This is a reland of 87a7b82520b83a6cf42da27cdc46142c2eb6248c
Original change's description:
> Refactoring of the noise suppressor and adding true multichannel support
>
> This CL adds proper multichannel support to the noise suppressor.
> To accomplish that in a safe way, a full refactoring of the noise
> suppressor code has been done.
>
> Due to floating point precision, the changes made are not entirely
> bitexact. They are, however, very close to being bitexact.
>
> As a safety measure, the former noise suppressor code is preserved
> and a kill-switch is added to allow revering to the legacy noise
> suppressor in case issues arise.
>
> Bug: webrtc:10895, b/143344262
> Change-Id: I0b071011b23265ac12e6d4b3956499d122286657
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158407
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29646}
Bug: webrtc:10895, b/143344262
Change-Id: I236f1e67bb0baa4e30908a4cf7a8a7bb55fbced3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158747
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29663}
This reverts commit 87a7b82520b83a6cf42da27cdc46142c2eb6248c.
Reason for revert: Speculative revert. Breaks downstream projects.
Original change's description:
> Refactoring of the noise suppressor and adding true multichannel support
>
> This CL adds proper multichannel support to the noise suppressor.
> To accomplish that in a safe way, a full refactoring of the noise
> suppressor code has been done.
>
> Due to floating point precision, the changes made are not entirely
> bitexact. They are, however, very close to being bitexact.
>
> As a safety measure, the former noise suppressor code is preserved
> and a kill-switch is added to allow revering to the legacy noise
> suppressor in case issues arise.
>
> Bug: webrtc:10895, b/143344262
> Change-Id: I0b071011b23265ac12e6d4b3956499d122286657
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158407
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29646}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: I4d4025bda01f484979961fe57380a705e4d78397
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10895, b/143344262
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158701
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29651}
This CL adds proper multichannel support to the noise suppressor.
To accomplish that in a safe way, a full refactoring of the noise
suppressor code has been done.
Due to floating point precision, the changes made are not entirely
bitexact. They are, however, very close to being bitexact.
As a safety measure, the former noise suppressor code is preserved
and a kill-switch is added to allow revering to the legacy noise
suppressor in case issues arise.
Bug: webrtc:10895, b/143344262
Change-Id: I0b071011b23265ac12e6d4b3956499d122286657
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158407
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29646}
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 removes and replaces the legacy fixed-point high-pass filter in
APM with the floating point high-pass filter in AEC3.
Bug: webrtc:10907
Change-Id: I88cf8f622ab139e4ffa97f89a72425aa3becfc58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150103
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28950}
This is a reland of b7b8e30cb44c41f51dbbefb9a9160e6dfe869c5b
Original change's description:
> Reland Process 8 kHz audio as 16 kHz internally of the audio processing module
>
> This CL relands the code from the CL "Process 8 kHz audio as 16 kHz internally
> of the audio processing module" which by mistake was reverted via a rebase in
> another CL.
>
> The CL changes the behavior of APM for 8 kHz so that it is internally
> processed as 16 kHz.
>
> Bug: webrtc:10863
> Change-Id: I32a57b2d279c2134125667c19b09cfda381a33c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150221
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28944}
Bug: webrtc:10863
Change-Id: Ic626b99b099248f0d8a677dc4cfe1505e14ae7cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150330
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28949}
This reverts commit b7b8e30cb44c41f51dbbefb9a9160e6dfe869c5b.
Reason for revert: Broke ApmTest.Process test in internal iOS waterfall
Original change's description:
> Reland Process 8 kHz audio as 16 kHz internally of the audio processing module
>
> This CL relands the code from the CL "Process 8 kHz audio as 16 kHz internally
> of the audio processing module" which by mistake was reverted via a rebase in
> another CL.
>
> The CL changes the behavior of APM for 8 kHz so that it is internally
> processed as 16 kHz.
>
> Bug: webrtc:10863
> Change-Id: I32a57b2d279c2134125667c19b09cfda381a33c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150221
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28944}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: Ia49e07b0c25c49da646917516e317f1d57cc4e84
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10863
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150326
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28948}
This CL relands the code from the CL "Process 8 kHz audio as 16 kHz internally
of the audio processing module" which by mistake was reverted via a rebase in
another CL.
The CL changes the behavior of APM for 8 kHz so that it is internally
processed as 16 kHz.
