Specifically, don't expect the ACM to be able to create encoders; we
have to give it an encoder that we make ourselves.
The new way of creating encoders used a 32 kbit/s bitrate
unconditionally for iSAC; I had to change it to 32 kbit/s for 16 kHz
and 56 kbit/s for 32 kHz, which is what the old way of creating
encoders has used since forever.
I also had to change some test expectations on Opus, because the new
way defaults to 32 kbit/s for mono and 64 kbit/s for stereo (which I
believe to be correct), while the old way defaults to 64 kbit/s in
both cases.
Bug: webrtc:8396
Change-Id: I3aab944175a8e27f4c63380e822b27e839bba7f2
Reviewed-on: https://webrtc-review.googlesource.com/94540
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24375}
This CL also excludes several codec mappings depending on compile-time flags.
Bug: webrtc:9289
Change-Id: I1a9183f88378307925b747576a5513e54be3782e
Reviewed-on: https://webrtc-review.googlesource.com/93462
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24259}
Specifically, don't expect the ACM to be able to create encoders; we
have to give it an encoder that we make ourselves.
To make it work, I had to add support for the "ptime" parameter to the
L16 codec.
Bug: webrtc:8396
Change-Id: I3869422882611d2eed65d6c849ea7cd3ad6bd126
Reviewed-on: https://webrtc-review.googlesource.com/87423
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24217}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I6a7d4964723a5e195189aac30a83d9e924e61dd7
Reviewed-on: https://webrtc-review.googlesource.com/89743
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24053}
WebRTC internal headers are always included starting from the root
(e.g. #include "modules/audio_coding/..."), so there is no need to
specify the include_dirs removed by this CL.
Bug: webrtc:9538
Change-Id: I91e70508c67020bbf70304df5e48ca757ad43221
Reviewed-on: https://webrtc-review.googlesource.com/89385
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24026}
Building NetEqEventLogInput requires protobuf support, while building
NetEqRtpDumpInput located in the same file does not. This makes both
classes unusable when protobuf support is lacking. With this CL, the
NetEqEventLogInput is broken out into separate files, to allow usage
of NetEqRtpDumpInput even when protobufs are not supported.
Bug: webrtc:9421
Change-Id: I55efec4ec259713654566cdaa00d2e34c5e9a60f
Reviewed-on: https://webrtc-review.googlesource.com/84587
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23803}
This is a reland of 80c4cca4915dbc6094a5bfae749f85f7371eadd1
Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
>
> The playout modes other than Normal have not been reachable for a long
> time, other than through tests. It is time to deprecate them.
>
> The only meaningful use was that Fax mode was sometimes set from
> tests, in order to avoid time-stretching operations (accelerate and
> pre-emptive expand) from messing with the test results. With this CL,
> a new config is added instead, which lets the user specify exactly
> this: don't do time-stretching.
>
> As a result of Fax and Off modes being removed, the following code
> clean-up was done:
> - Fold DecisionLogicNormal into DecisionLogic.
> - Remove AudioRepetition and AlternativePlc operations, since they can
> no longer be reached.
>
> Bug: webrtc:9421
> Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
> Reviewed-on: https://webrtc-review.googlesource.com/84123
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23704}
Bug: webrtc:9421
Change-Id: Ice351b635788167f2971b26470f73a5e5fa1a240
Reviewed-on: https://webrtc-review.googlesource.com/86543
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23799}
fft.c is third party library and have to be moved to proper third_party
directory. So this CL will extract it to separate gn target to be able
then to move it to proper location.
Bug: webrtc:8366
Change-Id: I228ebab3c821aa7095f7aa460c23c2ea0fb98f01
Reviewed-on: https://webrtc-review.googlesource.com/85640
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23753}
This reverts commit 80c4cca4915dbc6094a5bfae749f85f7371eadd1.
Reason for revert: Breaks downstream tests.
Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
>
> The playout modes other than Normal have not been reachable for a long
> time, other than through tests. It is time to deprecate them.
>
> The only meaningful use was that Fax mode was sometimes set from
> tests, in order to avoid time-stretching operations (accelerate and
> pre-emptive expand) from messing with the test results. With this CL,
> a new config is added instead, which lets the user specify exactly
> this: don't do time-stretching.
>
> As a result of Fax and Off modes being removed, the following code
> clean-up was done:
> - Fold DecisionLogicNormal into DecisionLogic.
> - Remove AudioRepetition and AlternativePlc operations, since they can
> no longer be reached.
>
> Bug: webrtc:9421
> Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
> Reviewed-on: https://webrtc-review.googlesource.com/84123
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23704}
TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org
Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9421
Reviewed-on: https://webrtc-review.googlesource.com/84680
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23706}
The playout modes other than Normal have not been reachable for a long
time, other than through tests. It is time to deprecate them.
The only meaningful use was that Fax mode was sometimes set from
tests, in order to avoid time-stretching operations (accelerate and
pre-emptive expand) from messing with the test results. With this CL,
a new config is added instead, which lets the user specify exactly
this: don't do time-stretching.
