This fixes inconsistency in names of variables and fields which
represent spatial/temporal index of layer:
simulcast_svc_idx -> spatial_idx
spatial_layer_idx -> spatial_idx
temporal_layer_idx -> temporal_idx
Also, this adds printing of spatial/temporal index and target bitrate
to RD report.
Bug: none
Change-Id: Ic4dfdadc57a1577bb3d35d1782a152a9dbef0280
Reviewed-on: https://webrtc-review.googlesource.com/69981
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22869}
This makes it easier to add new test cases without modifying the actual test class.
Bug: None
Change-Id: I48e4f14e26cd6610678ffb07ce9fd56e6bc1ac4e
Reviewed-on: https://webrtc-review.googlesource.com/69600
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22840}
It takes some time for rate controller to adapt to content. Quality of first
frames is usually worse than quality of following frames. It makes sense to
exclude first frames from analysis and, thus, avoid negative affect of this
interval on overall results.
Bug: none
Change-Id: Ib0a258889750cf794c7d6fdff26af958f7bbe48a
Reviewed-on: https://webrtc-review.googlesource.com/66100
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22782}
Measure time spent in frame encode callback, accumulate it for layers
and subtract it from measured encode time of next layer frame.
Bug: none
Change-Id: Ifc3baae2f9a49913a55a7de2de9507102edd0295
Reviewed-on: https://webrtc-review.googlesource.com/65981
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22720}
The configurator decides number of spatial layers, their resolution
and bitrate thresholds based on given input resolution and maximum
number of spatial layers.
The allocator distributes available bitrate across spatial and
temporal layers. If there is not enough bitrate to provide acceptable
quality for all spatial layers allocator disables enhancement layers
one by one until the condition is met or number of layers is reduced
to one.
VP9 SVC related unit tests have been updated. Input resolution and
bitrate in these tests have been increased to the level enough to
provide desirable number of spatial layers.
Bug: webrtc:8518
Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b
Reviewed-on: https://webrtc-review.googlesource.com/60340
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22672}
There is no need to use real video as input for encoder in unit tests.
Using generator simplifies testing on mobile devices (no need to upload
files to device).
Bug: none
Change-Id: Ic48609cc6f8eecf90d9956edfdd33135be949398
Reviewed-on: https://webrtc-review.googlesource.com/64526
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22648}
Build superframe out of the nearest non-dropped base layer and current layer.
Bug: none
Change-Id: I26720ea6de44f27046208b220d03942cd2a3d6c7
Reviewed-on: https://webrtc-review.googlesource.com/64921
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22631}
Have not figured out why this metric regressed, but submitting
this CL now to unblock Chromium roll into WebRTC.
Bug: webrtc:9057
Change-Id: I808ad194e1c9107d644a25502a55a7c6fddca7aa
Reviewed-on: https://webrtc-review.googlesource.com/64527
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22600}
As the rate allocation has been moved into entirely into
SimulcastRateAllocator, and the listeners are thus no longer needed,
this class doesn't fill any other purpose than to determine if
ScreenshareLayers or TemporalLayers should be created for a given
simulcast stream. This can however be done just from looking at the
VideoCodec instance, so changing this into a static factory method.
Due to dependencies from upstream projects, keep the class name and
field in VideoCodec around for now.
Bug: webrtc:9012
Change-Id: I028fe6b2a19e0d16b35956cc2df01dcf5bfa7979
Reviewed-on: https://webrtc-review.googlesource.com/63264
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22529}
For the buffering of |input_frames_|, we assume that frames
are ordered per simulcast layer but we make no assumptions
between layers.
For SVC, we still assume ordering of encode callbacks for
the spatial layers. If we ever add async codecs that support SVC,
they should still obey this assumption.
Bug: webrtc:8448
Change-Id: I4ebb0c1e1d0eef41d850ed5b92aacc79d0a11137
Reviewed-on: https://webrtc-review.googlesource.com/60801
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22380}
* Add support for SimulcastEncoderAdapter wrapping of encoder.
* Store input frame timestamps out-of-band, so we don't need to keep
a raw VideoFrame around just for it's timestamp.