Bug: webrtc:10863
Change-Id: I32a57b2d279c2134125667c19b09cfda381a33c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150221
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28944}
This is a reland of 81c0cf287c8514cb1cd6f3baca484d668c6eb128
Original change's description:
> Simplification and refactoring of the AudioBuffer code
>
> This CL performs a major refactoring and simplification
> of the AudioBuffer code that.
> -Removes 7 of the 9 internal buffers of the AudioBuffer.
> -Avoids the implicit copying required to keep the
> internal buffers in sync.
> -Removes all code relating to handling of fixed-point
> sample data in the AudioBuffer.
> -Changes the naming of the class methods to reflect
> that only floating point is handled.
> -Corrects some bugs in the code.
> -Extends the handling of internal downmixing to be
> more generic.
>
> Bug: webrtc:10882
> Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28928}
Bug: webrtc:10882
Change-Id: I2ddf327e80a03468c41662ae63c619ff34f2363a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150101
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28938}
This CL changes the behavior of APM for 8 kHz so that it is internally
processed as 16 kHz.
Bug: webrtc:10863
Change-Id: Ie17de6551c6e984b60534820374a49ca298f06ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148800
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28929}
This CL removes all external access to the integer sample data in the
AudioBuffer class. It also removes the API in AudioBuffer that provides this.
The purpose of this is to pave the way for removing the sample
duplicating and implicit conversions between integer and floating point
sample formats which is done inside the AudioBuffer.
Bug: webrtc:10882
Change-Id: I1438b691bcef98278aef8e3c63624c367c2d12e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149162
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28912}
- add test that checks that the computed VAD probability is within
tolerance *1
- speed-up some tests by reducing the input length and skipping frames
- remove unused code in test_utils
- fix some comments
*1: RnnVadTest::RnnBitExactness is replaced by
RnnVadTest::RnnVadProbabilityWithinTolerance
Bug: webrtc:10480
Change-Id: I19332d06eacffbbe671bf7749ff4c92798bdc55c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133910
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27803}
As non-linear mode uses a suppressed version of y (not e) as output, this change
uses Y2, rather than E2, as nearend spectrum when computing the suppression gains.
E2 is still used in linear mode.
This change also affects how the minimum suppression gains are calculated. The
minimum gain is now min_echo_power / weighted_residual_echo.
Bug: webrtc:10550
Change-Id: I2904c5a09dd64b06bf25eb5a37c18dab50297794
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133023
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27629}
This CL changes the APM unittests to use AEC3 instead of
AEC2.
Bug: webrtc:8671
Change-Id: I80f88dbafb7c31696abd8b7efb5a187a9fb30d1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129420
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27607}
The difference to the original is new bitexactness strings. The
reason for reland is breaking downstream projects.
Original CL description:
Tests for multi-stream Opus.
This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
tests are in audio_coding_unittest.cc. Some refactoring of
AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
possible. A few checks for "channels \in {1, 2}" are replaced with
"channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
other changes are made to be able to write and read multi-channel WAV
files.
The SDP changes are NOT included; as of this CL there is no way to set
up a multi-stream opus en/de-coder from SDP strings.
TBR=ossu@webrtc.org
Bug: webrtc:8649
Change-Id: I6261b18c69fd666d43ab34ed8f1bc9d5cc82b21f
Reviewed-on: https://webrtc-review.googlesource.com/c/123882
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26809}
This reverts commit 5341aaccdb64e3336abf5875e8828222446adffa.
Reason for revert: Order of initialization of global static strings.
Original change's description:
> Reland of https://webrtc-review.googlesource.com/c/src/+/114883
>
> The difference to the original is new bitexactness strings AND
> global static file string constants. The reason for reland is breaking
> downstream projects.
>
> Original CL description:
>
> Tests for multi-stream Opus.
>
> This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
> tests are in audio_coding_unittest.cc. Some refactoring of
> AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
> possible. A few checks for "channels \in {1, 2}" are replaced with
> "channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
> other changes are made to be able to write and read multi-channel WAV
> files.
>
> The SDP changes are NOT included; as of this CL there is no way to set
> up a multi-stream opus en/de-coder from SDP strings.
>
> Bug: webrtc:8649
> Change-Id: I9fd47c790c241c1876c4a731b0840bec30b4f1b2
> Reviewed-on: https://webrtc-review.googlesource.com/c/123387
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26774}
TBR=aleloi@webrtc.org,ossu@webrtc.org
Change-Id: I88060f2050ccee83d6091b042a10f79b3c4edc47
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8649
Reviewed-on: https://webrtc-review.googlesource.com/c/123580
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26777}
The difference to the original is new bitexactness strings AND
global static file string constants. The reason for reland is breaking
downstream projects.
