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Add periodic logging of sync delays.

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Bug: none
Change-Id: Ib2371651c7a912231c93742410a8aa1b01cc9896
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168344
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30495}
This commit is contained in:
Åsa Persson
2020-02-10 16:33:29 +01:00
committed by Commit Bot
parent c43fe2efd6
commit 74d2b1ded5
9 changed files with 48 additions and 14 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
audio/audio_receive_stream.cc
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@ -295,7 +295,7 @@ int AudioReceiveStream::PreferredSampleRate() const {
return channel_receive_->PreferredSampleRate();
}
int AudioReceiveStream::id() const {
uint32_t AudioReceiveStream::id() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return config_.rtp.remote_ssrc;
}

2
audio/audio_receive_stream.h
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@ -87,7 +87,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
int PreferredSampleRate() const override;
// Syncable
int id() const override;
uint32_t id() const override;
absl::optional<Syncable::Info> GetInfo() const override;
bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
int64_t* time_ms) const override;

2
call/syncable.h
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@ -33,7 +33,7 @@ class Syncable {
virtual ~Syncable();
virtual int id() const = 0;
virtual uint32_t id() const = 0;
virtual absl::optional<Info> GetInfo() const = 0;
virtual bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
int64_t* time_ms) const = 0;

32
video/rtp_streams_synchronizer.cc
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@ -13,12 +13,16 @@
#include "absl/types/optional.h"
#include "call/syncable.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/rtp_to_ntp_estimator.h"
namespace webrtc {
namespace {
// Time interval for logging stats.
constexpr int64_t kStatsLogIntervalMs = 10000;
bool UpdateMeasurements(StreamSynchronization::Measurements* stream,
const Syncable::Info& info) {
RTC_DCHECK(stream);
@ -38,7 +42,8 @@ RtpStreamsSynchronizer::RtpStreamsSynchronizer(Syncable* syncable_video)
: syncable_video_(syncable_video),
syncable_audio_(nullptr),
sync_(),
last_sync_time_(rtc::TimeNanos()) {
last_sync_time_(rtc::TimeNanos()),
last_stats_log_ms_(rtc::TimeMillis()) {
RTC_DCHECK(syncable_video);
process_thread_checker_.Detach();
}
@ -77,6 +82,13 @@ void RtpStreamsSynchronizer::Process() {
}
RTC_DCHECK(sync_.get());
bool log_stats = false;
const int64_t now_ms = rtc::TimeMillis();
if (now_ms - last_stats_log_ms_ > kStatsLogIntervalMs) {
last_stats_log_ms_ = now_ms;
log_stats = true;
}
absl::optional<Syncable::Info> audio_info = syncable_audio_->GetInfo();
if (!audio_info || !UpdateMeasurements(&audio_measurement_, *audio_info)) {
return;
@ -100,11 +112,21 @@ void RtpStreamsSynchronizer::Process() {
return;
}
if (log_stats) {
RTC_LOG(LS_INFO) << "Sync info stats: " << now_ms
<< ", {ssrc: " << sync_->audio_stream_id() << ", "
<< "cur_delay_ms: " << audio_info->current_delay_ms
<< "} {ssrc: " << sync_->video_stream_id() << ", "
<< "cur_delay_ms: " << video_info->current_delay_ms
<< "} {relative_delay_ms: " << relative_delay_ms << "} ";
}
TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay",
video_info->current_delay_ms);
TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay",
audio_info->current_delay_ms);
TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
int target_audio_delay_ms = 0;
int target_video_delay_ms = video_info->current_delay_ms;
// Calculate the necessary extra audio delay and desired total video
@ -114,6 +136,14 @@ void RtpStreamsSynchronizer::Process() {
return;
}
if (log_stats) {
RTC_LOG(LS_INFO) << "Sync delay stats: " << now_ms
<< ", {ssrc: " << sync_->audio_stream_id() << ", "
<< "target_delay_ms: " << target_audio_delay_ms
<< "} {ssrc: " << sync_->video_stream_id() << ", "
<< "target_delay_ms: " << target_video_delay_ms << "} ";
}
syncable_audio_->SetMinimumPlayoutDelay(target_audio_delay_ms);
syncable_video_->SetMinimumPlayoutDelay(target_video_delay_ms);
}

