This method sends arbitrary number rtp::RcpPackets into one or more IP packets.
It is implemented both in RtcpTranceiver and in RtpRtcp.
Change-Id: I00424ee2f1730ff98626f768846f4ac1ad864933
BUG: webrtc:10742
Change-Id: I00424ee2f1730ff98626f768846f4ac1ad864933
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156240
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29430}
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}
This can be better used to determine the length of test calls,
rather than using the interval metric.
Bug: webrtc:11017
Change-Id: I69f66fa750b061a7d010d591a718555e2b5b34b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156087
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29413}
Without this, we can end up with negative capture-to-render delays
if the jitter buffer sets the render time to an earlier time.
Bug: webrtc:11017
Change-Id: I590509136f630d025cde6e5e13d4a3ee620267ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156081
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29409}
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}
This happend because sdk_unittests is not built on arm/arm64 iOS build.
Bug: webrtc:11022
Change-Id: I8f9adfd48e11c8512c27992804cc9b69dff15ded
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156100
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29407}
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}
This enables creation and removal of receive streams with SSRC 0.
Several related methods, for example SetOutputVolume, still use 0 as a
special value.
Bug: webrtc:8694
Change-Id: I341e6bd6c981c9838997510d8d712ad2948f6460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152780
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29398}
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}
Some of the tests are currently flaky because FEC is disabled if the
RTT is <200 ms, and the simulated network is configured to use 100 ms
for the send transport, but nothing is configured for the receive
transport. This CL configures the receive transport to 100 ms delay.
Bug: webrtc:10920
Change-Id: I79995693ba73683406fa9ced92a7918e6c05473f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154571
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29394}
There's already a DCHECK at construction time ensuring that it's set.
Bug: webrtC:9883
Change-Id: I9f41b77273bb859626546ab3534d483d9172ea5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155581
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29393}
The pattern of using a static function rather than a regular function is
not very well motivated and we don't do that in other places.
To maintain consistency over the code base this Cl replaces those static
modifier functions with regular member functions.
Bug: webrtc:9883
Change-Id: I8edd1781d98905de82722458a0d272af90689a2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155522
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29391}
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}
