webrtc_m130/video/send_statistics_proxy.h

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/*
* Copyright (c) 2013 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.
*/
#ifndef VIDEO_SEND_STATISTICS_PROXY_H_
#define VIDEO_SEND_STATISTICS_PROXY_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "common_types.h" // NOLINT(build/include)
#include "modules/video_coding/include/video_codec_interface.h"
#include "modules/video_coding/include/video_coding_defines.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/numerics/exp_filter.h"
#include "rtc_base/ratetracker.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/clock.h"
#include "video/overuse_frame_detector.h"
#include "video/report_block_stats.h"
#include "video/stats_counter.h"
#include "video/video_stream_encoder.h"
#include "call/video_send_stream.h"
namespace webrtc {
class SendStatisticsProxy : public CpuOveruseMetricsObserver,
public RtcpStatisticsCallback,
public RtcpPacketTypeCounterObserver,
public StreamDataCountersCallback,
public BitrateStatisticsObserver,
public FrameCountObserver,
public SendSideDelayObserver {
public:
static const int kStatsTimeoutMs;
// Number of required samples to be collected before a metric is added
// to a rtc histogram.
static const int kMinRequiredMetricsSamples = 200;
SendStatisticsProxy(Clock* clock,
const VideoSendStream::Config& config,
VideoEncoderConfig::ContentType content_type);
virtual ~SendStatisticsProxy();
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ ) Reason for revert: Seem to be a flaky test rather than an issue with this cl. Creating reland, will add code to reduce flakiness to that test. Original issue's description: > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ ) > > Reason for revert: > This has resulted in failure of CallPerfTest.ReceivesCpuOveruseAndUnderuse test on the Win7 build bot https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1780 > > Original issue's description: > > Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ ) > > > > Reason for revert: > > Found issue with test case, will add fix to reland cl. > > > > Original issue's description: > > > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ ) > > > > > > Reason for revert: > > > Breaks perf tests: > > > https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679 > > > https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325 > > > > > > Original issue's description: > > > > Add framerate to VideoSinkWants and ability to signal on overuse > > > > > > > > In ViEEncoder, try to reduce framerate instead of resolution if the > > > > current degradation preference is maintain-resolution rather than > > > > balanced. > > > > > > > > BUG=webrtc:4172 > > > > > > > > Review-Url: https://codereview.webrtc.org/2716643002 > > > > Cr-Commit-Position: refs/heads/master@{#17327} > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/72acf2526177bb4dbb5103cd6e165eb4361a5ae6 > > > > > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org > > > # Skipping CQ checks because original CL landed less than 1 days ago. > > > NOPRESUBMIT=true > > > NOTREECHECKS=true > > > NOTRY=true > > > BUG=webrtc:4172 > > > > > > Review-Url: https://codereview.webrtc.org/2764133002 > > > Cr-Commit-Position: refs/heads/master@{#17331} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/8b45b11144c968b4173215c76f78c710c9a2ed0b > > > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org > > # Not skipping CQ checks because original CL landed more than 1 days ago. > > BUG=webrtc:4172 > > > > Review-Url: https://codereview.webrtc.org/2781433002 > > Cr-Commit-Position: refs/heads/master@{#17474} > > Committed: https://chromium.googlesource.com/external/webrtc/+/3ea3c77e93121b1ab9d5e46641e6764f2cca0d51 > > TBR=ilnik@webrtc.org,stefan@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:4172 > > Review-Url: https://codereview.webrtc.org/2783183003 > Cr-Commit-Position: refs/heads/master@{#17477} > Committed: https://chromium.googlesource.com/external/webrtc/+/f9ed235c9b7248694edcb46feb1f29ce7456ab59 R=ilnik@webrtc.org,stefan@webrtc.org BUG=webrtc:4172 Review-Url: https://codereview.webrtc.org/2789823002 Cr-Commit-Position: refs/heads/master@{#17498}
2017-04-02 23:53:04 -07:00
virtual VideoSendStream::Stats GetStats();
virtual void OnSendEncodedImage(const EncodedImage& encoded_image,
Reland of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #1 id:1 of https://codereview.webrtc.org/1903193002/ ) Reason for revert: A fix is being prepared downstream so this can now go in. Original issue's description: > Revert of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #5 id:80001 of https://codereview.webrtc.org/1897233002/ ) > > Reason for revert: > API changes broke downstream. > > Original issue's description: > > Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. > > EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport. > > EncodedImageCallback can of course be cleaned up in the future. > > > > This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter. > > > > BUG=webrtc::5687 > > > > Committed: https://crrev.com/f5d55aaecdc39e9cc66eb6e87614f04afe28f6eb > > Cr-Commit-Position: refs/heads/master@{#12436} > > TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:5687 > > Committed: https://crrev.com/a261e6136655af33f283eda8e60a6dd93dd746a4 > Cr-Commit-Position: refs/heads/master@{#12441} TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5687 Review URL: https://codereview.webrtc.org/1905583002 Cr-Commit-Position: refs/heads/master@{#12442}
2016-04-20 05:05:54 -07:00
const CodecSpecificInfo* codec_info);
// Used to update incoming frame rate.