Original CL description:
Tests for multi-stream Opus.
This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
tests are in audio_coding_unittest.cc. Some refactoring of
AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
possible. A few checks for "channels \in {1, 2}" are replaced with
"channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
other changes are made to be able to write and read multi-channel WAV
files.
The SDP changes are NOT included; as of this CL there is no way to set
up a multi-stream opus en/de-coder from SDP strings.
Bug: webrtc:8649
Change-Id: I9fd47c790c241c1876c4a731b0840bec30b4f1b2
Reviewed-on: https://webrtc-review.googlesource.com/c/123387
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26774}
This reverts commit 9c31ac23231a3494a794b3ba0a6b018969eaa7aa.
Reason for revert: Breaks downstream project.
Original change's description:
> Tests for multi-stream Opus.
>
> This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
> tests are in audio_coding_unittest.cc. Some refactoring of
> AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
> possible. A few checks for "channels \in {1, 2}" are replaced with
> "channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
> other changes are made to be able to write and read multi-channel WAV
> files.
>
> The SDP changes are NOT included; as of this CL there is no way to set
> up a multi-stream opus en/de-coder from SDP strings.
>
> Bug: webrtc:8649
> Change-Id: I1d93a9b8eecc3f6e19896ff2e2ce9b63da77a23c
> Reviewed-on: https://webrtc-review.googlesource.com/c/114883
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26742}
TBR=aleloi@webrtc.org,ossu@webrtc.org
Change-Id: I0ac48b7320d31d3e7af33bf8714c3db6c807b82e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8649
Reviewed-on: https://webrtc-review.googlesource.com/c/123385
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26747}
This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
tests are in audio_coding_unittest.cc. Some refactoring of
AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
possible. A few checks for "channels \in {1, 2}" are replaced with
"channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
other changes are made to be able to write and read multi-channel WAV
files.
The SDP changes are NOT included; as of this CL there is no way to set
up a multi-stream opus en/de-coder from SDP strings.
Bug: webrtc:8649
Change-Id: I1d93a9b8eecc3f6e19896ff2e2ce9b63da77a23c
Reviewed-on: https://webrtc-review.googlesource.com/c/114883
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26742}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/111750.
This time we don't use the multistream decoder unless we have to.
(Which is when #channels >2). Pros: don't make downstream projects
crash due to used up stack space, a few % more efficiency for the
typical case (because multistream adds some overhead). Cons: Messy
C-code with "union" types and #define MACROs, probably more
maintenance.
Bug: webrtc:8649
Change-Id: I4253a5e0c382f67ac7c6731dc6602a31e6779e63
Reviewed-on: https://webrtc-review.googlesource.com/c/120049
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26445}
This reverts commit 83ed89a45f4578ca07efef48e772b9aafb263163.
Reason for revert: breaks downstream project
Original change's description:
> Opus multistream.
>
> This is a backwards-compatible change. It makes WebRTC use the Opus
> multistream decoder for all Opus packets. Single-stream packets are a
> special case of multistream ones (with stream=1).
>
> The tricky parts are 'WebRtcOpus_GetMaxPlaybackRate' and
> 'WebRtcOpus_GetSurroundParameters'. GetMaxPlaybackRate is supposed to
> do what opus_encoder_ctl(encoder, OPUS_GET_MAX_BANDWIDTH(&bandwidth))
> did when we had single-stream encoders. Now there may be several
> independent encoders with possibly different BANDWIDTH. The new
> GetMaxPlaybackRate queries all of them, and returns a playback rate if
> all the encoder's rates are equal.
>
> WebRtcOpus_GetSurroundParameters is a configuration convention. It
> maps the number of channels to a multi-stream encoder/decoder
> configuration. As described in RFC 7845
> https://tools.ietf.org/html/rfc7845#section-5.1.1, a multi-stream
> encoder/decoder needs a number of streams, number of coupled streams
> and a 255-byte mapping array. The function GetSurroundParameters
> computes all of these from the number of channels. [1, 2, 4, 6, 8]
> channels are supported.
