Both simulcast stream and spatial layer can be described with the same
set of parameters. There is no need in two separate definitions.
1. Original definition of SpatialLayer is removed.
2. SimulcastStream is renamed to SpatialLayer.
3. SimulcastStream is equated to SpatialLayer using typedef.
Bug: webrtc:8518
Change-Id: I90761952b032a1b71fc4bba11f74a6daaf58880a
Reviewed-on: https://webrtc-review.googlesource.com/57102
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22272}
This fixes the issue when Init() with correct codec settings fails
because preceding Init() was called with wrong settings.
Bug: webrtc:8969
Change-Id: I50e618af6266ef593942fda27839c7c01e8717ae
Reviewed-on: https://webrtc-review.googlesource.com/59382
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22271}
This CL adds a lock to stashed_images_ in MultiplexEncoderAdapter.
Without lock, it is possible that different threads acts on
stashed_images_ simultaneously and leads to crash.
Bug: webrtc:8965
Change-Id: I887861092d185c3bd6047eb529d8c1cf57fa4648
Reviewed-on: https://webrtc-review.googlesource.com/59260
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22261}
This reverts commit e27e0aca9411b6990fcdf56d8a3475569ee5fd2f.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Revert "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers."
>
> This reverts commit d2ed0a4c9e7f04060d8e3358eb0006c31579bb86.
>
> 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 d2ed0a4c9e7f04060d8e3358eb0006c31579bb86.
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 CL fixes the bug that we must have some padding for H264 codec
when used in Multiplex encoder/decoder, otherwise H264 decoder will
crash.
And this CL fixes a bug that potential infinite loop exists in
MultiplexEncoderAdapter
Bug: webrtc:8921
Change-Id: I4124579c31ee69f72e29d118378aa1f8b3f05eb4
Reviewed-on: https://webrtc-review.googlesource.com/56960
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22175}
Overriding implementations of VideoEncoder::GetScalingSettings that
want to enable quality scaling must now provide the thresholds.
Bug: webrtc:8830
Change-Id: I75c47cb56ac1b9cf77401684980b3167e485f51c
Reviewed-on: https://webrtc-review.googlesource.com/46622
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22172}
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}
Apparently left over since 2015 refactorings, see cl
https://codereview.webrtc.org/1417283007
Bug: webrtc:5095
Change-Id: I899f2c018d1906b4336b2e80b511f7398bac4947
Reviewed-on: https://webrtc-review.googlesource.com/53200
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22055}
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 is a reland of 8fb22e71ee9bd77676838c5723f7e89a74a64aa9.
Original change's description:
> Updates tests for turning simulcast streams on/off.
>
> Due to libvpx we were restricted to always turning the low simulcast
> stream on, or else the encoder would always label the active streams'
> encoded frames as key frames. Now that libvpx has been updated and
> rolled in, this change updates tests to reflect that it is working.
>
> Bug: webrtc:8653
> Change-Id: I065ef817ace2292605e27e135802cf4a3e90647e
> Reviewed-on: https://webrtc-review.googlesource.com/46340
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21831}
TBR=sprang@webrtc.org
Bug: webrtc:8653
Change-Id: I32fa92649f3ff40b1e364f880040e52ae698f74d
Reviewed-on: https://webrtc-review.googlesource.com/46860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21918}
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}
This reverts commit 8fb22e71ee9bd77676838c5723f7e89a74a64aa9.
Reason for revert: breaks downstream projects
Original change's description:
> Updates tests for turning simulcast streams on/off.
>
> Due to libvpx we were restricted to always turning the low simulcast
> stream on, or else the encoder would always label the active streams'
> encoded frames as key frames. Now that libvpx has been updated and
> rolled in, this change updates tests to reflect that it is working.
>
> Bug: webrtc:8653
> Change-Id: I065ef817ace2292605e27e135802cf4a3e90647e
> Reviewed-on: https://webrtc-review.googlesource.com/46340
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21831}
TBR=deadbeef@webrtc.org,sprang@webrtc.org,shampson@webrtc.org
Change-Id: If14074a7fc56c83b75584d8e9a6a913a40514bad
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8653
Reviewed-on: https://webrtc-review.googlesource.com/46840
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21832}
Due to libvpx we were restricted to always turning the low simulcast
stream on, or else the encoder would always label the active streams'
encoded frames as key frames. Now that libvpx has been updated and
rolled in, this change updates tests to reflect that it is working.