void OnIncomingFrame(int width, int height);
// Dropped frame stats.
void OnFrameDroppedBySource();
void OnFrameDroppedInEncoderQueue();
void OnFrameDroppedByEncoder();
void OnFrameDroppedByMediaOptimizations();
// Adaptation stats.
void SetAdaptationStats(
const VideoStreamEncoder::AdaptCounts& cpu_counts,
const VideoStreamEncoder::AdaptCounts& quality_counts);
void OnCpuAdaptationChanged(
const VideoStreamEncoder::AdaptCounts& cpu_counts,
const VideoStreamEncoder::AdaptCounts& quality_counts);
void OnQualityAdaptationChanged(
const VideoStreamEncoder::AdaptCounts& cpu_counts,
const VideoStreamEncoder::AdaptCounts& quality_counts);
void OnMinPixelLimitReached();
void OnSuspendChange(bool is_suspended);
void OnInactiveSsrc(uint32_t ssrc);
// Used to indicate change in content type, which may require a change in
// how stats are collected and set the configured preferred media bitrate.
void OnEncoderReconfigured(const VideoEncoderConfig& encoder_config,
uint32_t preferred_bitrate_bps);
// Used to update the encoder target rate.
void OnSetEncoderTargetRate(uint32_t bitrate_bps);
// Implements CpuOveruseMetricsObserver.
void OnEncodedFrameTimeMeasured(int encode_time_ms,
const CpuOveruseMetrics& metrics) override;
int GetSendFrameRate() const;
protected:
// From RtcpStatisticsCallback.
void StatisticsUpdated(const RtcpStatistics& statistics,
uint32_t ssrc) override;
void CNameChanged(const char* cname, uint32_t ssrc) override;
// From RtcpPacketTypeCounterObserver.
void RtcpPacketTypesCounterUpdated(
uint32_t ssrc,
const RtcpPacketTypeCounter& packet_counter) override;
// From StreamDataCountersCallback.
void DataCountersUpdated(const StreamDataCounters& counters,
uint32_t ssrc) override;
// From BitrateStatisticsObserver.
void Notify(uint32_t total_bitrate_bps,
uint32_t retransmit_bitrate_bps,
uint32_t ssrc) override;
// From FrameCountObserver.
void FrameCountUpdated(const FrameCounts& frame_counts,
uint32_t ssrc) override;
void SendSideDelayUpdated(int avg_delay_ms,
int max_delay_ms,
uint32_t ssrc) override;
private:
class SampleCounter {
public:
SampleCounter() : sum(0), num_samples(0) {}
~SampleCounter() {}
void Add(int sample);
int Avg(int64_t min_required_samples) const;
private:
int64_t sum;
int64_t num_samples;
};
class BoolSampleCounter {
public:
BoolSampleCounter() : sum(0), num_samples(0) {}
~BoolSampleCounter() {}
void Add(bool sample);
void Add(bool sample, int64_t count);
int Percent(int64_t min_required_samples) const;
int Permille(int64_t min_required_samples) const;
private:
int Fraction(int64_t min_required_samples, float multiplier) const;
int64_t sum;
int64_t num_samples;
};
struct StatsUpdateTimes {
StatsUpdateTimes() : resolution_update_ms(0), bitrate_update_ms(0) {}
int64_t resolution_update_ms;
int64_t bitrate_update_ms;
};
struct TargetRateUpdates {
TargetRateUpdates()
: pause_resume_events(0), last_paused_or_resumed(false), last_ms(-1) {}
int pause_resume_events;
bool last_paused_or_resumed;
int64_t last_ms;
};
struct FallbackEncoderInfo {
bool is_possible = true;
bool is_active = false;
int on_off_events = 0;
int64_t elapsed_ms = 0;
rtc::Optional<int64_t> last_update_ms;
const int max_frame_diff_ms = 2000;
};
struct FallbackEncoderInfoDisabled {
bool is_possible = true;
bool min_pixel_limit_reached = false;
};
struct StatsTimer {
void Start(int64_t now_ms);
void Stop(int64_t now_ms);
void Restart(int64_t now_ms);
int64_t start_ms = -1;
int64_t total_ms = 0;
};
struct QpCounters {
SampleCounter vp8; // QP range: 0-127.
