a87c398a41
Move audio_codec_speed_tests into include_tests==1 condition.
...
I made a mistake in https://webrtc-codereview.appspot.com/37859004
and moved this target out of the include_tests==1 condition.
This moves it back in.
TBR=tina.legrand@webrtc.org
BUG=4185
Review URL: https://webrtc-codereview.appspot.com/33139004
Cr-Commit-Position: refs/heads/master@{#8198}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8198 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 10:39:45 +00:00
2d2a1f9f05
Remove <(webrtc_root) from source file entries.
...
This required to move the AGC tools source files
into webrtc/tools and create a new agc_test_utils target.
Since audio_codec_speed_tests.gypi referenced sources above,
the best approach I could come up with was to add an audio_coding.gypi
file at a higher level and move the targets in there (+ the includes from
modules.gyp which is an improvement IMO).
I also added a PRESUBMIT.py check to prevent new source
entries being added with <(webrtc_root) in the path.
BUG=4185
R=andrew@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37859004
Cr-Commit-Position: refs/heads/master@{#8197}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8197 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 10:24:44 +00:00
43c883954f
Allow rtp packet history to dynamically expand in size.
...
When using the paced sender, packets will be put into the rtp packet
history and then retreived from there again when it is time to send.
In some cases (low send bitrate and very large frames created) this
may overflow, causing packets to be overwritten in the packet history
before they have been sent.
Check this condition and expand history size if needed.
This is primarily triggered during screenshare, when
switching to a large picture with lots of high frequency
details in it.
BUG=4171
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34879004
Cr-Commit-Position: refs/heads/master@{#8195}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8195 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 09:09:41 +00:00
f17ee9c709
Add case to ApmTest.Process to test the extended filter mode
...
R=andrew@webrtc.org , bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40509004
Cr-Commit-Position: refs/heads/master@{#8192}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8192 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 00:04:18 +00:00
035e9123e9
Move channel_buffer.{h,cc} to common_audio.
...
In https://code.google.com/p/webrtc/source/detail?r=8166
I added a check preventing GYP files from referencing
sources above their directory level.
This CL fixes the disallowed reference added in
https://code.google.com/p/webrtc/source/detail?r=8157
by moving channel_buffer.{h,cc} to common_audio for real.
BUG=4185
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35939004
Cr-Commit-Position: refs/heads/master@{#8190}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8190 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 19:57:44 +00:00
7d2b6a9346
Enable Clang warning implicit-fallthrough and annotate the code.
...
BUG=4242
R=henrik.lundin@webrtc.org , stefan@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34899004
Cr-Commit-Position: refs/heads/master@{#8187}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8187 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 18:38:13 +00:00
664ccb7d8d
Reland r8125: Modify some tests to never use DTX disable mode
...
DTX disable mode will be removed as a part of the ACM redesign work.
This CL effectively reverts r8129, and relands r8125, but now using
assert instead of DCHECK.
COAUTHOR:kwiberg@webrtc.org
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37839004
Cr-Commit-Position: refs/heads/master@{#8185}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8185 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 14:49:12 +00:00
37c0559c1e
Notify jitter buffer about received FEC packets (to avoid sending NACK request for these packets).
...
Don't copy codec specific header for empty packets in the jitter buffer.
BUG=3135
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37659004
Cr-Commit-Position: refs/heads/master@{#8184}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8184 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 13:58:40 +00:00
4aecd008dd
Add support for 40 and 60 ms frames to AudioEncoderIlbc
...
BUG=3926
COAUTHOR:kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37789004
Cr-Commit-Position: refs/heads/master@{#8182}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8182 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 13:16:44 +00:00
2a6558c2a5
Make sure ByteReader<T>::Read* is properly constified.
...
Also, start using it in real code...
BUG=
R=holmer@google.com , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37809004
Cr-Commit-Position: refs/heads/master@{#8181}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8181 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 12:38:16 +00:00
9b64a6edd7
Adjust parameter in videoprocessor_integrationtest for VP9.
...
TBR=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/35919004
Cr-Commit-Position: refs/heads/master@{#8178}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8178 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 23:59:16 +00:00
dc8a9da386
Adjust qp-max settinhg in VP9 wrapper.
