3449 Commits

Author SHA1 Message Date
Erik Språng
78c82a4040 Adds trial to always start probes with a small padding packet.
This will reduce bias caused by uncertainty in averaging window.

Bug: None
Change-Id: I5c4fe39ffe69fb4af87d86995196a54115d3e0b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144720
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29374}
2019-10-03 17:19:22 +00:00
Erik Språng
6cf554ecb4 Reduces locking in RtpSenderVideo.
This CL removes some unnecessary locking, since we are already
serialized by the lock in VideoStreamEncoder. A simple RaceChecker is
used to verify this.

We also remove the usage of RegisterPayloadType() and replace it with
a parameter in SendVideo instead. This way we are prepared for removing
the payload type map and lock entirely. Some usage still exists
downstream and needs to be removed before cleaning this up.

Bug: webrtc:10809
Change-Id: Ie90163f15d11c8843f3beaf9a0df0dd2a1fd5ce6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154700
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29372}
2019-10-03 14:23:30 +00:00
Sebastian Jansson
62aee9379c Adds trial to calculate audio overhead based on available data.
This adds the ability to disable legacy overhead calculation so we'll
use the available data on per packet over head and frame length range
to set the min and max total  allocatable bitrate.

Bug: webrtc:11001
Change-Id: I2a94499433e15bad11a08f81fe7f1dfc27982cdf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155175
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29368}
2019-10-02 13:42:15 +00:00
Erik Språng
f1e97b9ebd Reland "Prepares RtpSenderVideo for batch forwarding of generated packets"
This is a reland of a21d50c1f3eab29fd9026cc67c8cb4017efda5e3

Original change's description:
> Prepares RtpSenderVideo for batch forwarding of generated packets
> 
> In order to reduce contention, this CL avoids taking locks per packet
> and prepares for forwarding all packets for a frame in one call, rather
> than one at a time. This will especially reduce contention in the paced
> sender during very high packet rates.
> 
> Bug: webrtc:10809
> Change-Id: Ifc5fe3759b76a2a45f418b69d29c329e876f96d0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154358
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29323}

Bug: webrtc:10809
Change-Id: I50e0a27eb3b0b1afa39f250febdd564e1e1f06eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155362
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29367}
2019-10-02 09:39:14 +00:00
Erik Språng
f4e0c29ed1 SimulcastEncoderAdapter: support per layer fallback and single encoder proxying
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}
2019-10-01 17:31:44 +00:00
Elad Alon
fddbe6c632 Improve readability in GoogCcNetworkController::OnSentPacket
Bug: None
Change-Id: Iff8a73611982506d44ac6818300663c3a4ac49b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155177
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29363}
2019-10-01 16:27:00 +00:00
Sam Zackrisson
8f736c0aeb AEC3: Analyze multi-channel SubtractorOutput in AecState
Updates SubtractorOutputAnalyzer and AecState::SaturationDetector
to multi-channel.

Bug: webrtc:10913
Change-Id: I39edafdc5d5a4db5cc853cf116d60af0f506b3bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154342
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29355}
2019-10-01 11:36:58 +00:00
philipel
b3bb2040a1 Remove unused RtpFrameObject ctor.
Bug: webrtc:10979
Change-Id: I9ab8cbd3da4c753f0fa318c41b6e74ddd9679901
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155172
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29354}
2019-10-01 11:23:26 +00:00
Sebastian Jansson
e00ea5ef11 Refactoring CapBitrateToThresholds in SendSideBandwidthEstimation.
Renaming and splitting it into helper methods. This is to more clearly
separate the things it does and prepares for moving things to GoogCC.

Additionally, replacing calls with current_target_ as input with
ApplyTargetLimits to better reflect the intended behavior.

Bug: webrtc:9883
Change-Id: I2c47ec74a9cbc271aff91645c763373297f26acc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154425
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29346}
2019-09-30 13:30:32 +00:00
Ilya Nikolaevskiy
002b6f4f23 Fixes for support of disabling lower spatial layers in VP9
1) Always allocate at least one spatial layer in svc rate allocator

2) Ensure tests reflect known existing failing scenario
(k-svc video with no external ref control).

3) Update log representation of bitrate allocation, as it looks very
confusing with lower layers disabled.

Was:
[
[],
[], [x, y, z]]
New:
[
[]
[]
[x,y,z]]

Bug: webrtc:10977
Change-Id: I248d9b44c8848710aa5a194a5c1b96df6a2734ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154744
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29345}
2019-09-30 13:20:12 +00:00
Sam Zackrisson
32eae4c231 AEC3: use different seed for different channels in CNG
Bug: webrtc:10913
Change-Id: Idca6be02b54b67753cfaf6ff588f5271e0cce892
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155160
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29344}
2019-09-30 13:04:00 +00:00
Erik Språng
08a9f98a5a Revert "Prepares RtpSenderVideo for batch forwarding of generated packets"
This reverts commit a21d50c1f3eab29fd9026cc67c8cb4017efda5e3.