As a result of Fax and Off modes being removed, the following code
clean-up was done:
- Fold DecisionLogicNormal into DecisionLogic.
- Remove AudioRepetition and AlternativePlc operations, since they can
no longer be reached.
Bug: webrtc:9421
Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
Reviewed-on: https://webrtc-review.googlesource.com/84123
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23704}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'modules/audio_coding'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8
Reviewed-on: https://webrtc-review.googlesource.com/84130
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23659}
It appears to have been created in mid-2013, and hasn't been changed
since except to keep the compiler happy when surrounding code changed.
It crashes when I try to run it without arguments, and no one
remembers how to use it.
Bug: webrtc:8396
Change-Id: I2eae36cf468f28c5bf05c85e6a3aaeebc48a1ffc
Reviewed-on: https://webrtc-review.googlesource.com/83581
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23618}
Since it isn't being run by the bots, it has bit rotted; when I try to
run it manually, it fails with a long list of error messages:
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 995
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 996
>>> Error Enabling VAD <<<
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 995
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 996
>>> Error Enabling DTX <<<
>>> Error Enabling VAD <<<
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 995
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 996
>>> Error Enabling VAD <<<
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 995
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 996
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 985
...and so on.
Bug: webrtc:8396
Change-Id: Id8f1e01a751b4bb3527702b7b7a4986ce0abb378
Reviewed-on: https://webrtc-review.googlesource.com/81745
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23542}
The audio processing code is using parts of the iSAC codec to do voice
activity detection (VAD), but it's undesirable for it to pull in the
entire iSAC codec as a dependency. So this CL factors out the parts of
iSAC that's needed for VAD to a separate build target.
Bug: webrtc:8396
Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e
Reviewed-on: https://webrtc-review.googlesource.com/69640
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23110}
Only specially taggged targets may transitively depend on poisonous
targets. We first apply it to audio codecs.
This makes it much clearer exactly what parts of the code still have
dependencies on the audio codecs (and we want to eventually get rid of
pretty much all of them).
Bug: webrtc:8396, webrtc:9121
Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43
Reviewed-on: https://webrtc-review.googlesource.com/69520
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22979}
The target modules/audio_coding:isac_neon needs to link with
transform_tables.c but adding a dependency between isac_neon and
isac_fix_c creates a circular dependency.
This CL moves transform_tables.c to isac_fix_common (which is already a
dependency of isac_neon).
Bug: None
Change-Id: I4135ec772b0017e77f1411e9a8093b495220c636
Reviewed-on: https://webrtc-review.googlesource.com/71581
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22968}
PCMFile.cc uses RTC_DCHECK. include and depend on rtc_base:checks target directly
change usage of value_or by using explicit constructor instead of implicit
Bug: webrtc:9078
Change-Id: I63c596b8a05b387e56df846b15c33a605fbad4e6
Reviewed-on: https://webrtc-review.googlesource.com/69985
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22897}
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.
Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
These two new histograms relate to the packet-loss concealment that
happens when audio packets are lost or late for decoding, and the
NetEq must resort to extrapolating audio from the previously
decoded data.
Bug: webrtc:9126
Change-Id: I99cc97e653169fb742da0092653ab99fd10e5d7b
Reviewed-on: https://webrtc-review.googlesource.com/67861
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22812}
So that we don't have to be capable of creating one ourselves, which
requires a dependency on the audio decoders.
BUG=webrtc:5801, webrtc:8396
Change-Id: I80749ec3b86cba73994307046d05964f59167d44
Reviewed-on: https://webrtc-review.googlesource.com/18440
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22774}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
BUG=webrtc:8445
NOPRESUBMIT=true
Change-Id: Ie2879aca5fc1667e4222499d2a8fc2bba9ae2425
Reviewed-on: https://webrtc-review.googlesource.com/21328
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22587}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
BUG=webrtc:8445
Change-Id: I440974da4d347b09ff042478720d7983056b62b9
Reviewed-on: https://webrtc-review.googlesource.com/21226
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22579}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
Bug: webrtc:8445
NOPRESUBMIT=true
Change-Id: I30d01fcb9cbe1427a7703a3cdd7befae751066b5
Reviewed-on: https://webrtc-review.googlesource.com/21982
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22550}
WebRTC internal code should always used include paths that starts
from the root of the project and that clearly identify the header file.
This allows 'gn check' to actually keep dependencies under control
because 'gn check' cannot enforce anything if the include path
is not fully qualified (starting from the root of the project).
Bug: webrtc:8815
Change-Id: I23fb4fed0c27a4d98bea360315b959af843587bc
Reviewed-on: https://webrtc-review.googlesource.com/46101
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21847}
The original rtc_event_log_api is refactored to a pure API target plus
multiple targets coupled with WebRTC implementations.
Bug: None
Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
Reviewed-on: https://webrtc-review.googlesource.com/43247
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#21811}
