This also has the benefit of deleting one unneeded call to
RTPPayloadRegistry::last_received_media_payload_type.
To make this work, also extend NackModule with a OnReceivedPacket
method taking only the sequence number and the is_keyframe flag,
rather than a complete VCMPacket.
Bug: webrtc:8995
Change-Id: Ice379581166e7b1609ec719e944a5a543d69acc1
Reviewed-on: https://webrtc-review.googlesource.com/64120
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22584}
In this CL the OnFrame function is implemented.
Bug: webrtc:8909
Change-Id: I68488a033e86eadd0b16d091faad14e9cda7cc36
Reviewed-on: https://webrtc-review.googlesource.com/64121
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22583}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
BUG=webrtc:8445
Change-Id: I51dfe8879c28c91bd1c667fc47b4892373671e0f
Reviewed-on: https://webrtc-review.googlesource.com/21540
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22569}
First step of the transition needed to reland cl
https://webrtc-review.googlesource.com/62062, and move payload_name
and payload_type out of VideoSendStream::Config::EncoderSettings.
If the new field is set to something different than kVideoCodecUnkown,
payload_name from EncoderSettings is ignored.
Bug: webrtc:8830
Change-Id: I515a91f8291cda79017332102cc6a10736d8a648
Reviewed-on: https://webrtc-review.googlesource.com/64001
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22555}
This reverts commit bc900cb1d1810fcf678fe41cf1e3966daa39c88c.
Reason for revert: Broke downstream projects.
Original change's description:
> Move rtp-specific config out of EncoderSettings.
>
> In VideoSendStream::Config, move payload_name and payload_type from
> EncoderSettings to Rtp.
>
> EncoderSettings now contains configuration for VideoStreamEncoder only,
> and should perhaps be renamed in a follow up cl. It's no longer
> passed as an argument to VideoCodecInitializer::SetupCodec.
>
> The latter then needs a different way to know the codec type,
> which is provided by a new codec_type member in VideoEncoderConfig.
>
> Bug: webrtc:8830
> Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> Reviewed-on: https://webrtc-review.googlesource.com/62062
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22532}
TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
Change-Id: I01f06c1fcf21eb2cd40dca7d4f268614200ee490
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8830
Reviewed-on: https://webrtc-review.googlesource.com/63720
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22537}
In VideoSendStream::Config, move payload_name and payload_type from
EncoderSettings to Rtp.
EncoderSettings now contains configuration for VideoStreamEncoder only,
and should perhaps be renamed in a follow up cl. It's no longer
passed as an argument to VideoCodecInitializer::SetupCodec.
The latter then needs a different way to know the codec type,
which is provided by a new codec_type member in VideoEncoderConfig.
Bug: webrtc:8830
Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
Reviewed-on: https://webrtc-review.googlesource.com/62062
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22532}
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}
This CL moves all temporal layer rate allocation from
DefaultTemporalLayers and ScreenshareLayers into SimulcastRateAllocator.
This means we don't need an extra call-out to the TemporalLayers
interface to get the last allocation, which simplifies the code path a
lot.
It also paves the wave for removing the TemporalLayersFactory interface
(in a separate cl), which will further simplify the ownership model.
Bug: webrtc:9012
Change-Id: I6540b1848efa1a136dce449f13902ad479d5ee37
Reviewed-on: https://webrtc-review.googlesource.com/62420
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22502}
Added for the structs VideoCodecVP8, VideoCodecVP9, VideoCodecH264,
and SpatialLayer.
New operators are used to replace memcmp in VCMEncoderDataBase. Using
memcmp to compare structs is generally unreliable, since the struct
may contain random padding bytes due to alignment requirements
(affects at least VideoCodecH264). And in the case of VideoCodecVP8,
we need to exclude the tl_factory pointers from the comparison.
Bug: webrtc:8830
Change-Id: I40432ea7834e288f8c89ce0a28a630ae1800dff8
Reviewed-on: https://webrtc-review.googlesource.com/62761
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22500}
Since we want the VideoStreamDecoder to callback with the last
continuous frame we need to move the FrameKey into the public API.
