merge two vectors of the same size into single vector
Remove redundant size_ variable.
Remove redundant variables in the StoredPacket internal struct.
Remove frame_created flags since shortly after it is set, used flag is set to false
Bug: webrtc:10979
Change-Id: Ia37944362abda4e2a6c6741f436f95c45e0f7069
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157174
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29535}
Rename file with tests to match code under test.
Rename fixture by moving 'Test' from prefix to suffix
Bug: None
Change-Id: I54c36d3b517bde7cdffa3a7e74528cc464ea7ad7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157301
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29532}
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).
Source sets always pass all the object files to the linker.
On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.
See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set
Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
Summary:
There is an issue with WebRTC for handling of certain H.264 bitstreams where the packets forming the H.264 stream has non-zero packets before the packet containing SPS.
Typically a IDR (key frame) will have SPS/PPS (if present) or the IDR slice in the first packet.
But this is not required in all cases, for example when packetization-mode = 0, you can have each NALU in separate packet. And certain NALUs can exist before SPS, for example SEI, AUD.
The way WebRTC associates width/height to encoded frames is by tracking the dependency of IDR slices to SPS/PPS.
RTP packets containing SPS/PPS have correct width/height stored in them during parsing of SPS in RtpDepacketizerH264::ProcessStapAOrSingleNalu
IDR packets refer to SPS using ppsid, spsid and the width/height fields get transferred from packet containing SPS to IDR packet in H264SpsPpsTracker::CopyAndFixBitstream.
When packets are assembled into a single encoded H264 frame in PacketBuffer::FindFrames, the loop goes through all the packets/nalus in backward scan from last RTP packet of IDR to first one.
Hence the order of NALUs during this scan is : Last parts of IDR Slice -> Mid parts of IDR Slice RTP packet -> first IDR slice Packet (this should have correct width / height) -> RTP packet containing SPS/PPS (this should have correct width/height)
start_index points to the first RTP packet of the frame and its passed into RtpFrameObject's constructor. RtpFrameObject will use the width/height stored in first RTP packet.
This works fine as long as the first RTP packet has width/height, which will be the case if first RTP packet is IDR or SPS.
In H.264 first RTP packet may be AUD, SEI in those cases, RtpFrameObject will create IDR with width/height = 0 and this causes problem for Android hardware decoders.
On Android hardware decoders rely on correct width/height to initialize the hardware decoder.
Verified on real scenario that we have.
Simulated on AppRTCMobile on IOS Simulator
Added unit tests : ninja -C out/Default && ./out/Default/modules_unittests --gtest_filter=*FrameResolution*
Bug: webrtc:11025
Change-Id: Ie2273aae5e81fd62497e1add084876a3aa05af4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156260
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Shyam Sadhwani <shyamsadhwani@fb.com>
Cr-Commit-Position: refs/heads/master@{#29515}
That allows to use SingleThreadedTaskQueueForTesting via TaskQueueBase interface
but still have access to test-only SendTask function.
Bug: webrtc:10933
Change-Id: I3cc397e55ea2f1ed9e5d885d6a2ccda412beb826
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29480}
Make sure the experiment-derived value is used for VP9.
Bug: webrtc:11024
Change-Id: I80b6d388486f2dec793bc8ca872babe6165dcfb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156562
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29453}
Before this change all layers were glued together at the receive side
into a single IVF frame. This confuses most bitstream parsers.
Since this change all spatial layers would be written as separate frames
on the receive side also (on the send side it's already done that way).
Bug: none
Change-Id: I68543e4d4b336f87699ec3b4a113b8c93af0b7e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156082
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29401}
Add missing includes to files that were transactivly depending on removed includes.
Bug: None
Change-Id: Id5923bb8dc3e1d8fbb664e460278ad3e5993be7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155963
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29396}
This issue happens for default case sps_pps_idr_is_h264_keyframe_ is false
The way PacketBuffer::FindFrames works for H264 is it keeps on skipping the packets till it finds a packet which has last=1
This is checked here : if (sequence_buffer_[index].frame_end)
Inside this block there is a loop, to go back and scan all the packets till start of the frame.
