9.

Upstream commit: https://webrtc.googlesource.com/src/+/995688c8e85b520d50961486abbe0cc03eae9558
    Revert "more p2p cleanups"

    This reverts commit f30c044cf9bd06f91017c171d98690094ce6d88b.

    Reason for revert: breaks roll to chromium: https://ci.chromium.org/ui/p/chromium/builders/try/mac_chromium_compile_dbg_ng/2290104/overview

    Original change's description:
    > more p2p cleanups
    >
    > Move test code from p2p/base and rtc_base/ into p2p/test/
    > This p2p/base much less crowded and
    > clarifies that the rtc_base/nat* is in fact only test code.
    >
    > BUG=webrtc:0
    >
    > Change-Id: I4d14fae24cb0eff6783962f4b4483b560367ca5d
    > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378900
    > Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
    > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
    > Auto-Submit: Jonas Oreland <jonaso@webrtc.org>
    > Cr-Commit-Position: refs/heads/main@{#43995}

    Bug: webrtc:0
    Change-Id: I6c79fa85f53fdb9a1dacbe38911771f9a4289c76
    No-Presubmit: true
    No-Tree-Checks: true
    No-Try: true
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/379040
    Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
    Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
    Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#43999}

Differential Revision: https://phabricator.services.mozilla.com/D244009
2025-03-07 16:59:33 -06:00