...
More closely matches the qp-max setting used in VP8.
TBR=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/39709004
Cr-Commit-Position: refs/heads/master@{#8177}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8177 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 23:08:39 +00:00
8bb32d600b
Minor updates to AudioEncoderCng
...
Removing sample_rate_hz_ from AudioEncoderCng and from the config
struct. The sample rate will now be read from the underlying speech
codec.
BUG=3926
COAUTHOR:kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40559004
Cr-Commit-Position: refs/heads/master@{#8173}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8173 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 20:54:22 +00:00
478cedc055
Add new methods to AudioEncoder interface
...
The following three methods are added:
rtp_timestamp_rate_hz()
SetTargetBitrate()
SetProjectedPacketLossRate()
Default implementations are provided, and a few overrides are
implemented. AudioEncoderCopyRed and AudioEncoderCng propagate the new
methods to the underlying speech codec.
BUG=3926
COAUTHOR:kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34049004
Cr-Commit-Position: refs/heads/master@{#8171}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8171 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 18:25:40 +00:00
5614cf16e7
audio_processing: Use fixed aggregation window in delay metrics
...
Previously, the delay estimate history was reset every time the metrics were pulled. This required all clients to be on the same thread and make use of one call.
Now we use a fixed aggregation window of one second and when a client pulls the metrics you get the latest value.
Under certain circumstances like tests you would like to have the aggregation window set to the recording length. We therefore turn on the fixed aggregation window after the first call.
BUG=2994
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38759004
Cr-Commit-Position: refs/heads/master@{#8170}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8170 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 18:10:27 +00:00
273fbbb921
Update StreamDataCounter with FEC bytes.
...
Add histograms stats for send/receive FEC bitrate:
- "WebRTC.Video.FecBitrateReceivedInKbps"
- "WebRTC.Video.FecBitrateSentInKbps"
Correct media payload bytes in StreamDataCounter to not include FEC bytes.
Fix stats for rtcp packets sent/received per minute (regression from r7910).
BUG=crbug/419657
R=holmer@google.com , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8164 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 12:17:29 +00:00
70117a83d4
AEC: Implements a new function for calculating delay metrics
...
Two new member variables have been added and the code for calculating the delay metrics have been moved to a function.
BUG=2994
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8163 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 11:30:54 +00:00
041035b390
Add an AudioRingBuffer class wrapper for the ring_buffer.h C interface.
...
Integrate it in Blocker to demonstrate use.
TEST=beamforming sounds good.
R=aluebs@webrtc.org , mgraczyk@chromium.org , sahark@google.com
Review URL: https://webrtc-codereview.appspot.com/36799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8157 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 21:23:53 +00:00
4dba2e98a2
Consolidate anonymous namespace content and file-static methods to all be in the
...
anonymous namespace, in preparation for refactoring a few of the functions a
little.
No code change.
BUG=none
TEST=none
R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8155 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 19:59:32 +00:00
d7e34e1086
Make it easier to use external libyuv + cleanup GYP files.
...
It is now easier to use an external libyuv library.
Fix some GYP errors.
Remove the temporary webrtc_base target (depends on
https://codereview.chromium.org/865603002/ being landed
first).
BUG=4185
R=andresp@webrtc.org , andrew@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8154 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 19:17:26 +00:00
38d11b8529
Enable encoder multi-threading for VP9.
...
R=stefan@webrtc.org
TBR=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/41489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8150 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 15:21:36 +00:00
b6fab2b1cd
Introduce rtc::CheckedDivExact
...
Use the new method to replace local ones in AudioEncoder{Opus,Isac}.
COAUTHOR:kwiberg@webrtc.org
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8148 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 11:08:53 +00:00
73ee4537be
Switch to use range based loops in the BWE simulation framework.
...
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8135 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-23 08:29:52 +00:00
ff108fe508
Revert 8125 "Modify some tests to never use DTX disable mode"
...