Since the scan is backwards, the sequence of nalus in this scan is IDR -> PPS -> SPS.
Once IDR is detected if (h264_header->nalus[j].type == H264::NaluType::kIdr) , the code will has_h264_idr = true.
When it scans the previous packets, it skips those as has_h264_idr is true. These packets have the SPS / PPS and hence has_h264_sps / pps flags were never set to true.
This resulted in warning as no SPS/PPS has been found for IDR.
Test plan : verified loopback call on IOS simulator using H264 codec and the warning log "Received H.264-IDR frame..." is not present anymore
Bug: webrtc:11006
Change-Id: Icbe8a393e3679a8d621af6c76e4999fd60db04a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155420
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Shyam Sadhwani <shyamsadhwani@fb.com>
Cr-Commit-Position: refs/heads/master@{#29386}
This CL adds an optional second encoder factory to SimulcastEncoderAdapter,
that can be used to create software fallback adapter per simulcast layer.
It also adds logic to check if the encoder supports simulcast natively, if so
it only allocates a single instance and delegates the simulcast logic to that
encoder instead. This means we will be able to remove EncoderSimulcastProxy.
Bug: webrtc:11000
Change-Id: Ifd5f029cc281ee2cedf9d18efa5e7e460884d6ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155171
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29364}
Modifying buffers passed in to the frame buffer breaks sharing. This
cl is also a preparation for deleting
VCMEncodedFrame::VerifyAndAllocate and EncodedImage::Allocate.
Bug: None
Change-Id: I4e14bc4708bbcbcd91af2d4b764cb9b8271ec090
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154569
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29336}
Now vp9 screenshare would enable new layers as soon as requested and will force all spatial layers present on the next frame, even if they should be dropped because of frame-rate limiting.
This might cause frame-rate liming to be exceeded if layer is toggling on and off very often, but this situation is bad itself. E.g. in realtime video it will cause too many key-frames.
Now SvcRateAllocator and VP9EncoderImpl are aware that there may be some skipped layers before the first enabled. Key-frames and ss_info triggering logic is also updated.
(This is a reland without changes after updates to downstream projects)
Original-Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153483
Bug: webrtc:10977
Change-Id: I02459c5982da2e0542a837514f5753c5f96401c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154355
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29330}
In this CL:
- Moved critical section out of RtpFrameReferenceFinder.
- RtpFrameReferenceFinder can now assign picture ids with an offset.
- RtpVideoStreamReceiver will now reset the |reference_finder_| in case
of a codec switch.
Bug: webrtc:10795, webrtc:10828
Change-Id: I22631c121a465c434de24af5ce8be2a647fe3556
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154353
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29317}
This reverts commit 88fe84b7fb.
Reason for revert: Downstream project isn't updated to the latest libvpx roll yet, thus some tests are broken.
Original change's description:
> VP9 encoder: handle disabled layers correctly
>
> Now vp9 screenshare would enable new layers as soon as requested and will
> force all spatial layers present on the next frame, even if they should be
> dropped because of frame-rate limiting.
>
> This might cause frame-rate liming to be exceeded if layer is toggling on
> and off very often, but this situation is bad itself. E.g. in realtime video
> it will cause too many key-frames.
>
> Now SvcRateAllocator and VP9EncoderImpl are aware that there may be some skipped
> layers before the first enabled. Key-frames and ss_info triggering logic is also
> updated.
>
> Bug: webrtc:10977
> Change-Id: Ie2555210c0368a1d3c51ddf6670d0052e6d679de
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153483
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29296}
TBR=ilnik@webrtc.org,ssilkin@webrtc.org
Change-Id: If33886a5f8a0c3b33168dcadfe45c11a6f4387c1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10977
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154354
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29299}
Now vp9 screenshare would enable new layers as soon as requested and will
force all spatial layers present on the next frame, even if they should be
dropped because of frame-rate limiting.