Reason for revert: Speculative revert due to unexpected perf changes.

Original change's description:
> Prepares RtpSenderVideo for batch forwarding of generated packets
> 
> In order to reduce contention, this CL avoids taking locks per packet
> and prepares for forwarding all packets for a frame in one call, rather
> than one at a time. This will especially reduce contention in the paced
> sender during very high packet rates.
> 
> Bug: webrtc:10809
> Change-Id: Ifc5fe3759b76a2a45f418b69d29c329e876f96d0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154358
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29323}

TBR=ilnik@webrtc.org,sprang@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10809
Change-Id: I1cbf0ce0cc06f9195b5e0716b8dd4c85f7f6bab1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155164
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29341}
2019-09-30 11:20:04 +00:00
Ilya Nikolaevskiy
e7314cd4a2 In ulpfec receiver check for malformed packets to avoid DCHECKS tirggering
If the packet can't be parsed, the buffer isn't moved to the packet.
Then, a new empty buffer is moved back from the packet.
Thus, the consequtive DCHECK fails because the data isn't the same anymore.

Bug: chromium:1009236
Change-Id: Ie27f438c40f38074d42d8491fe03df45d50eba50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155162
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29340}
2019-09-30 10:40:31 +00:00
Niels Möller
2449d7aa78 Refactor legacy FrameBuffer to use EncodedImageBuffer::Realloc
Preparation for deleting VCMEncodedFrame::VerifyAndAllocate and
EncodedImage::Allocate.

Bug: webrtc:9378
Change-Id: If7c16061962bbd58c3e7d5720189854e00a3d7bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154570
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29339}
2019-09-30 09:48:26 +00:00
philipel
fc3587418d Use new RtpFrameObject ctor for unittests.
Bug: webrtc:10979
Change-Id: I63f501b3a4538d65a73aae226f2006de191dbbec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154565
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29337}
2019-09-30 08:28:45 +00:00
Niels Möller
ff2e215bcd Change FrameBuffer::CombineAndDeleteFrames to allocate a new buffer
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}
2019-09-30 07:06:10 +00:00
Ilya Nikolaevskiy
bc8049ef0b Reland "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.

(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}
2019-09-27 09:28:38 +00:00
Michael Olbrich
63173d5bef pipewire: handle deleting the capturer while a D-Bus call is in progress
If a D-Bus call is in progress when a BaseCapturerPipeWire is deleted, then
the user_data is invalid when the callback function is called. This results
in memory corruption.

To fix this, use a GCancellable. If it is canceled, the callback will be
called with a corresponding error. Detect this error and abort before
accessing the user_data.

Note: The first argument is the 'source_object'. For g_dbus_proxy_call()
this is the proxy object not the connection. This was not a problem before,
because it was not used.

Bug: None
Change-Id: I8d5e3fb5c49fcc9afd61cdb8e8249f78b9434faf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149817
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29326}
2019-09-26 18:58:56 +00:00
Erik Språng
a21d50c1f3 Prepares RtpSenderVideo for batch forwarding of generated packets
In order to reduce contention, this CL avoids taking locks per packet
and prepares for forwarding all packets for a frame in one call, rather
than one at a time. This will especially reduce contention in the paced
sender during very high packet rates.

Bug: webrtc:10809
Change-Id: Ifc5fe3759b76a2a45f418b69d29c329e876f96d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154358
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29323}
2019-09-26 14:58:07 +00:00
philipel
7acc4a4a3a Reset |reference_finder_| on codec switch.
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}
2019-09-26 11:05:59 +00:00
philipel
5dacece70c Removed unused _rotation_set variable from EncodedFrame.
Bug: none
Change-Id: I398417541fb66e58b0ad90c4b17c5d36eb61a004
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154520
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29316}
2019-09-26 10:45:03 +00:00
Ilya Nikolaevskiy
741bab0f6c Add Slice method to CopyOnWriteBuffer and use it in FEC code.
This avoids unnecessary memcpy calls.

