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10 commits

Author SHA1 Message Date
Evan Shrubsole
a0ee64c57e Add test::FakeEncodedFrame for testing 
Change-Id: I1c8fabe5caf2c723487ec1cd71a379e922026a9d
Bug: webrtc:13996
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260001
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36654}
 2022-04-26 09:26:35 +00:00
Jonas Oreland
e02f9eedb3 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 10/inf 
This patch takes a stab at modules/video_coding,
but reaches only about half.

Bug: webrtc:10335
Change-Id: I0d47d0468b818145470c51ae4e8e75ff58d499ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256112
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36335}
 2022-03-25 12:35:36 +00:00
Johannes Kron
23bfff3383 Change default parameters for the low-latency video pipeline 
min_pacing:8ms, to avoid the situation where bursts of frames are sent
to the decoder at once due to network jitter. The bursts of frames
caused the queues further down in the processing to be full and
therefore drop all frames.

max_decode_queue_size:8, in the event that too many frames have piled
up, do as before and send all frames to the decoder to avoid building
up any latency.

These setting only affect the low-latency video pipeline that is enabled
by setting the playout RTP header extension to min=0ms, max>0ms.

Bug: chromium:1138888
Change-Id: I8154bf3efe7450b770da8387f8fb6b23f6be26bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233220
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35119}
 2021-09-29 09:53:17 +00:00
Danil Chapovalov
d08930d5fb Migrate test VideoDecoders to new VideoDecoder::Configure 
Bug: webrtc:13045
Change-Id: I3b66270de59b441bf8b92bc10f67f59f05e9995e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228436
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34742}
 2021-08-12 15:41:03 +00:00
philipel
ca18809ee5 Move RtpFrameObject and EncodedFrame out of video_coding namespace. 
Bug: webrtc:12579
Change-Id: Ib7ecd624eb5c54abb77fe08440a014aa1e963865
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33542}
 2021-03-23 14:22:47 +00:00
philipel
9aa9b8dbbe Prepare to replace VideoLayerFrameId with int64_t. 
Bug: webrtc:12206
Change-Id: I10bfdefbc95a79e0595956c1a0e688051da6d2b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207180
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33265}
 2021-02-15 14:42:02 +00:00
philipel
a028a1a0f7 Removed old OnDecodedFrame callback. 
Bug: webrtc:9106
Change-Id: Idb13ca9984f1e74585640e78da60a291218f4ade
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183981
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32093}
 2020-09-14 09:58:21 +00:00
Danil Chapovalov
91fdc607d8 In video/ replace mock macros with unified MOCK_METHOD macro 
Bug: webrtc:11564
Change-Id: Iec9dded7a3f045e048d0546c268ae206a7d3a7c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175128
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31264}
 2020-05-14 18:12:40 +00:00
Danil Chapovalov
0c626afcf3 Use newer version of TimeDelta and TimeStamp factories in webrtc 
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g"
git cl format

Bug: None
Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30491}
 2020-02-10 12:21:17 +00:00
philipel
539f9b376e Use a TaskQueue for decoding in VideoStreamDecoderImpl. 
Long term goal is to use the VideoStreamDecoder in the VideoReceiveStream so
that we can stop using legacy VideoCodingModule components and classes. This CL is
one of several in preparation for that.

Bug: webrtc:7408, webrtc:9378
Change-Id: Ifd7e4c3c7d38dbb7c4b0636aaad318c571a29158
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164525
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30211}
 2020-01-10 14:31:22 +00:00