Bug: webrtc:8653
Change-Id: I065ef817ace2292605e27e135802cf4a3e90647e
Reviewed-on: https://webrtc-review.googlesource.com/46340
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21831}
This fix the mismatch of resolution VP9 wrapper set for coded frame with
its actual resolution.
Bug: webm:1485, webrtc:5749
Change-Id: Ie1225d8f3a3d00e66229a1a79858d0a89b3d5fae
Reviewed-on: https://webrtc-review.googlesource.com/46040
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21819}
This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.
Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}
TBR=niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: I6f38dc46126f279f334d52b56339b40acdc30511
Reviewed-on: https://webrtc-review.googlesource.com/45820
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21794}
This reverts commit 4954a77cf81e6793245f52d485834acd3e6eab1c.
Reason for revert: Breaks downstream build which was depending on the name "kVideoCodecStereo". Will need to do some sort of trickery to make this change without breaking the relevant code. Sorry. :(
Original change's description:
> Reland "Rename stereo video codec to multiplex"
>
> This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.
> This was reverted because of breaking internal build. I contacted sheriff
> and looked at logs but cannot find anything related to this CL. This was landed
> with #3850 build which caused exception, but 3847-3855 seem to all have failed.
> I am relanding to see if it will work this time or it will give some related
> error message that can guide me.
>
> Original change's description:
> > Rename stereo video codec to multiplex
> >
> > This CL only does the rename from"stereo" to multiplex". With this we have a
> > better name that doesn't clash with audio's usage of stereo.
> >
> > Bug: webrtc:7671
> > Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> > Reviewed-on: https://webrtc-review.googlesource.com/43242
> > Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21769}
>
> TBR=niklas.enbom@webrtc.org
>
> Bug: webrtc:7671
> Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
> Reviewed-on: https://webrtc-review.googlesource.com/44520
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21780}
TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org
Change-Id: I0a71327c2ddfdd030b1e058cd6a41b1689836719
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44621
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21783}
This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.
This was reverted because of breaking internal build. I contacted sheriff
and looked at logs but cannot find anything related to this CL. This was landed
with #3850 build which caused exception, but 3847-3855 seem to all have failed.
I am relanding to see if it will work this time or it will give some related
error message that can guide me.
Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}
TBR=niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
Reviewed-on: https://webrtc-review.googlesource.com/44520
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21780}
This reverts commit bbdabe50db0cf09f6007dda12a6476dc4602b174.
Reason for revert: This breaks the internal build.
Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}
TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org
Change-Id: Icf019cb09e07de45821d31d7d8ea7707d01346ee
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44360
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21774}
This reverts commit 4dc891f5e3a4bcad4db31e1af0ad45b6c471eef2.
Reason for revert: Reverting this CL to make it possible to revert https://webrtc-review.googlesource.com/c/src/+/43242
Original change's description:
> Add unit tests covering MultiplexImageComponent
>
> This CL changes some types in MultiplexImage and MultiplexImageComponent. Also,
> adds unit test coverage in TestMultiplexAdapter for these structs.
>
> Bug: webrtc:7671
> Change-Id: I832d0466dc67d3b6b7fa0d3fb76f02c0190e474f
> Reviewed-on: https://webrtc-review.googlesource.com/44081
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Qiang Chen <qiangchen@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#21770}
TBR=qiangchen@chromium.org,emircan@webrtc.org
Change-Id: I9cce6ed5f2990a2f443e04a9e5913cbd296242e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44341
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21773}
This CL changes some types in MultiplexImage and MultiplexImageComponent. Also,
adds unit test coverage in TestMultiplexAdapter for these structs.
Bug: webrtc:7671
Change-Id: I832d0466dc67d3b6b7fa0d3fb76f02c0190e474f
Reviewed-on: https://webrtc-review.googlesource.com/44081
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Qiang Chen <qiangchen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21770}
This CL only does the rename from"stereo" to multiplex". With this we have a
better name that doesn't clash with audio's usage of stereo.
Bug: webrtc:7671
Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
Reviewed-on: https://webrtc-review.googlesource.com/43242
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21769}
This is in preparation for
https://webrtc-review.googlesource.com/c/src/+/36340
With these changes we can avoid some strange #ifdefs in the code
that uses temporal layers.