Bug: webrtc:10750
Change-Id: I73fe8f1c9659f2c5e59d7fb97b80349a3504a34a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145320
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29315}
2019-09-26 09:48:07 +00:00
philipel
85d5c197a8 Added RtpFrameObject ctor with no PacketBuffer pointer.
Bug: webrtc:10979
Change-Id: Ie6a2b56e7374d60d1f74d8c315216b27df22a19b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154426
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29314}
2019-09-26 08:55:00 +00:00
Sebastian Jansson
2bc55585f6 Renaming variables in SendSideBandwidthEstimation.
This makes them better reflect their contents and usage. Also replacing
zero with infinity where it's used to reflect the lack of a limit.

Bug: webrtc:9883
Change-Id: Ibc498aa3a41d34c16d363e892a927e482949ab51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154423
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29313}
2019-09-26 08:30:40 +00:00
Sebastian Jansson
ad10222289 Cleanup of unused field trials and options in SendSideBandwidthEstimation
Bug: webrtc:9883
Change-Id: Icbf4d6cb84da51f800343675f181e41b7cc45a6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154422
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29306}
2019-09-25 15:08:12 +00:00
Sebastian Jansson
461ee8538a Cleanup of target rates in GoogCC/SendSideBandwidthEstimation.
Removing the redundant last_estimated_bitrate_bps_ and renaming some
members to better reflect the contents. Also replacing the CurrentEstimate
method of SendSideBandwidthEstimation with value specific access methods.

Bug: webrtc:9883
Change-Id: I73cb08e09374adddf5991cb3793fa4a4fee20c85
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154351
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29304}
2019-09-25 14:31:39 +00:00
Per Åhgren
7bdf073c1c First step of adding multi-channel support to the echo subtractor
This CL contains the first step of adding multi-channel support to the
echo subtractor.

The CL is bitexact for the mono case.

Bug: webrtc:10913
Change-Id: I10647b45c692bc001407afc6ff00e26a3e2cffaa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154356
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29303}
2019-09-25 13:27:56 +00:00
Per Åhgren
0e3b1ff8c4 Moving e to comply to the rest of the stack/heap storage scheme
Bug: webrtc:10913
Change-Id: I7dada71fb86e1c7eea27d0aec01b870fd0a6a15e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154347
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29300}
2019-09-25 11:09:22 +00:00
Ilya Nikolaevskiy
90d6efbd4e Revert "VP9 encoder: handle disabled layers correctly"
This reverts commit 88fe84b7fbcb8dffe07b98d21d8a11572259c0d0.

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}
2019-09-25 09:06:59 +00:00
Gustaf Ullberg
7911d3705c AEC3: Simplify use of SignalTransition
Simplifying the use of signal transition and removing unused code.

Bug: webrtc:8671
Change-Id: I0b845405727936b2fa7df7c92ad2e83bea3bc823
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154348
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29298}
2019-09-25 09:00:24 +00:00
Sebastian Jansson
01dd88505c Moves contents of bitrate_controller to goog_cc
This CL moves send_side_bandwidth_estimation.cc/h and
loss_based_bandwidth_estimation.cc/h from modules/bitrate_controller
to modules/congestion_controller/goog_cc.

Bug: webrtc:9883
Change-Id: Ibb2c2ba3762007e7e5114f39042ee96431b73776
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154346
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29297}
2019-09-25 08:43:24 +00:00
Ilya Nikolaevskiy
88fe84b7fb 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}
2019-09-25 08:42:19 +00:00
Sebastian Jansson
f34116e356 Replacing bandwidth adaptation trial with stable target in Opus encoder.
This also means that the NetworkEstimate::bandwidth can be deprecated
as it's currently just a copy of the target_rate.

Bug: webrtc:10981
Change-Id: I1bc57b98480bd77ce052736b19d630c775428546
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153669
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29288}
2019-09-24 16:35:02 +00:00
Jakob Ivarsson
74344d2aa6 Support 2 byte payload size DTX packets in NetEq simulation.
Bug: none
Change-Id: I785f13555c650171e94e400cf15123e8cc17de22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154350
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29286}
2019-09-24 15:18:05 +00:00
Johannes Kron
9d281028c7 Remove deprecated method
Bug: None
Change-Id: Ia390e05e3bb462e0e79bf3ff7fae6cba891e73ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154262
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29285}
2019-09-24 14:52:20 +00:00
Gustaf Ullberg
af3fdc069d AEC3: Suppression filter handles multiple channels
Suppression filter is extended to support the synthesis
of multiple channels. This CL is also a major clean-up of ApplyGain.

The CL has been tested for bit-exactness for single channel output.

Bug: webrtc:10913
Change-Id: I1319f127981552e17dec66701a248d34dcf0e563
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154341
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29284}
2019-09-24 13:50:04 +00:00
Mirko Bonadei
1b575417b3 Always pass arguments to INSTANTIATE_TEST_SUITE_P.
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}
2019-09-24 08:56:24 +00:00
Niels Möller
ef14f072a9 Delete AudioDecoder method IncomingPacket
Only the ISAC codec had an non-trivial implementation, for its unused
adaptive mode. This cl deletes that implementation, and the call
from NetEq, and the interface method.