* Store current frame rate in |framerate_fps_|, instead of in
codec settings struct.
* Add some comments and reorder some data members.
* Explicitly include VideoBitrateAllocator.
* Change type of |input_frames_|, to avoid one layer of indirection.
* Move VideoProcessor::CalculateFrameQuality to anonymous namespace.
This change should have no functional implications.
Bug: webrtc:8448
Change-Id: I10c140eeda750d9bd37bfb6cb1e8acb401fb91d3
Reviewed-on: https://webrtc-review.googlesource.com/60520
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22346}
Previously, the only user of this code was the
VideoProcessorIntegrationTest. We have now changed that
test to directly calculate image quality metrics using libyuv,
similar to how the full stack tests and browser tests work.
Bug: webrtc:8448
Change-Id: Ia7a607d7ddc37741fba76d56aa7297851ffa1c6b
Reviewed-on: https://webrtc-review.googlesource.com/43760
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22341}
These tests cannot run on simulators but should be enabled on real device
bots in order to catch regressions or crashes in the iOS codecs.
Bug: webrtc:8950
Change-Id: I8e877aa4368683073fdb4586cd6f4add4a1284ad
Reviewed-on: https://webrtc-review.googlesource.com/59040
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22283}
This reverts commit e27e0aca94.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Revert "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers."
>
> This reverts commit d2ed0a4c9e.
>
> Reason for revert: Breaks downstream projects.
>
> Original change's description:
> > Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers.
> >
> > temporal_layer_thresholds_bps served only one purpose: its size was used
> > to infer number of temporal layers. I replaced it with num_temporal_layers,
> > which does what is says.
> >
> > The practical reason for this change is the need to have possibility to
> > distinguish between cases when VP9 SVC temporal layering was/not set
> > through field trial. That was not possible with
> > temporal_layer_thresholds_bps[] because empty vector means 1 temporal
> > layer.
> >
> > Bug: webrtc:8518
> > Change-Id: I275ec3a8c74e8ba409eb049878199f132a20ec51
> > Reviewed-on: https://webrtc-review.googlesource.com/58084
> > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22230}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org,ssilkin@webrtc.org
>
> Change-Id: Ic2940f7f78a74312170940d51ad8967cde8ad42f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8518
> Reviewed-on: https://webrtc-review.googlesource.com/58902
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22234}
TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,ssilkin@webrtc.org
Change-Id: I1900c6b845b9baa9430fb72c3f4e7f2a44b3a8b1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8518
Reviewed-on: https://webrtc-review.googlesource.com/59160
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22256}
* Do not simulate freeze in decoded output file when frames have been dropped.
* Add more DCHECKs and consts.
* Remove unused members |num_encoded_frames_| and |num_decoded_frames_|.
* Move SdpVideoFormat conversion to TestConfig.
Bug: webrtc:8448
Change-Id: Ia879141f36dc23427cd1abcaa66716656fbaac2a
Reviewed-on: https://webrtc-review.googlesource.com/56802
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22239}
This reverts commit d2ed0a4c9e.
Reason for revert: Breaks downstream projects.
Original change's description:
> Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers.
>
> temporal_layer_thresholds_bps served only one purpose: its size was used
> to infer number of temporal layers. I replaced it with num_temporal_layers,
> which does what is says.
>
> The practical reason for this change is the need to have possibility to
> distinguish between cases when VP9 SVC temporal layering was/not set
> through field trial. That was not possible with
> temporal_layer_thresholds_bps[] because empty vector means 1 temporal
> layer.
>
> Bug: webrtc:8518
> Change-Id: I275ec3a8c74e8ba409eb049878199f132a20ec51
> Reviewed-on: https://webrtc-review.googlesource.com/58084
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22230}
TBR=sprang@webrtc.org,stefan@webrtc.org,ssilkin@webrtc.org
Change-Id: Ic2940f7f78a74312170940d51ad8967cde8ad42f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8518
Reviewed-on: https://webrtc-review.googlesource.com/58902
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22234}
temporal_layer_thresholds_bps served only one purpose: its size was used
to infer number of temporal layers. I replaced it with num_temporal_layers,
which does what is says.
