This will make it align with protoc tools that use the relative
path from the project root to the files in the output path.
Having this, no hacks will need to be applied downstream.
TBR=henrik.lundin@webrtc.org
NOTRY=True
Review URL: https://codereview.webrtc.org/1673263002
Cr-Commit-Position: refs/heads/master@{#11540}
The level of the error signal after linear echo cancellation was based on non-buffered signal while that of the near-end and far-end signal based on buffered signal. This discrepancy made the comparison of them unfair.
This CL is to make calculating the error level rely on the same buffering.
BUG=
Review URL: https://codereview.webrtc.org/1510873004
Cr-Commit-Position: refs/heads/master@{#11408}
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests). Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks
All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1534193008
Cr-Commit-Position: refs/heads/master@{#11191}
Macro incorrectly displays DISABLED_ON_ANDROID in test names for
parameterized tests under --gtest_list_tests, causing tests to be
disabled on all platforms since they contain the DISABLED_ prefix rather
than their expanded variants.
This expands the macro variants to inline if they're disabled or not,
and removes building some tests under configurations where they should
fail, instead of building them but disabling them by default.
The change also removes gtest_disable.h as an unused include from many
other files.
BUG=webrtc:5387, webrtc:5400
R=kjellander@webrtc.org, phoglund@webrtc.orgTBR=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1547343002 .
Cr-Commit-Position: refs/heads/master@{#11150}
For applications with a strict filesize limit for debug files,
I added an option to specify a maximum filesize for AEC dumps. An
existing unit test is extended to check that the feature works as
advertised.
BUG=webrtc:4741
TBR=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1413483003
Cr-Commit-Position: refs/heads/master@{#11081}
We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
This is the second revert. The first attempt in https://codereview.webrtc.org/1423693008/
was missing a subtle curly brace caused by a merge conflict.
I'm going to let this one go through the CQ.
Reason for revert:
This breaks iOS GYP generation as described on http://www.webrtc.org/native-code/ios
I'm going to drive getting the build_with_libjingle=1 setting removed from the bots to match the official instructions.
See https://code.google.com/p/webrtc/issues/detail?id=4653 for more context, as this is exactly what that issue tries to solve.
Original issue's description:
> Add aecdump support to audioproc_f.
>
> Add a new interface to abstract away file operations. This CL temporarily
> removes support for dumping the output of reverse streams. It will be easy to
> restore in the new framework, although we may decide to only allow it with
> the aecdump format.
>
> We also now require the user to specify the output format, rather than
> defaulting to the input format.
>
> TEST=Bit-exact output to the previous audioproc_f version using an input wav
> file, and to the legacy audioproc using an aecdump file.
>
> Committed: https://crrev.com/bdafe31b86e9819b0adb9041f87e6194b7422b08
> Cr-Commit-Position: refs/heads/master@{#10460}
TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
BUG=
Review URL: https://codereview.webrtc.org/1412963007
Cr-Commit-Position: refs/heads/master@{#10532}
Reason for revert:
Oh dear, this broke compilation.
I guess more was built on top of this CL before I reverted it.
Reverting now for futher investigation (and re-land using CQ)
Original issue's description:
> Revert of Add aecdump support to audioproc_f. (patchset #8 id:200001 of https://codereview.webrtc.org/1409943002/ )
>
> Reason for revert:
> This breaks iOS GYP generation as described on http://www.webrtc.org/native-code/ios
> I'm going to drive getting the build_with_libjingle=1 setting removed from the bots to match the official instructions.
>
> See https://code.google.com/p/webrtc/issues/detail?id=4653 for more context, as this is exactly what that issue tries to solve.
>
> Original issue's description:
> > Add aecdump support to audioproc_f.
> >
> > Add a new interface to abstract away file operations. This CL temporarily
> > removes support for dumping the output of reverse streams. It will be easy to
> > restore in the new framework, although we may decide to only allow it with
> > the aecdump format.
> >
> > We also now require the user to specify the output format, rather than
> > defaulting to the input format.
> >
> > TEST=Bit-exact output to the previous audioproc_f version using an input wav
> > file, and to the legacy audioproc using an aecdump file.
> >
> > Committed: https://crrev.com/bdafe31b86e9819b0adb9041f87e6194b7422b08
> > Cr-Commit-Position: refs/heads/master@{#10460}
>
> TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/d279941bb54bfdc6e7324bf36cac76581474b96d
> Cr-Commit-Position: refs/heads/master@{#10523}
TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1419953010
Cr-Commit-Position: refs/heads/master@{#10524}
Reason for revert:
This breaks iOS GYP generation as described on http://www.webrtc.org/native-code/ios
I'm going to drive getting the build_with_libjingle=1 setting removed from the bots to match the official instructions.