Broke compile on the Chromium FYI bots:
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/3483
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac/builds/16028
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/14293
Error:
In file included from ../../third_party/webrtc/voice_engine/channel.cc:13:
In file included from ../../third_party/webrtc/base/checks.h:22:
In file included from ../../third_party/webrtc/overrides/webrtc/base/logging.h:35:
../../base/logging.h:367:9:error: 'LOG' macro redefined [-Werror,-Wmacro-redefined]
#define LOG(severity) LAZY_STREAM(LOG_STREAM(severity), LOG_IS_ON(severity))
^
../../third_party/webrtc/system_wrappers/interface/logging.h:123:9: note: previous definition is here
#define LOG(sev) \
^
In file included from ../../third_party/webrtc/voice_engine/channel.cc:13:
In file included from ../../third_party/webrtc/base/checks.h:22:
../../third_party/webrtc/overrides/webrtc/base/logging.h:189:9:error: 'LOG_V' macro redefined [-Werror,-Wmacro-redefined]
#define LOG_V(sev) DIAGNOSTIC_LOG(sev, NONE, 0)
^
../../third_party/webrtc/system_wrappers/interface/logging.h:129:9: note: previous definition is here
#define LOG_V(sev) \
^
2 errors generated.
> Modify some tests to never use DTX disable mode
>
> DTX disable mode will be removed as a part of the ACM redesign work.
>
> COAUTHOR:kwiberg@webrtc.org
>
> R=henrika@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/34769004
TBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8129 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 19:02:03 +00:00
043db24767
Modify some tests to never use DTX disable mode
...
DTX disable mode will be removed as a part of the ACM redesign work.
COAUTHOR:kwiberg@webrtc.org
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8125 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 13:30:58 +00:00
e5251ad63c
Integrate send-side BWE into simulation framework.
...
BUG=4173
R=mflodman@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8123 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 10:10:53 +00:00
cfd82dfc11
Split packets/bytes in StreamDataCounter into RtpPacketCounter struct.
...
Prepares for adding FEC bytes to the StreamDataCounter.
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8122 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 09:39:59 +00:00
3dd33a6787
Fix bug in thresholds for bitrate probing and adjust thresholds to allow a larger dispersion and concentration for successful probes.
...
BUG=crbug:425925
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8121 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 09:12:23 +00:00
fbd37bd737
Make iSAC SWB own its decoder
...
A bug in the ACM codec database caused iSAC-swb to behave differently
from iSAC-wb and -fb. With this fix, all iSAC codecs behave the same
with respect to decoder ownership.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8120 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 08:16:29 +00:00
e65d9d974c
Fix an unitialized variable warning.
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35819004
Patch from Sebastien Marchand <sebmarchand@chromium.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8118 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 22:05:12 +00:00
c429b824b3
GN: Prepare to remove webrtc_base target
...
Keep the webrtc_base target temporarily while waiting for
Chromium to pick up this revision. Then we'll update Chromium
and remove the webrtc_base target for real.
This should have been a part of https://code.google.com/p/webrtc/source/detail?r=7140
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8117 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 20:22:33 +00:00
c78d81ae89
Re-land "Support 48kHz in AEC"
...
Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz. The only difference is that there is no comfort noise added in this band. Could not test how this sounds because there are no aecdumps with 48kHz sample rate as nfar as I know.
Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.
Original: https://webrtc-codereview.appspot.com/28319004/
Reverted: https://webrtc-codereview.appspot.com/33949004/
BUG=webrtc:3146
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8116 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 19:10:55 +00:00
e81c5d6d7e
Fix TransientDetectorTest in modules_unittests on Android ARM64
...
BUG=webrtc:4200
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8115 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 18:01:28 +00:00
11af039590
Disable AcmSenderBitExactnessOldApi.Opus_stereo_20ms_voip on ARM64.
...
BUG=4199
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8114 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 14:22:39 +00:00
df7b65ba01
Change CreateOrGetReportBlockInformation to have one return path.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8113 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 13:07:04 +00:00
9ffd8fe96b
Indentation changes.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8107 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 08:22:50 +00:00
7dba7860c7
Setting Opus target application.
...
This CL is to allow to set Opus target application at the creation of an encoder.
