- RenderBuffer::Spectrum() loses its channel argument, allowing for
greater flexibility in passing the multi-channel spectrum data into
functions.
- The FFT spectra lengths are made compile-time constant, rendering
some DCHECKs obsolete.
Bug: webrtc:10913
Change-Id: Ied0c50cf72d974cfef7279fd2b9c572d049b8b16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157104
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29528}
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}
Do not fuzz with real renderer because it is merely frame copying and
doesn't exercise different control flows. This CL also improved fuzzing
performance and fixed a memory leak.
Bug: chromium:952606, chromium:1009077, chromium:1009073
Change-Id: I77c6f2581db82bfd95edb18e5f0e541a94c78208
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156620
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29522}
Post-pacer code now contained in RtpSenderEgress class.
For now, this is a member of RTPSender. More refactoring is needed to
make clean split.
Bug: webrtc:11036
Change-Id: I95264d013de120601784f130ba81c7b234446980
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157172
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29519}
It's impossible to ensure we are pulling in everything people
reasonably believe is used, but it should be a good chunk of it.
I don't plan to actually run this test on the bots, it's enough if
it is built (which it should, because I add it to the default set
of things to build).
Bug: webrtc:11027
Change-Id: I186936eeb450d2f63b3a5bed13189e84d5b3fb76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157175
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29518}
that brings RtpPacketReceived closer to the packet buffer
to allow strore original packets rather than VCMPacket in it.
Bug: webrtc:10979
Change-Id: Ia0fc0abf3551a843b19b0ee66ca0f20cae014479
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157164
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29516}
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}
This is a reland of 96f3de094566f32d842be6dd0906f1d13b8c8825
Downstream test is fixed, this is a pure reland.
TBR=danilchap@webrtc.org,srte@webrtc.org
Original change's description:
> Use just a lookup map of RTP modules in PacketRouter
>
> Since SSRCs of RTP modules are now set at construction time, we can
> use just a simple unordered map from SSRC to module in packet router.
>
> Bug: webrtc:11036
> Change-Id: I0b3527f17c9ee2df9253c778e5b9e3651a70b355
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155965
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29510}
Bug: webrtc:11036
Change-Id: I0731339dfd0781cc7f2f7ca78ac903539f25ff9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157304
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29514}
This reverts commit 96f3de094566f32d842be6dd0906f1d13b8c8825.
Reason for revert: Downstream test is borked.
Original change's description:
> Use just a lookup map of RTP modules in PacketRouter
>
> Since SSRCs of RTP modules are now set at construction time, we can
> use just a simple unordered map from SSRC to module in packet router.
>
> Bug: webrtc:11036
> Change-Id: I0b3527f17c9ee2df9253c778e5b9e3651a70b355
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155965
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29510}
TBR=danilchap@webrtc.org,sprang@webrtc.org,srte@webrtc.org
Change-Id: I31330fd68ab809ff3951573791e9a79b81599958
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11036
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157281
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29511}
Since SSRCs of RTP modules are now set at construction time, we can
use just a simple unordered map from SSRC to module in packet router.
Bug: webrtc:11036
Change-Id: I0b3527f17c9ee2df9253c778e5b9e3651a70b355
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155965
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29510}
Also, change test target from rtc_static_library to rtc_source_set so that it is actually linked and run.
Bug: webrtc:11010, webrtc:11037
Change-Id: I05173718ee7de8a9fad73b62c0efd0da4d4f1a7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157166
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29509}
Fixes an issue with NullAudioPoller calling the mixer every 10 ms when
no call is ongoing.
Bug: b/142775365
Change-Id: I77eeddadaf08b358cce2b389c70e4f2baf1d5627
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157176
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29508}
The use of SetTransportWideSequenceNumber() and AllocateSequenceNumber()
is gone from webrtc, but some downstream code still references them.
This means we can do some simplifications.
The member that stores the sequence number is now always accessed while
holding the modules lock, so we can just use that and don't need to add
atomic operations on top.
SetTransportWideSequenceNumber() is only used to set the start sequence
number, it would be nice to set that in the constructor instead.
AllocateSequnceNumber() is now actually only used as a getter, so this
can be replace by a proper const getter method instead.