See https://code.google.com/p/webrtc/issues/detail?id=4653 for more context, as this is exactly what that issue tries to solve.
Original issue's description:
> Add aecdump support to audioproc_f.
>
> Add a new interface to abstract away file operations. This CL temporarily
> removes support for dumping the output of reverse streams. It will be easy to
> restore in the new framework, although we may decide to only allow it with
> the aecdump format.
>
> We also now require the user to specify the output format, rather than
> defaulting to the input format.
>
> TEST=Bit-exact output to the previous audioproc_f version using an input wav
> file, and to the legacy audioproc using an aecdump file.
>
> Committed: https://crrev.com/bdafe31b86e9819b0adb9041f87e6194b7422b08
> Cr-Commit-Position: refs/heads/master@{#10460}
TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1423693008
Cr-Commit-Position: refs/heads/master@{#10523}
This test is to verify that the debug dump can perfectly reproduce APM states if the recording is made from the first input sample.
BUG=
Review URL: https://codereview.webrtc.org/1393353003
Cr-Commit-Position: refs/heads/master@{#10506}
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
Add a new interface to abstract away file operations. This CL temporarily
removes support for dumping the output of reverse streams. It will be easy to
restore in the new framework, although we may decide to only allow it with
the aecdump format.
We also now require the user to specify the output format, rather than
defaulting to the input format.
TEST=Bit-exact output to the previous audioproc_f version using an input wav
file, and to the legacy audioproc using an aecdump file.
Review URL: https://codereview.webrtc.org/1409943002
Cr-Commit-Position: refs/heads/master@{#10460}
The AEC dump was not self-contented enough in the sense that APM configuration is missing, and therefore, given an AEC dump, it is sometimes not clear how to reproduce problems.
This CL tries to address the problem.
Note that this cannot guarantee a perfect reproduction in all cases. Dumping from the middle of a call makes the initial states unknown and thus may make the result non-reproducible.
BUG=
TEST= 1. new dump in Chromium and unpack
2. unpack old dump
R=andrew@webrtc.org, peah@webrtc.org
Review URL: https://codereview.webrtc.org/1348903004 .
Cr-Commit-Position: refs/heads/master@{#10155}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
- Integrates intelligibility into audio_processing.
- Allows modification of reverse stream if intelligibility enabled.
- Makes intelligibility available in audioproc_float test.
- Adds reverse stream processing to audioproc_float.
- (removed) Makes intelligibility toggleable in real time in voe_cmd_test.
- Cleans up intelligibility construction, parameters, constants and dead code.
TBR=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1234463003
Cr-Commit-Position: refs/heads/master@{#9713}
Reason for revert:
Breaks Chromium FYI content_browsertest on all platforms. The testcase that fails is WebRtcAecDumpBrowserTest.CallWithAecDump.
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/19388
Sample output:
[ RUN ] WebRtcAecDumpBrowserTest.CallWithAecDump
Xlib: extension "RANDR" missing on display ":9".
[4:14:0722/211548:1282124453:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: ISAC/48000/1 (105)
[4:14:0722/211548:1282124593:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMU/8000/2 (110)
[4:14:0722/211548:1282124700:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMA/8000/2 (118)
[4:14:0722/211548:1282124815:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: G722/8000/2 (119)
[19745:19745:0722/211548:1282133667:INFO:CONSOLE(64)] "Looking at video in element remote-view-1", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
[19745:19745:0722/211548:1282136892:INFO:CONSOLE(64)] "Looking at video in element remote-view-2", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
../../content/test/webrtc_content_browsertest_base.cc:62: Failure
Value of: ExecuteScriptAndExtractString( shell()->web_contents(), javascript, &result)
Actual: false
Expected: true
Failed to execute javascript call({video: true, audio: true});.
From javascript: (nothing)
When executing 'call({video: true, audio: true});'
../../content/test/webrtc_content_browsertest_base.cc:75: Failure
Failed
../../content/browser/media/webrtc_aecdump_browsertest.cc:26: Failure
Expected: (base::kNullProcessId) != (*id), actual: 0 vs 0
../../content/browser/media/webrtc_aecdump_browsertest.cc:95: Failure
Value of: GetRenderProcessHostId(&render_process_id)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:99: Failure
Value of: base::PathExists(dump_file)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:101: Failure
Value of: base::GetFileSize(dump_file, &file_size)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:102: Failure
Expected: (file_size) > (0), actual: 0 vs 0
[ FAILED ] WebRtcAecDumpBrowserTest.CallWithAecDump, where TypeParam = and GetParam() = (361 ms)
Original issue's description:
> Allow more than 2 input channels in AudioProcessing.