SampleCounter vp9; // QP range: 0-255.
SampleCounter h264; // QP range: 0-51.
};
// Map holding encoded frames (mapped by timestamp).
// If simulcast layers are encoded on different threads, there is no guarantee
// that one frame of all layers are encoded before the next start.
struct TimestampOlderThan {
bool operator()(uint32_t ts1, uint32_t ts2) const {
return IsNewerTimestamp(ts2, ts1);
}
};
struct Frame {
Frame(int64_t send_ms, uint32_t width, uint32_t height)
: send_ms(send_ms), max_width(width), max_height(height) {}
const int64_t
send_ms; // Time when first frame with this timestamp is sent.
uint32_t max_width; // Max width with this timestamp.
uint32_t max_height; // Max height with this timestamp.
};
typedef std::map<uint32_t, Frame, TimestampOlderThan> EncodedFrameMap;
void PurgeOldStats() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
VideoSendStream::StreamStats* GetStatsEntry(uint32_t ssrc)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
void SetAdaptTimer(const VideoStreamEncoder::AdaptCounts& counts,
StatsTimer* timer) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
void UpdateAdaptationStats(
const VideoStreamEncoder::AdaptCounts& cpu_counts,
const VideoStreamEncoder::AdaptCounts& quality_counts)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
void UpdateEncoderFallbackStats(const CodecSpecificInfo* codec_info,
int pixels)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
void UpdateFallbackDisabledStats(const CodecSpecificInfo* codec_info,
int pixels)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
Clock* const clock_;
const std::string payload_name_;
const VideoSendStream::Config::Rtp rtp_config_;
const rtc::Optional<int> fallback_max_pixels_;
const rtc::Optional<int> fallback_max_pixels_disabled_;
rtc::CriticalSection crit_;
VideoEncoderConfig::ContentType content_type_ RTC_GUARDED_BY(crit_);
const int64_t start_ms_;
VideoSendStream::Stats stats_ RTC_GUARDED_BY(crit_);
std::map<uint32_t, StatsUpdateTimes> update_times_ RTC_GUARDED_BY(crit_);
rtc::ExpFilter encode_time_ RTC_GUARDED_BY(crit_);
int quality_downscales_ RTC_GUARDED_BY(crit_);
int cpu_downscales_ RTC_GUARDED_BY(crit_);
rtc::RateTracker media_byte_rate_tracker_ RTC_GUARDED_BY(crit_);
rtc::RateTracker encoded_frame_rate_tracker_ RTC_GUARDED_BY(crit_);
// Contains stats used for UMA histograms. These stats will be reset if
// content type changes between real-time video and screenshare, since these
// will be reported separately.