The practical reason for this change is the need to have possibility to
distinguish between cases when VP9 SVC temporal layering was/not set
through field trial. That was not possible with
temporal_layer_thresholds_bps[] because empty vector means 1 temporal
layer.
Bug: webrtc:8518
Change-Id: I275ec3a8c74e8ba409eb049878199f132a20ec51
Reviewed-on: https://webrtc-review.googlesource.com/58084
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22230}
Uploading of the file to device causes timeouts. I removed it from
resources for iOS and Android builds and disabled the test that used
the file since we don't really need to run it on test bots.
Bug: webrtc:8936
Change-Id: Ia5e04c4630544eca8e56826c9e89c9c9f4dcb600
Reviewed-on: https://webrtc-review.googlesource.com/58090
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22222}
This script is for running on device tests in parallel.
BUG=webrtc:8448
NOTRY=TRUE
Change-Id: I6b13f76223653ddb6ec999613ef66ac4f82d8567
Reviewed-on: https://webrtc-review.googlesource.com/55561
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22117}
Slicing, aggregation and analysis has been moved to Stats class.
Data of all spatial layers is stored in single Stats object.
Bug: webrtc:8524
Change-Id: Ic9a64859a36a1ccda661942a201cdeeed470686a
Reviewed-on: https://webrtc-review.googlesource.com/50301
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22094}
Prior to this change, the VideoProcessor was run on the main thread
in the unit tests. Using a TaskQueue there instead, we can be
stricter in the thread checks.
Bug: webrtc:8524
Change-Id: Ice7b68f7344fc52801dff7a98cbc219b7231bfbc
Reviewed-on: https://webrtc-review.googlesource.com/48921
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21942}
Accessing this method from the test thread is illegal,
but doesn't always fail.
Bug: webrtc:8524
Change-Id: Ie0e84cc2fb63268fb6d7cbf0c3a58cb35312c16b
Reviewed-on: https://webrtc-review.googlesource.com/49061
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21930}
This helps separate concerns, so that the VideoProcessorIntegrationTest
is almost oblivious to the fact that it needs to connect to the JVM
to get the Android HW codecs.
Bug: webrtc:8448
Change-Id: I4359b31f84be48eaf99d83525bcce6e593e874f8
Reviewed-on: https://webrtc-review.googlesource.com/47384
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21890}
The VideoCodecTest class is a fixture base class for the
libvpx-VP8, libvpx-VP9, and OpenH264 unit tests. It is unrelated
to the VideoProcessor tests, which we colloquially refer to as
the "codec test".
This rename is thus to reduce this confusion. It should have no
functional impact.
Bug: webrtc:8448
Change-Id: If73443bda5df0f29a71ce6ce069ac128795ff0ad
Reviewed-on: https://webrtc-review.googlesource.com/47160
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21867}
This can be used to wrap Objective-C components in C++ classes, so users
can use the WebRTC C++ API directly together with the iOS specific
components provided by our SDK.
Bug: webrtc:8832
Change-Id: I6d34f7ec62d51df8d3a5340a2e17d30ae73e13e8
Reviewed-on: https://webrtc-review.googlesource.com/46162
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21850}
Encoded frames are preserved and decoded after all layers are
encoded.
Each spatial layer is decoded with separate decoder.
For quality evaluation of lowres layers original input frame is
downscaled with bilinear interpolation.
Encoded and decoded frames are dumped into separate files.
For async codecs encoded frames are passed to decoder in encode
callback, as before.
Bug: webrtc:8524
Change-Id: Idb0c92c7274c1915cff9a011a2794f1cf4bc8cb1
Reviewed-on: https://webrtc-review.googlesource.com/43381
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21844}
Each simulcast stream requires dedicated decoder for decoding. SVC
can be decoded by single decoder. But in prod each receiver has its
decoder. We want to replicate this and also use one decoder per
spatial layer.
Also we create one frame writer per simulcast/spatial layer to dump
encoded/decoded frames of different layers to separate files.
Note that videoprocessor is still initialized with single
decoder/writer. It will be updated in next CL and start using
separate decoder/writer per layer.
