84 Commits

Author SHA1 Message Date
Henrik Lundin
441f634731 Re-land r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
(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}
2015-06-09 14:03:23 +00:00
Henrik Lundin
3fbf3f8841 Revert r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
This reverts commit 5f4b7e2873864c61e2ad6d88679dcd5d321bfd16, since it
broke some of the build bots.

BUG=4696
TBR=bjornv@webrtc.org

Review URL: https://codereview.webrtc.org/1166463006

Cr-Commit-Position: refs/heads/master@{#9380}
2015-06-05 09:04:20 +00:00
Henrik Lundin
5f4b7e2873 Rename APM Config DelayCorrection to ExtendedFilter
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}
2015-06-05 07:55:40 +00:00
Henrik Lundin
8e6fd46cc3 Route time-stretching metrics through libjingle
This change connects currentAccelerateRate and currentPreemptiveRate
in webrtc::NetworkStatistics, through corresponding variables in
VoiceReceiverInfo, to googAccelerateRate and googPreemptiveExpandRate.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50179004

Cr-Commit-Position: refs/heads/master@{#9350}
2015-06-02 07:25:03 +00:00
Henrik Lundin
5263b3c1dd Add options for NetEq fast accelerate mode through libjingle
This CL connects RTCConfiguration::audioJitterBufferFastMode in
PeerConnection.java, through libjingle, down to
NetEq::Config::enable_fast_accelerate in native WebRTC.

When enabled, it will allow NetEq to do faster time-compression when
the buffer level is very high.

BUG=4691
R=henrika@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55479004

Cr-Commit-Position: refs/heads/master@{#9344}
2015-06-01 08:29:55 +00:00
Fredrik Solenberg
ccb49e79fd Remove Soundclip handling from libjingle.
BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51009004

Cr-Commit-Position: refs/heads/master@{#9216}
2015-05-19 09:37:39 +00:00
Henrik Lundin
64dad838e6 Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
The original change was reverted due to a breakage in the chrome build.
This change includes a fix for this.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49329004

Cr-Commit-Position: refs/heads/master@{#9169}
2015-05-11 10:44:20 +00:00
Henrik Lundin
1f629232d5 Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit fd32f35aff8fc28ec084bddc274de284e0422a57.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55369004

Cr-Commit-Position: refs/heads/master@{#9165}
2015-05-10 09:06:20 +00:00
Henrik Lundin
fd32f35aff Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit cdb47a4533b7b1e29e803ed6591a68bb1a4f1692.

Contains a tentative fix to the chrome build breakage caused by the
original change.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47139004

Cr-Commit-Position: refs/heads/master@{#9164}
2015-05-10 09:03:00 +00:00
Henrik Lundin
cdb47a4533 Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit 208a2294cde839025318f1b3d57559cb0611a4e7.
Breaks the Chrome build.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53399004

Cr-Commit-Position: refs/heads/master@{#9161}
2015-05-08 12:03:46 +00:00
Henrik Lundin
208a2294cd Adding a new constraint to set NetEq buffer capacity from peerconnection
This change makes it possible to set a custom value for the maximum
capacity of the packet buffer in NetEq (the audio jitter buffer). The
default value is 50 packets, but any value can be set with the new
functionality.

R=jmarusic@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50869004

Cr-Commit-Position: refs/heads/master@{#9159}
2015-05-08 10:58:51 +00:00
Fredrik Solenberg
4b60c73e74 Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE.
BUG=4574,3109
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49269004

Cr-Commit-Position: refs/heads/master@{#9150}
2015-05-07 12:07:46 +00:00
Peter Boström
8ed6a4bba4 Remove unused non-standard capture stats.
Removes 'googCaptureJitterMs' and 'googCaptureQueueDelayMsPerS' from
talk/. The overuse-detection method used is based on encoding time,
so these stats aren't useful enough to warrant having them showing up in
GetStats().

