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author | Philipp Hancke <phancke@nvidia.com> | 2022-05-05 15:55:36 +0200 |
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committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | 2022-05-18 09:16:10 +0000 |
commit | 0359ba22256f26c723edac508886eb5df8073487 (patch) | |
tree | f99c4f0526724c01670801b2e2b546ef5bbf2df6 /media | |
parent | 94c09169a22a847a6aa8d9d13f1949d1bc497c58 (diff) | |
download | webrtc-0359ba22256f26c723edac508886eb5df8073487.tar.gz |
stats: add frame assembly time stats
implements a total frame assembly time statistic that measures the
cumulative time between the arrival of the first packet of a frame
(the lowest reception time) and the time all packets of the frame have
been received (i.e. the highest reception time)
This is similar to totalProcessingDelay
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame.
This statistic is useful for evaluating mechanisms like NACK and FEC
and gives some insight into the behavior of the pacer sending the
packets.
Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added.
Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as
totalAssemblyTime of type double
Only exists for video. The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received.
Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible.
This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket.
framesAssembledFromMultiplePacket of type unsigned long
Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet.
For such frames the totalAssemblyTime is incremented.
BUG=webrtc:13986
Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36922}
Diffstat (limited to 'media')
-rw-r--r-- | media/base/media_channel.h | 2 | ||||
-rw-r--r-- | media/engine/webrtc_video_engine.cc | 3 | ||||
-rw-r--r-- | media/engine/webrtc_video_engine_unittest.cc | 5 |
3 files changed, 10 insertions, 0 deletions
diff --git a/media/base/media_channel.h b/media/base/media_channel.h index 3673169939..bb07f15b9d 100644 --- a/media/base/media_channel.h +++ b/media/base/media_channel.h @@ -615,6 +615,8 @@ struct VideoReceiverInfo : public MediaReceiverInfo { uint64_t total_decode_time_ms = 0; // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay webrtc::TimeDelta total_processing_delay = webrtc::TimeDelta::Millis(0); + webrtc::TimeDelta total_assembly_time = webrtc::TimeDelta::Millis(0); + uint32_t frames_assembled_from_multiple_packets = 0; double total_inter_frame_delay = 0; double total_squared_inter_frame_delay = 0; int64_t interframe_delay_max_ms = -1; diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc index a8f33a960f..8135a526fb 100644 --- a/media/engine/webrtc_video_engine.cc +++ b/media/engine/webrtc_video_engine.cc @@ -3180,6 +3180,9 @@ WebRtcVideoChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo( info.qp_sum = stats.qp_sum; info.total_decode_time_ms = stats.total_decode_time_ms; info.total_processing_delay = stats.total_processing_delay; + info.total_assembly_time = stats.total_assembly_time; + info.frames_assembled_from_multiple_packets = + stats.frames_assembled_from_multiple_packets; info.last_packet_received_timestamp_ms = stats.rtp_stats.last_packet_received_timestamp_ms; info.estimated_playout_ntp_timestamp_ms = diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc index 231fc84f6d..e1bfa73cb1 100644 --- a/media/engine/webrtc_video_engine_unittest.cc +++ b/media/engine/webrtc_video_engine_unittest.cc @@ -6116,6 +6116,8 @@ TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesDecodeStatsCorrectly) { stats.frames_decoded = 14; stats.qp_sum = 15; stats.total_decode_time_ms = 16; + stats.total_assembly_time = webrtc::TimeDelta::Millis(4); + stats.frames_assembled_from_multiple_packets = 2; stream->SetStats(stats); cricket::VideoMediaInfo info; @@ -6144,6 +6146,9 @@ TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesDecodeStatsCorrectly) { info.receivers[0].key_frames_decoded); EXPECT_EQ(stats.qp_sum, info.receivers[0].qp_sum); EXPECT_EQ(stats.total_decode_time_ms, info.receivers[0].total_decode_time_ms); + EXPECT_EQ(stats.total_assembly_time, info.receivers[0].total_assembly_time); + EXPECT_EQ(stats.frames_assembled_from_multiple_packets, + info.receivers[0].frames_assembled_from_multiple_packets); } TEST_F(WebRtcVideoChannelTest, |