222 lines
9.9 KiB
Diff

From: Andreas Pehrson <apehrson@mozilla.com>
Date: Thu, 6 Jan 2022 00:16:00 +0000
Subject: Bug 1748458 - Add TRACE_EVENTs for dropped frames and packets for
received video. r=bwc
This lets us see in the profiler how many received frames and packets we decide
to drop and the reasons why.
Differential Revision: https://phabricator.services.mozilla.com/D135062
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/08e252da94c4752eccfd845eef13d8517953cc6a
Also includes:
Bug 1804288 - (fix-de7ae5755b) reimplement Bug 1748458 - Add TRACE_EVENTs for dropped frames and packets for received video. r=pehrsons
Differential Revision: https://phabricator.services.mozilla.com/D163960
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/8e9a326a99cd5eaa6e447ff57c01ad9d79a09744
---
api/video/frame_buffer.cc | 33 +++++++++++++++++++++++++
video/receive_statistics_proxy.cc | 11 +++++++++
video/rtp_video_stream_receiver2.cc | 4 +++
video/video_stream_buffer_controller.cc | 7 ++++++
4 files changed, 55 insertions(+)
diff --git a/api/video/frame_buffer.cc b/api/video/frame_buffer.cc
index 4981acf0ca..f0c48f3702 100644
--- a/api/video/frame_buffer.cc
+++ b/api/video/frame_buffer.cc
@@ -25,6 +25,7 @@
#include "api/video/encoded_frame.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/sequence_number_util.h"
+#include "rtc_base/trace_event.h"
namespace webrtc {
namespace {
@@ -77,7 +78,12 @@ FrameBuffer::FrameBuffer(int max_size,
decoded_frame_history_(max_decode_history) {}
bool FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
+ const uint32_t ssrc =
+ frame->PacketInfos().empty() ? 0 : frame->PacketInfos()[0].ssrc();
if (!ValidReferences(*frame)) {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frame dropped (Invalid references)",
+ "remote_ssrc", ssrc, "frame_id", frame->Id());
RTC_DLOG(LS_WARNING) << "Frame " << frame->Id()
<< " has invalid references, dropping frame.";
return false;
@@ -87,23 +93,35 @@ bool FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
if (legacy_frame_id_jump_behavior_ && frame->is_keyframe() &&
AheadOf(frame->RtpTimestamp(),
*decoded_frame_history_.GetLastDecodedFrameTimestamp())) {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frames dropped (OOO + PicId jump)",
+ "remote_ssrc", ssrc, "frame_id", frame->Id());
RTC_DLOG(LS_WARNING)
<< "Keyframe " << frame->Id()
<< " has newer timestamp but older picture id, clearing buffer.";
Clear();
} else {
// Already decoded past this frame.
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frame dropped (Out of order)",
+ "remote_ssrc", ssrc, "frame_id", frame->Id());
return false;
}
}
if (frames_.size() == max_size_) {
if (frame->is_keyframe()) {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frames dropped (KF + Full buffer)",
+ "remote_ssrc", ssrc, "frame_id", frame->Id());
RTC_DLOG(LS_WARNING) << "Keyframe " << frame->Id()
<< " inserted into full buffer, clearing buffer.";
Clear();
} else {
// No space for this frame.
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frame dropped (Full buffer)",
+ "remote_ssrc", ssrc, "frame_id", frame->Id());
return false;
}
}
@@ -158,16 +176,31 @@ void FrameBuffer::DropNextDecodableTemporalUnit() {
void FrameBuffer::UpdateDroppedFramesAndDiscardedPackets(FrameIterator begin_it,
FrameIterator end_it) {
+ uint32_t dropped_ssrc = 0;
+ int64_t dropped_frame_id = 0;
unsigned int num_discarded_packets = 0;
unsigned int num_dropped_frames =
std::count_if(begin_it, end_it, [&](const auto& f) {
if (f.second.encoded_frame) {
const auto& packetInfos = f.second.encoded_frame->PacketInfos();
+ dropped_frame_id = f.first;
+ if (!packetInfos.empty()) {
+ dropped_ssrc = packetInfos[0].ssrc();
+ }
num_discarded_packets += packetInfos.size();
}
return f.second.encoded_frame != nullptr;
});
+ if (num_dropped_frames > 0) {
+ TRACE_EVENT2("webrtc", "FrameBuffer Dropping Old Frames", "remote_ssrc",
+ dropped_ssrc, "frame_id", dropped_frame_id);
+ }
+ if (num_discarded_packets > 0) {
+ TRACE_EVENT2("webrtc", "FrameBuffer Discarding Old Packets", "remote_ssrc",
+ dropped_ssrc, "frame_id", dropped_frame_id);
+ }
+
num_dropped_frames_ += num_dropped_frames;
num_discarded_packets_ += num_discarded_packets;
}
diff --git a/video/receive_statistics_proxy.cc b/video/receive_statistics_proxy.cc
index 8fbaf0ac0e..7924ef2594 100644
--- a/video/receive_statistics_proxy.cc
+++ b/video/receive_statistics_proxy.cc
@@ -20,6 +20,7 @@
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/thread.h"
#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/metrics.h"
#include "video/video_receive_stream2.h"
@@ -766,6 +767,9 @@ void ReceiveStatisticsProxy::OnCompleteFrame(bool is_keyframe,
VideoContentType content_type) {
RTC_DCHECK_RUN_ON(&main_thread_);
+ TRACE_EVENT2("webrtc", "ReceiveStatisticsProxy::OnCompleteFrame",
+ "remote_ssrc", remote_ssrc_, "is_keyframe", is_keyframe);
+
// 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.
@@ -791,6 +795,8 @@ void ReceiveStatisticsProxy::OnCompleteFrame(bool is_keyframe,
void ReceiveStatisticsProxy::OnDroppedFrames(uint32_t frames_dropped) {
// Can be called on either the decode queue or the worker thread
// See FrameBuffer2 for more details.
+ TRACE_EVENT2("webrtc", "ReceiveStatisticsProxy::OnDroppedFrames",
+ "remote_ssrc", remote_ssrc_, "frames_dropped", frames_dropped);
worker_thread_->PostTask(
SafeTask(task_safety_.flag(), [frames_dropped, this]() {
RTC_DCHECK_RUN_ON(&main_thread_);
@@ -801,6 +807,9 @@ void ReceiveStatisticsProxy::OnDroppedFrames(uint32_t frames_dropped) {
void ReceiveStatisticsProxy::OnDiscardedPackets(uint32_t packets_discarded) {
// Can be called on either the decode queue or the worker thread
// See FrameBuffer2 for more details.
+ TRACE_EVENT2("webrtc", "ReceiveStatisticsProxy::OnDiscardedPackets",
+ "remote_ssrc", remote_ssrc_, "packets_discarded",
+ packets_discarded);
worker_thread_->PostTask(
SafeTask(task_safety_.flag(), [packets_discarded, this]() {
RTC_DCHECK_RUN_ON(&main_thread_);
@@ -829,6 +838,8 @@ void ReceiveStatisticsProxy::OnStreamInactive() {
void ReceiveStatisticsProxy::OnRttUpdate(int64_t avg_rtt_ms) {
RTC_DCHECK_RUN_ON(&main_thread_);
+ TRACE_EVENT2("webrtc", "ReceiveStatisticsProxy::OnRttUpdate",
+ "remote_ssrc", remote_ssrc_, "avg_rtt_ms", avg_rtt_ms);
avg_rtt_ms_ = avg_rtt_ms;
}
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index 73a8da11aa..5bf7f51c8e 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -49,6 +49,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/trace_event.h"
#include "system_wrappers/include/metrics.h"
#include "system_wrappers/include/ntp_time.h"
@@ -1325,6 +1326,9 @@ void RtpVideoStreamReceiver2::FrameDecoded(int64_t picture_id) {
packet_infos_.upper_bound(unwrapped_rtp_seq_num));
uint32_t num_packets_cleared = packet_buffer_.ClearTo(seq_num);
if (num_packets_cleared > 0) {
+ TRACE_EVENT2("webrtc",
+ "RtpVideoStreamReceiver2::FrameDecoded Cleared Old Packets",
+ "remote_ssrc", config_.rtp.remote_ssrc, "seq_num", seq_num);
vcm_receive_statistics_->OnDiscardedPackets(num_packets_cleared);
}
reference_finder_->ClearTo(seq_num);
diff --git a/video/video_stream_buffer_controller.cc b/video/video_stream_buffer_controller.cc
index 01063b12f7..e12ce75ba0 100644
--- a/video/video_stream_buffer_controller.cc
+++ b/video/video_stream_buffer_controller.cc
@@ -31,6 +31,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/thread_annotations.h"
+#include "rtc_base/trace_event.h"
#include "video/frame_decode_scheduler.h"
#include "video/frame_decode_timing.h"
#include "video/task_queue_frame_decode_scheduler.h"
@@ -152,6 +153,9 @@ std::optional<int64_t> VideoStreamBufferController::InsertFrame(
std::unique_ptr<EncodedFrame> frame) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
FrameMetadata metadata(*frame);
+ const uint32_t ssrc =
+ frame->PacketInfos().empty() ? 0 : frame->PacketInfos()[0].ssrc();
+ const int64_t frameId = frame->Id();
int complete_units = buffer_->GetTotalNumberOfContinuousTemporalUnits();
if (buffer_->InsertFrame(std::move(frame))) {
RTC_DCHECK(metadata.receive_time) << "Frame receive time must be set!";
@@ -162,6 +166,9 @@ std::optional<int64_t> VideoStreamBufferController::InsertFrame(
*metadata.receive_time);
}
if (complete_units < buffer_->GetTotalNumberOfContinuousTemporalUnits()) {
+ TRACE_EVENT2("webrtc",
+ "VideoStreamBufferController::InsertFrame Frame Complete",
+ "remote_ssrc", ssrc, "frame_id", frameId);
stats_proxy_->OnCompleteFrame(metadata.is_keyframe, metadata.size,
metadata.contentType);
MaybeScheduleFrameForRelease();