This change keeps the original 48 kHz signal and uses it for the
fullband processing given that the following requirements are
fulfilled:
- Input signal is 48 kHz
- Output signal is 48 kHz
- Multiband processing is performed at 32 kHz
- The multiband processing does not modify the original signal
This avoids unnecessary, lossy resampling and band merging.
Bug: b/130016532
Change-Id: I690c26faba07eab0cbff6c0a95a81d89255dd1a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155966
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29425}
This will allow RtcpPackets to be sent in a more generic way where the
PacketRouter does not have to know about the type.
App::SetSsrc is replaced with SetSenderSsrc
Bug: webrtc:10742
Change-Id: I9fa18d408250f15818dc6898093d9b116603facb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156166
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29420}
This CL passes the spectral power estimates for all channels into
the AecState.
Bug: webrtc:10913
Change-Id: Ie3b5c443be0c63f205e23ed2bfea06d9c447eb39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156165
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29417}
The audio processing in the band-split domain on ARM platforms
operate at a sampling frequency of 32 kHz. This CL upsamples
the signal to fullband before the "fullband processing"
if an output rate of 48 kHz is chosen.
Change-Id: I268acd33aff1fcfa4f75ba8c0fb3e16abb9f74e8
Bug: b/130016532
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155640
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29415}
This CL removes the redundant class in preparation
for adding multichannel functionality to the
reverb computation.
The changes are bitexact.
Bug: webrtc:10913
Change-Id: I284665f7143cb5e1c79bfa573638fdff5f2411c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155960
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29414}
Multi-channel behaviors introduced in this CL:
- All filters are analyzed independently. The filtering is considered
consistent if any filter is consistent.
- The filter echo path gain used to detect saturation is maxed across
capture channels.
- The filter delay is taken to be the minimum of all filters:
Any module that looks in the render data starting from the filter
delay will iterate over all render audio present in any channel.
- The FilterAnalyzer will consider a render block to be active if any
render channel has activity.
The changes in the CL has been shown to be bitexact on a
large set of mono recordings.
Bug: webrtc:10913
Change-Id: I1e360cd7136ee82d1f6e0f8a1459806e83f4426d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155363
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29408}
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}
This CL adds support for multichannel in the residual echo
estimator code. It also adds placeholder functionality in
the surrounding code to ensure that the residual echo
estimator receives the require inputs.
The changes in the CL has been shown to be bitexact on a
large set of mono recordings.
Bug: webrtc:10913
Change-Id: I726128ca928648b1dcf36c5f479eb243f3ff3f96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155361
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29400}
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}
The new configuration path is via AudioProcessing::ApplyConfig and
AudioProcessing::GetStatistics.
ApmTest.Process passes with unchanged reference files if
audio_processing_impl would initialize the VAD with
VoiceDetection::kLowLikelihood instead of kVeryLowLikelihood.
This was verified by testing this CL with that modification.
Bug: webrtc:9878
Change-Id: I4d08df37a07e5c72feeec02a07d6b9435f917d72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155445
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29395}
This CL adds support in the echo subtractor for handling multiple
capture and render channels.
The changes have passed bitexactness tests for substantial set
of mono recordings.
Bug: webrtc:10913
Change-Id: Ib448c9edf172ebc31e8c28db7b2f2a389a53adb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155168
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29389}
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 support in the audio coding module for sending more than
2 channels to the encoder.
Bug: webrtc:11007
Change-Id: I0909b5c37a54c9d2e1353b864e55008cda50ffae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155583
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29385}
This CL moves the various parameters in the the RTPSenderVideo ctor into
a struct, and adds the red/ulpfec payload types to it.
Once the downstream usage of SetUlpfecConfig() is gone, we can make
those members const and avoid locking in SendVideo().
Bug: webrtc:10809
Change-Id: I9a96ab5b2a4eb2997ebf4a3a3e3cd2eb5715fd79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155365
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29384}
Just fixing some minor TODOs in my name. Not worth splitting into
separate CLs as the changes are minor.
Bug: webrtc:9883
Change-Id: I05c54b76507a1d51b92cad080ca4e2dfe8546bf1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155520
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29377}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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 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}