struct UmaSamplesContainer {
UmaSamplesContainer(const char* prefix,
const VideoSendStream::Stats& start_stats,
Clock* clock);
~UmaSamplesContainer();
void UpdateHistograms(const VideoSendStream::Config::Rtp& rtp_config,
const VideoSendStream::Stats& current_stats);
void InitializeBitrateCounters(const VideoSendStream::Stats& stats);
bool InsertEncodedFrame(const EncodedImage& encoded_frame);
void RemoveOld(int64_t now_ms);
const std::string uma_prefix_;
Clock* const clock_;
SampleCounter input_width_counter_;
SampleCounter input_height_counter_;
SampleCounter sent_width_counter_;
SampleCounter sent_height_counter_;
SampleCounter encode_time_counter_;
BoolSampleCounter key_frame_counter_;
BoolSampleCounter quality_limited_frame_counter_;
SampleCounter quality_downscales_counter_;
BoolSampleCounter cpu_limited_frame_counter_;
BoolSampleCounter bw_limited_frame_counter_;
SampleCounter bw_resolutions_disabled_counter_;
SampleCounter delay_counter_;
SampleCounter max_delay_counter_;
rtc::RateTracker input_frame_rate_tracker_;
RateCounter input_fps_counter_;
RateCounter sent_fps_counter_;
RateAccCounter total_byte_counter_;
RateAccCounter media_byte_counter_;
RateAccCounter rtx_byte_counter_;
RateAccCounter padding_byte_counter_;
RateAccCounter retransmit_byte_counter_;
RateAccCounter fec_byte_counter_;
int64_t first_rtcp_stats_time_ms_;
int64_t first_rtp_stats_time_ms_;
StatsTimer cpu_adapt_timer_;
StatsTimer quality_adapt_timer_;
BoolSampleCounter paused_time_counter_;
TargetRateUpdates target_rate_updates_;
BoolSampleCounter fallback_active_counter_;
FallbackEncoderInfo fallback_info_;
FallbackEncoderInfoDisabled fallback_info_disabled_;
ReportBlockStats report_block_stats_;
const VideoSendStream::Stats start_stats_;
EncodedFrameMap encoded_frames_;
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ ) Reason for revert: Seem to be a flaky test rather than an issue with this cl. Creating reland, will add code to reduce flakiness to that test. Original issue's description: > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ ) > > Reason for revert: > This has resulted in failure of CallPerfTest.ReceivesCpuOveruseAndUnderuse test on the Win7 build bot https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1780 > > Original issue's description: > > Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ ) > > > > Reason for revert: > > Found issue with test case, will add fix to reland cl. > > > > Original issue's description: > > > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ ) > > > > > > Reason for revert: > > > Breaks perf tests: > > > https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679 > > > https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325 > > > > > > Original issue's description: > > > > Add framerate to VideoSinkWants and ability to signal on overuse > > > > > > > > In ViEEncoder, try to reduce framerate instead of resolution if the > > > > current degradation preference is maintain-resolution rather than > > > > balanced. > > > > > > > > BUG=webrtc:4172 > > > > > > > > Review-Url: https://codereview.webrtc.org/2716643002 > > > > Cr-Commit-Position: refs/heads/master@{#17327} > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/72acf2526177bb4dbb5103cd6e165eb4361a5ae6 > > > > > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org > > > # Skipping CQ checks because original CL landed less than 1 days ago. > > > NOPRESUBMIT=true > > > NOTREECHECKS=true > > > NOTRY=true > > > BUG=webrtc:4172 > > > > > > Review-Url: https://codereview.webrtc.org/2764133002 > > > Cr-Commit-Position: refs/heads/master@{#17331} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/8b45b11144c968b4173215c76f78c710c9a2ed0b > > > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org > > # Not skipping CQ checks because original CL landed more than 1 days ago. > > BUG=webrtc:4172 > > > > Review-Url: https://codereview.webrtc.org/2781433002 > > Cr-Commit-Position: refs/heads/master@{#17474} > > Committed: https://chromium.googlesource.com/external/webrtc/+/3ea3c77e93121b1ab9d5e46641e6764f2cca0d51 > > TBR=ilnik@webrtc.org,stefan@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:4172 > > Review-Url: https://codereview.webrtc.org/2783183003 > Cr-Commit-Position: refs/heads/master@{#17477} > Committed: https://chromium.googlesource.com/external/webrtc/+/f9ed235c9b7248694edcb46feb1f29ce7456ab59 R=ilnik@webrtc.org,stefan@webrtc.org BUG=webrtc:4172 Review-Url: https://codereview.webrtc.org/2789823002 Cr-Commit-Position: refs/heads/master@{#17498}
2017-04-02 23:53:04 -07:00
std::map<int, QpCounters>
qp_counters_; // QP counters mapped by spatial idx.
};
std::unique_ptr<UmaSamplesContainer> uma_container_ RTC_GUARDED_BY(crit_);
};
} // namespace webrtc
#endif // VIDEO_SEND_STATISTICS_PROXY_H_