According to Opus spec, there are three applications:
OPUS_APPLICATION_VOIP
OPUS_APPLICATION_AUDIO
OPUS_APPLICATION_RESTRICTED_LOWDELAY
BUG=
R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8103 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 16:01:50 +00:00
853049fa30
Move internal capture+render to build_with_chromium==0 condition
...
This will avoid errors related to DirectX not being found
for Chromium bots (mainly GN, but it's safest to do the same
changes for GYP since they also make sense there as GYP generation
go slightly faster without having to process those targets).
Thanks to vchigrin@yandex-team.ru for originally suggesting
this being fixed in
https://webrtc-codereview.appspot.com/37639004/
TESTED=
Successfully ran:
webrtc/build/gyp_webrtc
webrtc/build/gyp_webrtc -Dbuild_with_chromium=1
and trybots.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8102 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 11:40:45 +00:00
ee0c100d54
Revert 8080 "Support 48kHz in AEC"
...
> Support 48kHz in AEC
>
> Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz. The only difference is that there is no comfort noise added in this band. Could not test how this sounds because there are no aecdumps with 48kHz sample rate as nfar as I know.
> Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.
>
> BUG=webrtc:3146
> R=andrew@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/28319004
TBR=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8100 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 10:22:49 +00:00
9691b36995
Cleanup for Rtp Rtcp API test.
...
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8098 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 05:42:52 +00:00
474e36e623
Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps.
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The previous CL was reverted for two reasons:
- Added a static initializer because std::string.
- Landed before the corresponding chromium CL, which has now been landed.
BUG=crbug:425925
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8094 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 15:44:47 +00:00
a32d15448d
Disable tests failing on Android ARM64 (Nexus9).
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BUG=4198,4199,4200
TBR=andrew@webrtc.org
TESTED=Printed using #pragma message to check that the define was properly used.
Review URL: https://webrtc-codereview.appspot.com/33919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8090 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 12:46:01 +00:00
a1aea10af2
Revert r8076 "Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps."
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TBR=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8085 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-16 13:52:52 +00:00
4ba1e44ff0
Remove unnecessary remote bitrate estimator build rule which serves no purpose.
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BUG=4185
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8084 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-16 07:50:17 +00:00
64d3c4b9ac
Support 48kHz in AEC
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Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz. The only difference is that there is no comfort noise added in this band. Could not test how this sounds because there are no aecdumps with 48kHz sample rate as nfar as I know.
Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.
BUG=webrtc:3146
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8080 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 19:52:05 +00:00
d82f55d2a7
Only adapt AGC when the desired signal is present
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Take the 50% quantile of the mask and compare it to certain threshold to determine if the desired signal is present. A hold is applied to avoid fast switching between states.
is_signal_present_ has been plotted and looks as expected. The AGC adaptation sounds promising, specially for the cases when the speaker fades in and out from the beam direction.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28329005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8078 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 18:07:21 +00:00
3e42a8a56a
Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps.
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BUG=crbug:425925
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8076 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 14:45:27 +00:00
1f67b53c88
Remove dual stream functionality in ACM
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This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. With this change, there is no longer need for the
ProcessDualStream method, which is removed. Consequently, the method
ProcessSingleStream is renamed to Process.
BUG=3520
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8074 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 09:36:30 +00:00
0800db74b9
Add percentage of fec packets and recovered media packets to histogram stats:
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- "WebRTC.Video.ReceivedFecPacketsInPercent"
- "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec"
BUG=crbug/419657
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8072 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 07:40:20 +00:00
6c3855258d
Add WebRtcIsacfix_AllpassFilter2FixDec16Neon()'s intrinsics version.
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This intrinsics version gives bit-exact result as the current C
code. And the performance is 14% better than current assembly
neon version, 3.4 times faster than current C version. The test runs
under Cortex-a53 aarch32 mode, other cpu should give similar performance
result.
Change-Id: Icce5eaf2e17790ce44513d52b53b9f600cc16f96
BUG=4002
R=andrew@webrtc.org , jridges@masque.com
Review URL: https://webrtc-codereview.appspot.com/36689004
Patch from Zhongwei Yao <zhongwei.yao@arm.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8070 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 02:56:06 +00:00