Bug: webrtc:11036
Change-Id: I69b06e613ca3361cf24ef835b92dd0a894cbd27e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157167
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29507}
Fixes the below build warnings when building with a newer version of
glib. Seen when updating the linux sysroots for crbug.com/1012850
[ 11629/38237 - 588 process @ 649.7/s : 17.899s ] CXX obj/third_party/webrtc/modules/desktop_capture/desktop_capture_generic/base_capturer_pipewire.o
../../third_party/webrtc/modules/desktop_capture/linux/base_capturer_pipewire.cc:253:5: warning: Not available before 2.34 [-W#pragma-messages]
g_clear_object(&cancellable_);
^
../../build/linux/debian_sid_amd64-sysroot/usr/include/glib-2.0/gobject/gobject.h:678:36: note: expanded from macro 'g_clear_object'
^
../../build/linux/debian_sid_amd64-sysroot/usr/include/glib-2.0/glib/gmem.h:142:3: note: expanded from macro 'g_clear_pointer'
GLIB_AVAILABLE_MACRO_IN_2_34
^
../../build/linux/debian_sid_amd64-sysroot/usr/include/glib-2.0/glib/gversionmacros.h:473:49: note: expanded from macro 'GLIB_AVAILABLE_MACRO_IN_2_34'
^
../../build/linux/debian_sid_amd64-sysroot/usr/include/glib-2.0/glib/gmacros.h:991:41: note: expanded from macro 'GLIB_UNAVAILABLE_MACRO'
^
../../build/linux/debian_sid_amd64-sysroot/usr/include/glib-2.0/glib/gmacros.h:988:33: note: expanded from macro '_GLIB_GNUC_DO_PRAGMA'
^
<scratch space>:249:6: note: expanded from here
GCC warning "Not available before " "2" "." "34"
^
../../third_party/webrtc/modules/desktop_capture/linux/base_capturer_pipewire.cc:257:5: warning: Not available before 2.34 [-W#pragma-messages]
g_clear_object(&proxy_);
^
../../build/linux/debian_sid_amd64-sysroot/usr/include/glib-2.0/gobject/gobject.h:678:36: note: expanded from macro 'g_clear_object'
^
../../build/linux/debian_sid_amd64-sysroot/usr/include/glib-2.0/glib/gmem.h:142:3: note: expanded from macro 'g_clear_pointer'
GLIB_AVAILABLE_MACRO_IN_2_34
^
../../build/linux/debian_sid_amd64-sysroot/usr/include/glib-2.0/glib/gversionmacros.h:473:49: note: expanded from macro 'GLIB_AVAILABLE_MACRO_IN_2_34'
^
../../build/linux/debian_sid_amd64-sysroot/usr/include/glib-2.0/glib/gmacros.h:991:41: note: expanded from macro 'GLIB_UNAVAILABLE_MACRO'
^
../../build/linux/debian_sid_amd64-sysroot/usr/include/glib-2.0/glib/gmacros.h:988:33: note: expanded from macro '_GLIB_GNUC_DO_PRAGMA'
^
<scratch space>:254:6: note: expanded from here
GCC warning "Not available before " "2" "." "34"
^
2 warnings generated.
BUG=chromium:1012850, chromium:1014947
R=tommi@webrtc.org
Change-Id: I0f72e1cd6e9b9311cf2cbd5635e7ad8fe489c350
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156980
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29506}
Corresponding mock class is deleted rather than updated,
since it appears unused.
Bug: webrtc:8422
Change-Id: If1c6c5ed73abff0d2545e8666c4bb8b63ee5b53f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/13862
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29505}
This CL sets the RTP stats callback on construction, by adding a field
next to the other observers in RtpRtcp::Configuration.
We can then remove the RegisterCallback() methods and the unused
GetCallback() method.
Bug: webrtc:11036
Change-Id: I4eb86ea63b4b2ebeff60b311ddf3bed06b279ce4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157169
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29504}
Or, well, to be fair it still kind of does the same thing, but
the thing it's (void)ing in is a lot more related to what it
actually happening. I could not find another way to solve this
since fileutils is fundamentally optional to unit tests, but the
flag isn't.
Bug: webrtc:9792
Change-Id: I6ebf012246bc259883bc0aaf73ac7fea5525dd1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157101
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29501}
Video and audio senders are missing mid, rid and rrid extensions in
their GetCapabilities call.
Bug: chromium:1007894
Change-Id: Ie9edba28ae32fda5e501913cac694f43bfb185ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156560
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29493}
This is a reland of 17608dc4592fe25c1effdd75bf856f4af251942e
Downstream test now fixed.
As a precaution, also avoid DCHECKS for non-zero SSRC.
First patch set is reland, second makes checks more lenient.
Original change's description:
> RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
>
> Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
> remove them, make the members const, and remove now unnecessary locking.
>
> Bug: webrtc:10774
> Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29475}
Bug: webrtc:10774
Change-Id: I540b49a31a31e98d87f02ae04083d5206e71c1b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157100
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29491}
This was just checked in all places were it was used, moving the check
into RtpRtcp reduces the boiler plate required at the call sites.
Also changing to always register and unregister extensions by URI to
synchronize the code in AudioSendStream with the code in RtpVideoSender.
This prepares for reducing the scope of ChannelSend.
Bug: webrtc:9883
Change-Id: Ia64d79f20eb98f46cbbbe8318770e4fcf9caa1ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155620
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29490}
I had to change approach. Unfortunately we can't expect
that test_main_lib users link with fileutils, which causes it to
not link when the override symbol is missing.
New approach: resources_dir_flag is now a separate target, it
will be depended upon by the downstream override, which just
reads the flag and returns it as the resource dir. This gets
rid of the mutable state downstream as well.
So:
1) Land this
2) Make downstream read the flag instead of keeping its own state
3) Remove OverrideResourcesDir upstream and clean up the hacks
4) Remove the now orphaned OverrideResourcesDir downstream
Bug: webrtc:9792
Change-Id: Ic2ef3910bb5d39d9fb71e06fbbbb6aec4de52e78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157041
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29487}
This is a reland of 17608dc4592fe25c1effdd75bf856f4af251942e
Downstream fixed, relanding.
Original change's description:
> RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
>
> Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
> remove them, make the members const, and remove now unnecessary locking.
>
> Bug: webrtc:10774
> Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29475}
TBR=nisse@webrtc.org
Bug: webrtc:10774
Change-Id: I759bed3ff1909857696c6d1b13df595a5e552f03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157049
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29486}
This CL corrects the EchoAudibility and StationarityEstimator
code to work properly with multiple channels.
It also changes the naming of the FilterDelayBlocks() method
to better reflect what it does.
The changes have been verified to be bitexact over a large number
of recordings.
Bug: webrtc:10913
Change-Id: I070b531efcdff4c33f70fd5b37fbb556dcebe5b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156565
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29482}
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}