5
video/rtp_streams_synchronizer.h
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
// RtpStreamsSynchronizer is responsible for synchronization audio and video for
// a given voice engine channel and video receive stream.
// RtpStreamsSynchronizer is responsible for synchronizing audio and video for
// a given audio receive stream and video receive stream.
#ifndef VIDEO_RTP_STREAMS_SYNCHRONIZER_H_
#define VIDEO_RTP_STREAMS_SYNCHRONIZER_H_
@ -58,6 +58,7 @@ class RtpStreamsSynchronizer : public Module {
rtc::ThreadChecker process_thread_checker_;
int64_t last_sync_time_ RTC_GUARDED_BY(&process_thread_checker_);
int64_t last_stats_log_ms_ RTC_GUARDED_BY(&process_thread_checker_);
};
} // namespace webrtc

6
video/stream_synchronization.cc
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@ -24,8 +24,8 @@ static const int kFilterLength = 4;
// Minimum difference between audio and video to warrant a change.
static const int kMinDeltaMs = 30;
StreamSynchronization::StreamSynchronization(int video_stream_id,
int audio_stream_id)
StreamSynchronization::StreamSynchronization(uint32_t video_stream_id,
uint32_t audio_stream_id)
: video_stream_id_(video_stream_id),
audio_stream_id_(audio_stream_id),
base_target_delay_ms_(0),
@ -53,6 +53,7 @@ bool StreamSynchronization::ComputeRelativeDelay(
video_measurement.latest_receive_time_ms -
audio_measurement.latest_receive_time_ms -
(video_last_capture_time_ms - audio_last_capture_time_ms);
if (*relative_delay_ms > kMaxDeltaDelayMs ||
*relative_delay_ms < -kMaxDeltaDelayMs) {
return false;
@ -177,7 +178,6 @@ void StreamSynchronization::SetTargetBufferingDelay(int target_delay_ms) {
// The video delay is compared to the last value (and how much we can update
// is limited by that as well).
video_delay_.last_ms += target_delay_ms - base_target_delay_ms_;
video_delay_.extra_ms += target_delay_ms - base_target_delay_ms_;
// Video is already delayed by the desired amount.

9
video/stream_synchronization.h
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@ -26,7 +26,7 @@ class StreamSynchronization {
uint32_t latest_timestamp;
};
StreamSynchronization(int video_stream_id, int audio_stream_id);
StreamSynchronization(uint32_t video_stream_id, uint32_t audio_stream_id);
bool ComputeDelays(int relative_delay_ms,
int current_audio_delay_ms,
@ -44,16 +44,19 @@ class StreamSynchronization {
// |target_delay_ms|.
void SetTargetBufferingDelay(int target_delay_ms);
uint32_t audio_stream_id() const { return audio_stream_id_; }
uint32_t video_stream_id() const { return video_stream_id_; }
private:
struct SynchronizationDelays {
int extra_ms = 0;
int last_ms = 0;
};
const uint32_t video_stream_id_;
const uint32_t audio_stream_id_;
SynchronizationDelays audio_delay_;
SynchronizationDelays video_delay_;
const int video_stream_id_;
const int audio_stream_id_;
int base_target_delay_ms_;
int avg_diff_ms_;
};

2
video/video_receive_stream.cc
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@ -580,7 +580,7 @@ void VideoReceiveStream::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
rtp_video_stream_receiver_.UpdateRtt(max_rtt_ms);
}
int VideoReceiveStream::id() const {
uint32_t VideoReceiveStream::id() const {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
return config_.rtp.remote_ssrc;
}

2
video/video_receive_stream.h
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@ -117,7 +117,7 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
// Implements Syncable.
int id() const override;
uint32_t id() const override;
absl::optional<Syncable::Info> GetInfo() const override;
bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
int64_t* time_ms) const override;