Bug: webrtc:7925
Change-Id: I472210738ccc9f73812b8863951befeabec56f15
Reviewed-on: https://webrtc-review.googlesource.com/41280
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21759}
This CL is a followup to https://webrtc-review.googlesource.com/c/src/+/38481.
With the new approach we can just use the generic RTP packetizer to pass frames
over the wire as the specific info is contained within the bitstream. This makes
the new codec more modular and reduces its footprint.
Bug: webrtc:7671
Change-Id: Ib07f72a9d338e3cbfdbf39cba9891a959b5f7552
Reviewed-on: https://webrtc-review.googlesource.com/43220
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21753}
This reverts commit d756fd06fed1b6c65dcb263cbd8f00ca23d72f3b.
Original change's description:
> Revert "Wrap Alpha and YUV frame into one EncodedImage for transmission"
>
> This reverts commit 5670c86aeccc9bc1191725431de7998d21b73c07.
>
> Reason for revert: Breaks downstream build. Need to add "#include <cstring>" to stereo_encoder_adapter.cc to use std::memcpy.
>
> Original change's description:
> > Wrap Alpha and YUV frame into one EncodedImage for transmission
> >
> > With alpha channel, we observe the artifacts on the receiver side, and
> > the reason is that when YUV channel has a key frame, it gives frame_buffer2
> > a chance to drop some previous frames. Then it is possible that some alpha
> > frames got dropped, which break the alpha frame dependence chain.
> >
> > In this CL, we pack the YUV frame and alpha encoded frame together as one
> > entity to solve the issue.
> >
> > Bug: webrtc:8773
> > Change-Id: Ibe746a46cb41fd92b399a7069e1d89f02f292af7
> > Reviewed-on: https://webrtc-review.googlesource.com/38481
> > Commit-Queue: Qiang Chen <qiangchen@chromium.org>
> > Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21737}
>
> TBR=qiangchen@chromium.org,emircan@webrtc.org
>
> Change-Id: I11eff814ce093bf6db327ebcd21b1b71a1929849
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8773
> Reviewed-on: https://webrtc-review.googlesource.com/43260
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21739}
TBR=deadbeef@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org
Change-Id: I0d64b7e7a62e4f35aa012270d3826a23b3fb2337
Bug: webrtc:8773
Reviewed-on: https://webrtc-review.googlesource.com/43440
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Reviewed-by: Qiang Chen <qiangchen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21749}
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 reverts commit 5670c86aeccc9bc1191725431de7998d21b73c07.
Reason for revert: Breaks downstream build. Need to add "#include <cstring>" to stereo_encoder_adapter.cc to use std::memcpy.
Original change's description:
> Wrap Alpha and YUV frame into one EncodedImage for transmission
>
> With alpha channel, we observe the artifacts on the receiver side, and
> the reason is that when YUV channel has a key frame, it gives frame_buffer2
> a chance to drop some previous frames. Then it is possible that some alpha
> frames got dropped, which break the alpha frame dependence chain.
>
> In this CL, we pack the YUV frame and alpha encoded frame together as one
> entity to solve the issue.
>
> Bug: webrtc:8773
> Change-Id: Ibe746a46cb41fd92b399a7069e1d89f02f292af7
> Reviewed-on: https://webrtc-review.googlesource.com/38481
> Commit-Queue: Qiang Chen <qiangchen@chromium.org>
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21737}
TBR=qiangchen@chromium.org,emircan@webrtc.org
Change-Id: I11eff814ce093bf6db327ebcd21b1b71a1929849
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8773
Reviewed-on: https://webrtc-review.googlesource.com/43260
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21739}
With alpha channel, we observe the artifacts on the receiver side, and
the reason is that when YUV channel has a key frame, it gives frame_buffer2
a chance to drop some previous frames. Then it is possible that some alpha
frames got dropped, which break the alpha frame dependence chain.
In this CL, we pack the YUV frame and alpha encoded frame together as one
entity to solve the issue.
Bug: webrtc:8773
Change-Id: Ibe746a46cb41fd92b399a7069e1d89f02f292af7
Reviewed-on: https://webrtc-review.googlesource.com/38481
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21737}
Try to use frame timestamps first if they look reasonable, otherwise
use realtime clock.
Also, lower limit from 90% of target to 85%.
Bug: webrtc:4172, chromium:802290
Change-Id: Iad489be7c7cf637345be4795e5089936ab9fab07
Reviewed-on: https://webrtc-review.googlesource.com/41041
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21729}
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}