Bug: webrtc:10098
Change-Id: Iaf7667e0ae867fc9d64286dff4c01a8ce0b6e2a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153882
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29279}
2019-09-24 08:30:24 +00:00
Niels Möller
834a554962 Include module_common_types.h only where needed
Bug: None
Change-Id: I73d493f8f186b429c7be808f4dfac0398f150931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153891
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29277}
2019-09-24 08:22:38 +00:00
Niels Möller
f2690a15b7 Delete unused method SendSideBandwidthEstimation::UpdateReceiverBlock
Bug: None
Change-Id: Ieca6dd99c7e5e06f1bb2306686a6a0f1e788e22f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153844
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29274}
2019-09-24 06:37:33 +00:00
Gustaf Ullberg
a99b89b41c AEC3: Echo remover handles multiple capture signals.
Echo remover processes all microphone signals. Suppression gains are
computed separately for each capture signal. The minimum gains determine
the final suppression gains applied.

Only the first channel is synthesized. A follow-up CL will add the
synthesis of the remaining channels.

Bug: webrtc:10913
Change-Id: Ife7e74c9a9c6c208fca3992e3cfa840b6b7afcfa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153526
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29269}
2019-09-23 14:43:50 +00:00
Danil Chapovalov
f7457e55fe Store PacketBuffer by value instead of as reference counted object
Bug: None
Change-Id: I5a594972e8a8dad731c927a1a374301e549f5d71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153887
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29267}
2019-09-23 13:28:09 +00:00
Niels Möller
544dfb5a97 Delete isac GetBandwidthInfo/SetBandwidthInfo
Bug: webrtc:10098
Change-Id: I4a56cdc6d081b15a1fc52cba2051783daf4e5ae3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153881
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29256}
2019-09-20 13:53:52 +00:00
Danil Chapovalov
ef83cc5458 Add fuzzer testing for Dependency Descriptor rtp header extension
Bug: webrtc:10342
Change-Id: I46c61b9a137a7148ed80ad38da62132dacb270f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153662
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29255}
2019-09-20 12:40:24 +00:00
Danil Chapovalov
04fd21513b Cleanup passing rtp packet to ulpfec receiver.
Pass RtpPacket class of header and raw packet separately

Bug: None
Change-Id: Id6d107db0e3751ff3dec87321ce6f850da0ee33a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153740
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29254}
2019-09-20 11:09:11 +00:00
philipel
0cff4fce55 Removed unused frame_size param from RtpFrameObject ctor.
Bug: webrtc:10979
Change-Id: Idde493dc7f5165e3ca173d5a38861b444b5904a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153668
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29253}
2019-09-20 10:56:01 +00:00
Niels Möller
48b32b748e Delete support for enabling adaptive isac mode
This appears unused. If deleted, other code related to isac bandwidth
estimation becomes unused and may be deleted in followup cls.

Bug: webrtc:10098
Change-Id: Ifeac2e90de895b12c337ea28cc33704350b9abf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153667
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29252}
2019-09-20 10:41:09 +00:00
philipel
b5e4785464 RtpFrameObject now takes an EncodedImageBuffer in its ctor.
Bug: webrtc:10979
Change-Id: Ibc8b4a524ca95b5faa8850a41df8f2f0136a2969
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153666
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29251}
2019-09-20 10:15:01 +00:00
Niels Möller
e0b31677b6 Delete dead code inside #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
Bug: None
Change-Id: If31f2a5e4a2536b3c7fda596f2c251e8074a18d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153671
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29249}
2019-09-20 07:40:10 +00:00
Sam Zackrisson
feee1e4c36 Add flag to APM to force multichannel even with AEC3
Currently, APM fakes multichannel in two ways:
 - With injected AECs, capture processing is only performed on the left
channel. The result is copied into the other channels.
 - With multichannel render audio, all channels are mixed into one
before analysing.

This CL adds a flag to disable these behaviors, ensuring proper
multichannel processing happens throughout the APM pipeline.

Adds killswitches to separately disable render / capture multichannel.

Additionally - AEC3 currently crashes when running with multichannel.
This CL adds the missing pieces to at least have it run without
triggering any DCHECKS, including making the high pass filter properly
handle multichannel.

Bug: webrtc:10913, webrtc:10907
Change-Id: I38795bf8f312b959fcc816a056fba2c68d4e424d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152483
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29248}
2019-09-20 06:36:12 +00:00