This might cause frame-rate liming to be exceeded if layer is toggling on
and off very often, but this situation is bad itself. E.g. in realtime video
it will cause too many key-frames.
Now SvcRateAllocator and VP9EncoderImpl are aware that there may be some skipped
layers before the first enabled. Key-frames and ss_info triggering logic is also
updated.
Bug: webrtc:10977
Change-Id: Ie2555210c0368a1d3c51ddf6670d0052e6d679de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153483
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29296}
Passing an empty arg is working at the moment but it is not
guaranteed to continue to work in the future.
This CL has been generated with:
git grep -l "INSTANTIATE_TEST_SUITE_P(," | xargs sed -i \
"s/INSTANTIATE_TEST_SUITE_P(,/INSTANTIATE_TEST_SUITE_P(All,/g"
Bug: None
Change-Id: Icd2fb9d9d29aed5d692a234124bd990d0f097db4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153890
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29282}
This method used to be wired down to VCMReceiver and to
VCMJitterBuffer::Stop, but has become a nop. Also delete some
obsoleted comments.
Bug: webrtc:7408
Change-Id: I4c1e67272b1ffda786cc0ff358fa38e594aff304
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152620
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29167}
This mode was added by libvpx team specificaly for this usecase: if a
layer is dropped, all lower layers have to be dropped also.
This ensures that higher layers always have higher framerate than the
lower layers and stream is RTP compatible.
This CL also renames full_superframe_drop_ to !layer_buffering, as it
closer reflects the purpose of that flag (in screenshare mode, no
buffering is needed, because the highest layer is always present in the
superframe, yet, it's not a full-superframe dropping mode).
Bug: webrtc:10257
Change-Id: I2589bfd2b9b63de0e410f277a716276234993843
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151764
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29155}
This CL adds a field trial parameter WebRTC-SlowDownDecoder that is
used to simulate a slow decoder. The parameter specifies how many
extra ms it takes to decode each video frame. This must only be used
in manual testing.
Bug: None
Change-Id: Iad4079100d67b95c224277aaeaf572e38068717f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151911
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29153}
The new target, modules/video_coding:video_coding_legacy, is not
depended upon by any webrtc non-test code.
Bug: webrtc:7408
Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29133}
A followup cl will move VideoCodingModule and related code into this
target.
Bug: webrtc:7408
Change-Id: Iade572b597769456c9b8c76f584500e2bd9a58f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29122}
The vcm::VideoReceiver class is used by both VideoReceiveStream and
the legacy api VideoCodingModule. They have different requirements,
since the latter uses the old jitterbuffer and runs the code on a
ProcessThread.
By making a copy and trimming it down to what's actually used by
VideoReceiveStream, we can drop the dependency on the old
jitterbuffer, without breaking the legacy api. This should also make
it easier to do follow-up refactorings to trim down the class further,
and ultimately remove it.
Bug: webrtc:7408
Change-Id: Iec8a167fe5d0425114b0b67a5b4c2fd5fc4fa150
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151910
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29108}
if higher layer is enabled, then disabled, then key-frame is issued, then
the layer is enabled again, the buffer would contain a picture from before
the key-frame and it might have a higher pid than the currently encoded one.
This would trigger the DCHECK. It's safe to remove the DCHECK completely, because
such occasions would cause unsigned overflow and cause the following check for
maximum allowed picture difference to fail and the wrong picture won't
be used as a temporal reference.
This error only caused failures in debug builds and couldn't lead to corruptions
because there're periodical key-frames generated and pid difference can never become so
big that negative value would overflow to something close to 0.
Bug: webrtc:10257
Change-Id: Ie3b3ed0e24421787e3b40a37987ccecb75d04635
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151643
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29099}