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50469004

Cr-Commit-Position: refs/heads/master@{#8874}
2015-03-27 09:01:11 +00:00
Bjorn Volcker
bf395c1fc0 Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android
If built-in Echo Cancellation is available on a device it is automatically enabled. The reason is that it in most cases performs better than the WebRTC software echo control for mobile. The drawback is that we can not develop, test and rollout the delay agnostic AEC (DA-AEC) on Android as for desktops.

This CL includes
- adding a media constraint to enable/disable DA-AEC.
- automatically turning on echo cancellation if DA-AEC is enabled.
- a fix in the AEC that enables delay estimation when DA-AEC is enabled, but delay metrics is disabled.
- sets the Config struct ReportedDelay, which controls DA-AEC internally in the AEC.

The test code to verify that it works in AppRTCDemo can be found here:
https://webrtc-codereview.appspot.com/50479004/

BUG=4472
TESTED=locally on N7, N6, Android One
R=glaznev@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48699004

Cr-Commit-Position: refs/heads/master@{#8861}
2015-03-25 21:46:10 +00:00
pbos@webrtc.org
058b1f17ac Remove GetReceiveBandwidthEstimatorStats.
Removes unnecessary non-standard stats that we don't really make use of.

BUG=
R=pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47379004

Cr-Commit-Position: refs/heads/master@{#8588}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8588 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 08:55:16 +00:00
pbos@webrtc.org
1ed6224eaf Revert r8430 "Remove dead stats from Video{Sender,Receiver}Info."
This breaks compilation outside this codebase that needs to have it
removed before.

BUG=4322
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42009004

Cr-Commit-Position: refs/heads/master@{#8432}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8432 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 13:57:43 +00:00
pbos@webrtc.org
8ad05b7628 Remove dead stats from Video{Sender,Receiver}Info.
These stats are neither filled nor plumbed further and might as well be
removed (as proven by how easy they were to remove).

BUG=
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39219004

Cr-Commit-Position: refs/heads/master@{#8430}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8430 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 13:00:46 +00:00
minyue@webrtc.org
c0bd7be0df Adding two new stats to VoiceReceiverInfo
There have been requests of two new stats namely

speech_expand_rate and secondary_decoded_rate.

BUG=3867
R=henrik.lundin@webrtc.org, henrika@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40789004

Cr-Commit-Position: refs/heads/master@{#8415}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8415 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-18 15:24:39 +00:00
pkasting@chromium.org
16825b1a82 Use int64_t more consistently for times, in particular for RTT values.
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t.  Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 21:51:21 +00:00
stefan@webrtc.org
742386a136 Enable payload-based padding by default and remove the API.
BUG=1812
R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7964 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-19 15:33:17 +00:00
pthatcher@webrtc.org
40b276ea7b Cleanup little things found when refactoring.
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/33519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7880 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 02:44:30 +00:00
tkchin@webrtc.org
14146e40aa arm64 iOS build.
Allows successful build of arm64 libraries using
GYP_DEFINES="OS=ios target_arch=arm64 target_subarch=arm64".
Note that not all libraries will be NEON optimized (eg common_audio),
however most importantly libvpx will be. WEBRTC_ARCH_ARM needs to be
defined so that libvpx doesn't post-process, which is significantly
detrimental to performance.

BUG=3898
R=kjellander@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7573 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 00:14:39 +00:00
buildbot@webrtc.org
1abc146aa5 (Auto)update libjingle 78738075-> 78738103
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7554 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 08:14:14 +00:00
buildbot@webrtc.org
81ddc78536 (Auto)update libjingle 77701902-> 77709729
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7450 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 22:39:24 +00:00
pbos@webrtc.org
43336b6b9f Remove unused (no-op) VideoOptions.
Removing VideoOptions: adapt_input_to_encoder, adapt_view_switch,
video_one_layer_screencast and video_high_bitrate.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/23079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7448 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 19:12:06 +00:00
buildbot@webrtc.org
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2014-08-22 14:08:15 +00:00
henrike@webrtc.org
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wu@webrtc.org
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