Bug: webrtc:8524
Change-Id: I3bb3de77f97d51138b8b7675dd01bc281a078b2f
Reviewed-on: https://webrtc-review.googlesource.com/43280
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21744}
These parameters allow to configure number of simulcast/spatial layers
in video codec tests.
Bug: webrtc:8524
Change-Id: Iad1332732758a8297abcf740c24c483e5fccec9a
Reviewed-on: https://webrtc-review.googlesource.com/43020
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21741}
This feature is not needed in video codec testing framework. In WebRTC
video codecs never deal with packet loss. Packet loss is handled by
jitter buffer which prevents passing of incomplete frames to decoder.
Bug: webrtc:8768
Change-Id: I211cf51d913bec6a1f935e30691661d428ebd3b6
Reviewed-on: https://webrtc-review.googlesource.com/40740
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21722}
No stats is logged in this format any longer.
Bug: none
Change-Id: I5f91e93636b6d03ebd91c3b2518857275fb94de7
Reviewed-on: https://webrtc-review.googlesource.com/40700
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21690}
This is a reland of 1880c7162b
Original change's description:
> Updated analysis in videoprocessor.
>
> - Run analysis after all frames are processed. Before part of it was
> done at bitrate change points;
> - Analysis is done for whole stream as well as for each rate update
> interval;
> - Changed units from number of frames to time units for some metrics
> and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
> 'time to reach target bitrate, sec';
> - Changed data type of FrameStatistic::max_nalu_length (renamed to
> max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
> use such advanced data type in such low level data structure.
>
> Bug: webrtc:8524
> Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
> Reviewed-on: https://webrtc-review.googlesource.com/31901
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21653}
TBR=brandtr@webrtc.org, stefan@webrtc.org
Bug: webrtc:8524
Change-Id: Ie0ad7790689422ffa61da294967fc492a13b75ae
Reviewed-on: https://webrtc-review.googlesource.com/40202
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21668}
This reverts commit 1880c7162b.
Reason for revert: breaks internal tests
Original change's description:
> Updated analysis in videoprocessor.
>
> - Run analysis after all frames are processed. Before part of it was
> done at bitrate change points;
> - Analysis is done for whole stream as well as for each rate update
> interval;
> - Changed units from number of frames to time units for some metrics
> and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
> 'time to reach target bitrate, sec';
> - Changed data type of FrameStatistic::max_nalu_length (renamed to
> max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
> use such advanced data type in such low level data structure.
>
> Bug: webrtc:8524
> Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
> Reviewed-on: https://webrtc-review.googlesource.com/31901
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21653}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,ssilkin@webrtc.org
Change-Id: Id0b7d387bbba02e71637b229aeed6f6cf012af46
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8524
Reviewed-on: https://webrtc-review.googlesource.com/40220
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21656}
- Run analysis after all frames are processed. Before part of it was
done at bitrate change points;
- Analysis is done for whole stream as well as for each rate update
interval;
- Changed units from number of frames to time units for some metrics
and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
'time to reach target bitrate, sec';
- Changed data type of FrameStatistic::max_nalu_length (renamed to
max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
use such advanced data type in such low level data structure.
Bug: webrtc:8524
Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
Reviewed-on: https://webrtc-review.googlesource.com/31901
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21653}
We currently use raw jobject in our code mixed with sporadic
ScopedLocalRefFrame. This CL moves every jobject into a scoped object,
either local, global, or a parameter. Also, this CL uses the JNI
generation script to generate declaration stubs for the Java->C++
functions so that it no longer becomes possible to mistype them
without getting compilation errors.
TBR=brandt@webrtc.org
Bug: webrtc:8278,webrtc:6969
Change-Id: Ic7bac74a89c11180177d65041086d7db1cdfb516
Reviewed-on: https://webrtc-review.googlesource.com/34655
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21387}
The problem was that the encoder was feeded with frames that had 0 as
a timestamp. This confused the encoder. H264 high profile support
clause was also wrong and is corrected.
Bug: webrtc:8601
Change-Id: Ic5a893b4b7573e694f865b63620843b2c9aa489f
Reviewed-on: https://webrtc-review.googlesource.com/32300
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21234}