Bug: webrtc:8909
Change-Id: I39634145d848b8163778e31a1e0d04d91f9bbeb8
Reviewed-on: https://webrtc-review.googlesource.com/60864
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22495}
It previously owned only the picture id and only in the
WebRTC-VP8-Forced-Fallback-Encoder-v2 experiment.
Moving responsibility to PayloadRouter ensures that both
picture id and tl0 idx are continuous over codec changes,
as required by the specs for VP8 and VP9 over RTP.
Bug: webrtc:8830
Change-Id: Ie77356dfec6d1e372b6970189e4c3888451920e6
Reviewed-on: https://webrtc-review.googlesource.com/61640
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22448}
This work is in preparation for refactoring the TemporalLayers api.
Bug: webrtc:9012
Change-Id: I01908ee034fb79996e687ff72d10178acf102321
Reviewed-on: https://webrtc-review.googlesource.com/61781
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22445}
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}
* Removed video files which were not used by any tests.
* Removed ConferenceMotion_1280_720_50.yuv for mobile builds.
Bug: webrtc:8936
Change-Id: I0539e9fce20470fcc2f0af84bd297faffc4b587a
Reviewed-on: https://webrtc-review.googlesource.com/60942
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22359}
* 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}
Add native api conversions for video frames and video renderer. This
also requires some changes to sdk/BUILD to avoid cyclic dependencies.
Bug: webrtc:8832
Change-Id: Ibf21e63bdcae195dcb61d63f9262e6a8dc4fa790
Reviewed-on: https://webrtc-review.googlesource.com/57142
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22340}
It holds the same information as codecType, but in different format.
Bug: webrtc:8830
Change-Id: Ia83e2dff4fd9a5ddb489501b7a1fe80759fa4218
Reviewed-on: https://webrtc-review.googlesource.com/56100
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22307}
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}
Keyframe interval is configurable in codec settings, with no need for
a setter method to toggle it on or off.
Bug: webrtc:8830
Change-Id: Ic20d8829884ed22588f8f8c0cceddd76144a9858
Reviewed-on: https://webrtc-review.googlesource.com/56040
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22280}
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}
Now VCMTiming::MaxWaitingTime will not clip negative values. Thus frame
buffer will be able to distinguish between late frames and when waiting
cycle was simply interrupted by a new inserted frame right before the
waiting timer would expire.
Bug: webrtc:8917
Change-Id: I6b253f459fcb3a346064a103cc92ee332b074e1b
Reviewed-on: https://webrtc-review.googlesource.com/57741
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22210}
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 is a reland of c75f1e45093a8d5cc62937c7708b87aa5c5bf0b0.
Original change's description:
> Reduce locking in VideoReceiver and check the threading model.
>
> Note: This is a subset of code that was previously reviewed here:
> - https://codereview.webrtc.org/2764573002/
>
> * Added two notification methods, DecoderThreadStarting() and DecoderThreadStopped()
> * Allows us to establish a period when the decoder thread is not running and it is
> safe to modify variables such as callbacks, that are only read when the decoder
> thread is running.
> * Allows us to DCHECK thread guarantees/correctness.
> * Allows synchronizing callbacks from the module process thread and have them only
> active while the decoder thread is running.
> * The above, allows us to establish two modes for the thread,
> single-threaded-mutable and multi-threaded-const.
> * Using that knowledge, we can remove |receive_crit_| as well as locking for a
> number of member variables.
> * Removed |VCMFrameBuffer _frameFromFile| (unused).
> * Clean up several of my TODOs
>
> Bug: webrtc:7361, chromium:695438
> Change-Id: Id0048ee9624f76103c088d02825eb5c0d6c8913c
> Reviewed-on: https://webrtc-review.googlesource.com/55000
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22133}
Bug: webrtc:7361, chromium:695438
Change-Id: I32e1dc6c62cb30ad96e6366106f39fe415de49f1
Tbr: philipel@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/56803
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22163}
The plan is to:
1. Move FrameObject to api/video.
2. Rename FrameObject to EncodedFrame.
3. Move EncodedFrame out of the video_coding namespace.
This is the 2nd CL.
Bug: webrtc:8909
Change-Id: I5e76a0a3b306156b8bc1de67834b4adf14bebef9
Reviewed-on: https://webrtc-review.googlesource.com/56182
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22158}