>
> Bug: webrtc:8649
> Change-Id: I271de8e387d738254d6aa53af7fcf8644a53edb5
> Reviewed-on: https://webrtc-review.googlesource.com/c/111750
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26293}
TBR=aleloi@webrtc.org,minyue@webrtc.org
Change-Id: I1002e3273b44d3cccacdba84b8c363eefd537c4b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8649
Reviewed-on: https://webrtc-review.googlesource.com/c/118201
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26306}
This is a backwards-compatible change. It makes WebRTC use the Opus
multistream decoder for all Opus packets. Single-stream packets are a
special case of multistream ones (with stream=1).
The tricky parts are 'WebRtcOpus_GetMaxPlaybackRate' and
'WebRtcOpus_GetSurroundParameters'. GetMaxPlaybackRate is supposed to
do what opus_encoder_ctl(encoder, OPUS_GET_MAX_BANDWIDTH(&bandwidth))
did when we had single-stream encoders. Now there may be several
independent encoders with possibly different BANDWIDTH. The new
GetMaxPlaybackRate queries all of them, and returns a playback rate if
all the encoder's rates are equal.
WebRtcOpus_GetSurroundParameters is a configuration convention. It
maps the number of channels to a multi-stream encoder/decoder
configuration. As described in RFC 7845
https://tools.ietf.org/html/rfc7845#section-5.1.1, a multi-stream
encoder/decoder needs a number of streams, number of coupled streams
and a 255-byte mapping array. The function GetSurroundParameters
computes all of these from the number of channels. [1, 2, 4, 6, 8]
channels are supported.
Bug: webrtc:8649
Change-Id: I271de8e387d738254d6aa53af7fcf8644a53edb5
Reviewed-on: https://webrtc-review.googlesource.com/c/111750
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26293}
This replaces the current usage of AudioProcessing::level_estimator()
in that test.
The unit tests that specifically test the level_estimator API are left
in place, until the level_estimator API itself is removed.
Bug: webrtc:9947
Change-Id: I73301c1478d2c9763bb49598a692142229102876
Reviewed-on: https://webrtc-review.googlesource.com/c/114550
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26049}
None of these scripts or files have been used in a very long time. They
are removed for the same reason, and since the data files add to the
weight of the resources folder.
Bug: webrtc:5289
Change-Id: Ia14a46aed180f286fa881fe5f60da6973a5fe027
Reviewed-on: https://webrtc-review.googlesource.com/c/109022
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25502}
The test is refitted to use the AudioProcessingStats struct to get
reference data.
The old metrics do not map entirely injectively to the new ones, so the
reference protobuf and files are updated as well.
Bug: webrtc:9535
Change-Id: I546dca2979380e03895af0077bfc77ffd24abe36
Reviewed-on: https://webrtc-review.googlesource.com/100100
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24740}
This is a reland of 404be7f302358e6be16aadeba8bc8f8aba348c0f
It adds support for reading .yuv files as well to not break anything.
Original change's description:
> Add Y4mFileReader
>
> Encapsulate logic for reading .y4m video files in a single class. We
> currently have spread out logic for opening .y4m files with partial
> parsing. This CL consolidates this logic into a single class with a well
> defined interface.
>
> Change-Id: Id61673b3c95a0053b30e95b4cf382e1c6b05fc30
> Bug: webrtc:9642
> Reviewed-on: https://webrtc-review.googlesource.com/94772
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Paulina Hensman <phensman@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24398}
TBR=phensman,phoglund
Bug: webrtc:9642
Change-Id: Idecc5ec5da767221a5f5b439989f4fe07e3b3615
Reviewed-on: https://webrtc-review.googlesource.com/97983
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24571}
Encapsulate logic for reading .y4m video files in a single class. We
currently have spread out logic for opening .y4m files with partial
parsing. This CL consolidates this logic into a single class with a well
defined interface.
Change-Id: Id61673b3c95a0053b30e95b4cf382e1c6b05fc30
Bug: webrtc:9642
Reviewed-on: https://webrtc-review.googlesource.com/94772
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24398}
For example, currently "resources/audio_coding/F02_tlm10.OUT20.sha1" would have been ignored by the pattern "**/*.OUT*".
No-Try: True
Bug: None
Change-Id: I91243a301950485cb61d5f72a00af08372ec7951
Reviewed-on: https://webrtc-review.googlesource.com/92085
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24185}
All clients who do not own their own APM turn it off by default
(in WebrtcVoiceEngine). AECM with comfort noise is a little-exercised
code path. Configurability of this setting is going away, so we're
better off disabling it by default.
