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PeerConnection Level Framework

API

Documentation

The PeerConnection level framework is designed for end-to-end media quality testing through the PeerConnection level public API. The framework uses the Unified plan API to generate offers/answers during the signaling phase. The framework also wraps the video encoder/decoder and inject it into webrtc::PeerConnection to measure video quality, performing 1:1 frames matching between captured and rendered frames without any extra requirements to input video. For audio quality evaluation the standard GetStats() API from PeerConnection is used.

The framework API is located in the namespace webrtc::webrtc_pc_e2e.

Supported features

  • Single or bidirectional media in the call
  • RTC Event log dump per peer
  • AEC dump per peer
  • Compatible with webrtc::TimeController for both real and simulated time
  • Media
    • AV sync
  • Video
    • Any amount of video tracks both from caller and callee sides
    • Input video from
      • Video generator
      • Specified file
      • Any instance of webrtc::test::FrameGeneratorInterface
    • Dumping of captured/rendered video into file
    • Screen sharing
    • Vp8 simulcast from caller side
    • Vp9 SVC from caller side
    • Choosing of video codec (name and parameters), having multiple codecs negotiated to support codec-switching testing.
    • FEC (ULP or Flex)
    • Forced codec overshooting (for encoder overshoot emulation on some mobile devices, when hardware encoder can overshoot target bitrate)
  • Audio
    • Up to 1 audio track both from caller and callee sides
    • Generated audio
    • Audio from specified file
    • Dumping of captured/rendered audio into file
    • Parameterizing of cricket::AudioOptions
    • Echo emulation
  • Injection of various WebRTC components into underlying webrtc::PeerConnection or webrtc::PeerConnectionFactory. You can see the full list here
  • Scheduling of events, that can happen during the test, for example:
    • Changes in network configuration
    • User statistics measurements
    • Custom defined actions
  • User defined statistics reporting via webrtc::webrtc_pc_e2e::PeerConnectionE2EQualityTestFixture::QualityMetricsReporter interface

Exported metrics

General

  • <peer_name>_connected - peer successfully established connection to remote side
  • cpu_usage - CPU usage excluding video analyzer
  • audio_ahead_ms - Used to estimate how much audio and video is out of sync when the two tracks were from the same source. Stats are polled periodically during a call. The metric represents how much earlier was audio played out on average over the call. If, during a stats poll, video is ahead, then audio_ahead_ms will be equal to 0 for this poll.
  • video_ahead_ms - Used to estimate how much audio and video is out of sync when the two tracks were from the same source. Stats are polled periodically during a call. The metric represents how much earlier was video played out on average over the call. If, during a stats poll, audio is ahead, then video_ahead_ms will be equal to 0 for this poll.

Video

See documentation for DefaultVideoQualityAnalyzer

Audio

  • accelerate_rate - when playout is sped up, this counter is increased by the difference between the number of samples received and the number of samples played out. If speedup is achieved by removing samples, this will be the count of samples removed. Rate is calculated as difference between nearby samples divided on sample interval.
  • expand_rate - the total number of samples that are concealed samples over time. A concealed sample is a sample that was replaced with synthesized samples generated locally before being played out. Examples of samples that have to be concealed are samples from lost packets or samples from packets that arrive too late to be played out
  • speech_expand_rate - the total number of samples that are concealed samples minus the total number of concealed samples inserted that are "silent" over time. Playing out silent samples results in silence or comfort noise.
  • preemptive_rate - when playout is slowed down, this counter is increased by the difference between the number of samples received and the number of samples played out. If playout is slowed down by inserting samples, this will be the number of inserted samples. Rate is calculated as difference between nearby samples divided on sample interval.
  • average_jitter_buffer_delay_ms - average size of NetEQ jitter buffer.
  • preferred_buffer_size_ms - preferred size of NetEQ jitter buffer.
  • visqol_mos - proxy for audio quality itself.
  • asdm_samples - measure of how much acceleration/deceleration was in the signal.
  • word_error_rate - measure of how intelligible the audio was (percent of words that could not be recognized in output audio).

Network

  • bytes_sent - represents the total number of payload bytes sent on this PeerConnection, i.e., not including headers or padding
  • packets_sent - represents the total number of packets sent over this PeerConnection’s transports.
  • average_send_rate - average send rate calculated on bytes_sent divided by test duration.
  • payload_bytes_sent - total number of bytes sent for all SSRC plus total number of RTP header and padding bytes sent for all SSRC. This does not include the size of transport layer headers such as IP or UDP.
  • sent_packets_loss - packets_sent minus corresponding packets_received.
  • bytes_received - represents the total number of bytes received on this PeerConnection, i.e., not including headers or padding.
  • packets_received - represents the total number of packets received on this PeerConnection’s transports.
  • average_receive_rate - average receive rate calculated on bytes_received divided by test duration.
  • payload_bytes_received - total number of bytes received for all SSRC plus total number of RTP header and padding bytes received for all SSRC. This does not include the size of transport layer headers such as IP or UDP.

Framework stability

  • frames_in_flight - amount of frames that were captured but wasn't seen on receiver in the way that also all frames after also weren't seen on receiver.
  • bytes_discarded_no_receiver - total number of bytes that were received on network interfaces related to the peer, but destination port was closed.
  • packets_discarded_no_receiver - total number of packets that were received on network interfaces related to the peer, but destination port was closed.

Examples

Examples can be found in

Stats plotting

Description

Stats plotting provides ability to plot statistic collected during the test. Right now it is used in PeerConnection level framework and give ability to see how video quality metrics changed during test execution.

Usage

To make any metrics plottable you need:

  1. Collect metric data with SamplesStatsCounter which internally will store all intermediate points and timestamps when these points were added.
  2. Then you need to report collected data with webrtc::test::PrintResult(...). By using these method you will also specify name of the plottable metric.

After these steps it will be possible to export your metric for plotting. There are several options how you can do this:

  1. Use webrtc::TestMain::Create() as main function implementation, for example use test/test_main.cc as main function for your test.

    In such case your binary will have flag --plot, where you can provide a list of metrics, that you want to plot or specify all to plot all available metrics.

    If --plot is specified, the binary will output metrics data into stdout. Then you need to pipe this stdout into python plotter script rtc_tools/metrics_plotter.py, which will plot data.

    Examples:

    $ ./out/Default/test_support_unittests \
  --gtest_filter=PeerConnectionE2EQualityTestSmokeTest.Svc \
  --nologs \
  --plot=all \
| python rtc_tools/metrics_plotter.py
    $ ./out/Default/test_support_unittests \
  --gtest_filter=PeerConnectionE2EQualityTestSmokeTest.Svc \
  --nologs \
  --plot=psnr,ssim \
| python rtc_tools/metrics_plotter.py

    Example chart: PSNR changes during the test

  2. Use API from test/testsupport/perf_test.h directly by invoking webrtc::test::PrintPlottableResults(const std::vector<std::string>& desired_graphs) to print plottable metrics to stdout. Then as in previous option you need to pipe result into plotter script.