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}
Sounds better according to a MUSHRA listening test.
The computational complexity is unaffected.
An empirically estimated gain was added to compensate for the attenuation introduced by the algorithm.
There are some TODOs, which I will address in follow up CLs.
It was tested in Hangouts without headphones and highest volume, to make sure it doesn't affect the AEC.
Review URL: https://codereview.webrtc.org/1378973003
Cr-Commit-Position: refs/heads/master@{#10308}
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 used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.
This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means,
likely just the linker omitting compilation units with no incoming
references.
(This was previously landed as revisions 10046 and 10060, and got
reverted because it broke several of the Chromium FYI bots.)
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1368843003
Cr-Commit-Position: refs/heads/master@{#10127}
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}
Replaced the wav file dumping functionality in aec_core.c with the newly added corresponding macros
Added macros for logging of AEC internal data
BUG=
Review URL: https://codereview.webrtc.org/1272403003
Cr-Commit-Position: refs/heads/master@{#9808}
- 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}
These are mostly trivial changes and are separated out just to reduce the
diff on that change to the minimum possible.
Note explanatory comments on patch set 1.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1235643003
Cr-Commit-Position: refs/heads/master@{#9617}
Android has not yet finalized its libc++ build. Allow compilation with
stlport by removing several C++11 library usages.
BUG=427718,487341,webrtc:4866
R=andrew@webrtc.org
Review URL: https://codereview.webrtc.org/1250663007 .
Patch from Jared Duke <jdduke@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#9616}
- New option for computing variance that is more adaptive with lower complexity.
- Fixed related off-by-one errors.
- Added intelligibility unittests.
- Do not enhance if experiencing variance underflow.
R=andrew@webrtc.org, henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1207353002 .
Cr-Commit-Position: refs/heads/master@{#9567}
We have two histograms today that trigger on large jumps in either platform reported stream delays (WebRTC.Audio.PlatformReportedStreamDelayJump) or the system delay in the AEC (WebRTC.Audio.AecSystemDelayJump). The latter is the internal buffer size in the AEC.
The sizes of such jumps are of relevance since it can harm the AEC and even put it in a complete failure state. It is hard, not to say impossible, to tell how frequent it is.
Therefore, two complementary histograms are added; number of jumps in each metric.
This way we get a quick way to determine how often a jump occurs in general and also how frequent it is within a call.
This is solved by adding a counter for each metric.
The counter is activated either upon an event trigger or if we know for sure when the AEC is running.
Unfortunately, we can't rely on the destructor at the end of a call so we add a public API for the user to take on the action of calling it at the end of a call.
Tested locally by building ToT chromium including changes and three triggered jumps (200, 50 and 60 ms).
The stats picked up the 60 and 200 ms jumps as expected.
BUG=488124
R=asapersson@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1229443003.
Cr-Commit-Position: refs/heads/master@{#9544}
The UMA histograms WebRTC.Audio.AecSystemDelayJump and WebRTC.Audio.PlatformReportedStreamDelayJump triggers if the jump is larger than kMinDiffDelayMs.
Especially WebRTC.Audio.AecSystemDelayJump is sensitive around 50 ms differences, since the granularity is 4 ms and we can get a significant amount of hits at 52 ms.
Therefore, a change to 60 ms can make the logging more robust. The effect of not logging jumps in the interval 50-60 ms is of minor importance since they are not likely to affect the AEC performance. It's when we get values from ~100 ms and above that we should be worried.
Tested with a local ToT Chromium build where 52, 64 and 200 ms jumps were forced.
BUG=488124
TBR=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1208313003.
Cr-Commit-Position: refs/heads/master@{#9540}
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}
Sudden platform system delay jumps can hurt AEC and we have no stats that monitor these jumps. How often do they occur, and when they are reported are they accurate?
This CL logs all jumps in both the reported and actual delay.
The histogram has been tested with a chromium build where a fake jump of 200 ms was applied after 5 seconds and it was registered correctly in chrome://histograms
BUG=488124
R=henrik.lundin@webrtc.org, peah@webrtc.org
Review URL: https://codereview.webrtc.org/1213733004.
Cr-Commit-Position: refs/heads/master@{#9513}
Revert of original: https://codereview.webrtc.org/1187033005/
Changes in original:
- Added files to gyp and BUILD
- Made minor fixes to get everything to compile
and intelligibility_proc to run
- Added comments
- Auto-reformatting
New Changes:
- Added <numeric> header to intelligibility_enhancer.cc to address buildbot errors
- Switched to use WAV for i/o in intelligibility_proc.cc to address windows errors
- clean up
Note: Patch 1 duplicates Patch 7 of https://codereview.webrtc.org/1182323005/R=andrew@webrtc.org
Review URL: https://codereview.webrtc.org/1190733004.
Cr-Commit-Position: refs/heads/master@{#9486}