Bug: webrtc:9535
Change-Id: Iba839aa18e79ae29ff20bdf6e30de77870ba4143
Reviewed-on: https://webrtc-review.googlesource.com/89583
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24078}
This CL adds helper functions to be used for the spectral features
computation. Namely, it includes the following:
- band boundaries (frequency to FFT coeffcient index)
- band energy coefficients
- log band energy coefficients
- fixed size DCT table and computation
Bug: webrtc:9076
Change-Id: I03a8799b226d986bc1e37cefd0c3039f94b5592a
Reviewed-on: https://webrtc-review.googlesource.com/73687
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23170}
RNN implementation for the AGC2 VAD that includes a fully connected
layer and a gated recurrent unit layer.
Bug: webrtc:9076
Change-Id: Ibb8b0b4e9213f09eb9dbe118bbdc94d7e8e4f91b
Reviewed-on: https://webrtc-review.googlesource.com/72060
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23101}
Functions to estimate the inverse filter via LPC and compute the LP
residual applying the inverse filter.
This CL also includes test utilities, in particular BinaryFileReader,
used to read chunks of data and optionally cast them on the fly, and
Create*Reader() functions to read resource files available at test
time.
Bug: webrtc:9076
Change-Id: Ia4793b8ad6a63cb3089ed11ddad89d1aa0b840f6
Reviewed-on: https://webrtc-review.googlesource.com/70244
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22946}
This tests NetEq with a stream encoded with Opus using it's internal
DTX/CNG.
Also adding a new resource file which is encoded using Opus with DTX.
Bug: webrtc:8488
Change-Id: Icfba5bc5dc7f9c9d0e637a90f4df674e8ba40358
Reviewed-on: https://webrtc-review.googlesource.com/26028
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20905}
This CL includes the patch from oprypin@webrtc.org, which is also applied
to the MIPS code (also affected), and the protobuf for ApmTest.Process
(audio_processing_unittest.cc), which used when WEBRTC_AUDIOPROC_FIXED_PROFILE
is set.
This change has been tested on mobile platforms.
Bug: webrtc:8200
Change-Id: Ic50a5ab57c16551397756b1fb473e1067b8e7ece
Reviewed-on: https://webrtc-review.googlesource.com/10811
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20394}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
Reason for revert:
Resetting the estimate means that we need to start gathering data from scratch again. The combination of
1) DelayBasedEstimator not reacting to overuse unless there is a valid estimate of the acknowledged bitrate, and
2) AcknowledgedBitrateEstimator needing a significant amount of time/data to obtain an provide an estimate
causes poor performance in simulations/tests. It is not clear whether this will affect real networks negatively, but I suggest reverting this to be on the safe side.
See also https://bugs.chromium.org/p/webrtc/issues/detail?id=7884
Original issue's description:
> Test and fix for huge bwe drop after alr state.
>
> BUG=webrtc:7746
>
> Review-Url: https://codereview.webrtc.org/2931873002
> Cr-Commit-Position: refs/heads/master@{#18692}
> Committed: 37aa8ba616TBR=solenberg@webrtc.org,kwiberg@webrtc.org,minyue@webrtc.org,holmer@chromium.org,philipel@webrtc.org,oprypin@webrtc.org,holmer@google.com,stefan@webrtc.org,tschumim@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7746
Review-Url: https://codereview.webrtc.org/2964213002
Cr-Commit-Position: refs/heads/master@{#18866}
This CL corrects the initialization of the AEC3, as well
as for the other submodules in the whole audio processing module
in the sense that it properly update the submodule states also
for the case when reinitialization is trigger from the render
side of the audio processing module.
Bug: chromium:736889,webrtc:7879
Change-Id: I423e963835d0c3227caa8e186b29031bcb912515
Reviewed-on: https://chromium-review.googlesource.com/549315
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18784}
Publicly available dataset of conversational speech audio recordings.
This CL includes the following:
- README.md: dataset description file, it also includes the scripts
- *.wav.sha1: hash files for each audio track in the dataset
The overall size of the wav files is ~36MB.
The primary intended use of this dataset is in combination with the conversational speech tool (see https://chromium.googlesource.com/external/webrtc/+/master/webrtc/modules/audio_processing/test/conversational_speech/), using which longer recordings with custom turn switch timing can be created.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2869833002
Cr-Commit-Position: refs/heads/master@{#18068}