>
> The number of output channels is constrained to be equal to either 1 or the
> number of input channels.
>
> R=aluebs@webrtc.org, andrew@webrtc.org, pbos@webrtc.org
>
> Committed: c204754b7aTBR=andrew@webrtc.org,aluebs@webrtc.org,ajm@chromium.org,pbos@chromium.org,pbos@webrtc.org,mgraczyk@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1253573005
Cr-Commit-Position: refs/heads/master@{#9621}
We use this Config struct for enabling/disabling the delay agnostic
AEC. This change renames it to DelayAgnostic for readability reasons.
NOTE: The logic is reversed in this CL. The old ReportedDelay config
turned DA-AEC off, while the new DelayAgnostic turns it on.
The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, ReportedDelay is disabled or DelayAgnostic is enabled, DA-AEC
is engaged in APM.
BUG=webrtc:4651
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1211053006
Cr-Commit-Position: refs/heads/master@{#9531}
(This reverts commit 3fbf3f8841b5460503fb646eaedcb063620434a8.)
The original submission was reverted because it broke the Chrome build. This is fixed in patch set 2 of this change by keeping the old MediaConstraintsInterface string kExperimentalEchoCancellation. It will be removed once the Chrome code has been updated.
Original description:
"We use this Config struct for enabling/disabling Extended filter mode in AEC. This change renames it to ExtendedFilter for readability reasons. The corresponding media constraint is also renamed to kExtendedFilterEchoCancellation.
The old Config is kept in parallel with the new during a transition period. This is to avoid problems with API breakages. During this period, if any of the two Configs are enabled, the extended filter mode is engaged in APM. That is, the two Configs are combined with an "OR" operation.
This change also renames experimental_aec in AudioOptions to extended_filter_aec."
BUG=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1151573021.
Cr-Commit-Position: refs/heads/master@{#9401}
We use this Config struct for enabling/disabling Extended filter mode
in AEC. This change renames it to ExtendedFilter for readability
reasons. The corresponding media constraint is also renamed to
kExtendedFilterEchoCancellation.
The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, if any of the two Configs are enabled, the extended filter
mode is engaged in APM. That is, the two Configs are combined with an
"OR" operation.
This change also renames experimental_aec in AudioOptions to extended_filter_aec.
BUG=4696
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54659004
Cr-Commit-Position: refs/heads/master@{#9378}
Because of the Finch experiment, this will not affect Chrome's behaviour at all.
The SNRs in AudioProcessingTest.Formats were only increased to the next multiple of 5.
BUG=webrtc:3146
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43359004
Cr-Commit-Position: refs/heads/master@{#9263}
This adds functionality from audioproc_float. The geometry parsing code
is now shared from test_utils.h. I removed the "mic_spacing" flag from
audioproc_float because it's a redundancy that I suspect isn't very
useful.
Includes a cleanup of the audio_processing test utils. They're now
packaged in targets, with the protobuf-using ones split out to avoid
requiring users to depend on protobufs.
pcm_utils is no longer needed and removed.
The primary motivation for this CL is that AudioProcessing currently
doesn't support more than two channels and we'd like a way to pass
more channels to the beamformer.
R=aluebs@webrtc.org, mgraczyk@chromium.org
Review URL: https://webrtc-codereview.appspot.com/50899004
Cr-Commit-Position: refs/heads/master@{#9157}
For quick and easy aecdump verifiation storing data as text speeds up the issue tracking process, since anyone can simply view values like mic volume.
BUG=4609
TESTED=verified unpacking an aecdump with flag --txt stores that data in text files
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50849004
Cr-Commit-Position: refs/heads/master@{#9142}
The way SetExtraOptions() is used today only applies for any one configuration change. The correct way is to set it after all flags have been scanned.
The prefered way to solve this is to use gflags and scan once, followed by applying the configuration when creating audio_processing. This is what is done in the new test tool audioproc_float.cc, but there are still some things left to do before we can replace this one.
BUG=N/A
TESTED=locally
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45279004
Cr-Commit-Position: refs/heads/master@{#9097}
