Commit Graph

4245 Commits

Author SHA1 Message Date
2c670dbf13 Added GN target for webrtc_opus_fec_test.
BUG=webrtc:6191
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2268213002
Cr-Commit-Position: refs/heads/master@{#13893}
2016-08-24 13:11:27 +00:00
98468bb456 Revert of GN build rules for four audio processing test executables (patchset #3 id:40001 of https://codereview.webrtc.org/2267403003/ )
Reason for revert:
Breaks most of chromium.webrtc.fyi bots.

Original issue's description:
> GN build rules for four audio processing test executables
>
> click_annotate, intelligibility_proc, nonlinear_beamformer_test, and
> transient_suppression_test.
>
> BUG=webrtc:5949
>
> Committed: https://crrev.com/538b5606a3fb6310aab7a7e747aee16eac885f02
> Cr-Commit-Position: refs/heads/master@{#13890}

TBR=kjellander@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5949

Review-Url: https://codereview.webrtc.org/2274813004
Cr-Commit-Position: refs/heads/master@{#13891}
2016-08-24 12:04:31 +00:00
538b5606a3 GN build rules for four audio processing test executables
click_annotate, intelligibility_proc, nonlinear_beamformer_test, and
transient_suppression_test.

BUG=webrtc:5949

Review-Url: https://codereview.webrtc.org/2267403003
Cr-Commit-Position: refs/heads/master@{#13890}
2016-08-24 11:38:54 +00:00
0561bdf833 Only use payload size within the know send/receive interval for probing calculations.
BUG=webrtc:5859

Review-Url: https://codereview.webrtc.org/2254733005
Cr-Commit-Position: refs/heads/master@{#13889}
2016-08-24 10:44:01 +00:00
619a211562 iLBC: Handle a case of bad input data
We detect an unreasonable state (caused by a bad encoded stream)
before it can lead to problems, and handle it by resetting the
decoder.

NOPRESUBMIT=true
BUG=chromium:617124

Review-Url: https://codereview.webrtc.org/2255203002
Cr-Commit-Position: refs/heads/master@{#13888}
2016-08-24 09:46:48 +00:00
0aa9d1808b Set send side bitrate estimate on successful probing attempt.
BUG=webrtc:5859

Review-Url: https://codereview.webrtc.org/2263973004
Cr-Commit-Position: refs/heads/master@{#13887}
2016-08-24 09:45:42 +00:00
e51b41ae44 Added GN target for isac_test.
BUG=webrtc:6191
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2267423002
Cr-Commit-Position: refs/heads/master@{#13884}
2016-08-24 09:26:04 +00:00
5d167d6829 Removals and renamings in the new audio mixer.
Removed the OutputMixer part of the new mixer and renamed the new
mixer from NewAudioConferenceMixer to AudioMixer.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2249213005
Cr-Commit-Position: refs/heads/master@{#13883}
2016-08-24 09:21:00 +00:00
30be5d7cf4 Updated mixer unittests and fixed a related bug in the new mixer.
Changes to the mixer unittests:

Removed the tests related to the former 'OutputMixer', as it's going
to be removed. Removed incorrect comparison tests with the old mixer
because doing identical mixing decisions with the old mixer proved
unviable.

When the new mixer went from kMaximumAmountOfMixedAudioSources in the
last iteration to kMaximumAmountOfMixedAudioSources+1, it could hit an
RTC_NOTREACHED(); Added fix to mixer and test
AudioMixer.RampedOutSourcesShouldNotBeMarkedMixed that covers that
case.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2253153004
Cr-Commit-Position: refs/heads/master@{#13880}
2016-08-24 08:38:50 +00:00
616df1e95c Added a level indicator to new mixer.
Added a level indicator to the new mixer. The level indicator is
webrtc::voe::AudioLevel. It computes the current audio level, which is
used all the way up to peerconnection.

This is part of the project to rewrite the old conference mixer and
output mixer.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2230823004
Cr-Commit-Position: refs/heads/master@{#13878}
2016-08-24 08:17:20 +00:00
f99a9de069 ProbingEstimator: Erase history based on time threshold
Erases history based on time threshold instead of retaining really old cluster data. Also does a bunch of clean up.

BUG=
R=danilchap@webrtc.org, philipel@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2239143002 .

Cr-Commit-Position: refs/heads/master@{#13870}
2016-08-23 21:23:12 +00:00
a246cfb8b5 Don't include RTP headers in send-side BWE.
When they are included there will be a mismatch between what the BWE says and
what the encoder is allowed to use, causing us to send more than the network
can handle.

BUG=webrtc:6247
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/2269923003 .

Cr-Commit-Position: refs/heads/master@{#13866}
2016-08-23 15:51:57 +00:00
9a11784a7f Migrated GN target :g722_test
Migrated GN target :g722_test from
webrtc/modules/audio_coding/codecs/g722/g722.gypi

NOTRY=True

BUG=webrtc:6191

Review-Url: https://codereview.webrtc.org/2275463002
Cr-Commit-Position: refs/heads/master@{#13865}
2016-08-23 15:36:15 +00:00
16f55a10c4 Migrated GN target :g711_test
Migrated GN target :g711_test from
webrtc/modules/audio_coding/codecs/g711/g711.gypi

NOTRY=True

BUG=webrtc:6191

Review-Url: https://codereview.webrtc.org/2273623002
Cr-Commit-Position: refs/heads/master@{#13864}
2016-08-23 15:08:30 +00:00
2e486462e0 RTC_CHECK and RTC_DCHECK macros for C
So that we don't have to use assert(). Includes one sample call site.

NOTRY=true
BUG=chromium:617124

Review-Url: https://codereview.webrtc.org/2262173002
Cr-Commit-Position: refs/heads/master@{#13862}
2016-08-23 12:54:31 +00:00
d8dd190a08 GN: Fix test_support_unittests and MIPS compile issue.
Move the webrtc/test/test_support/metrics sources into
test_support[_unittests] targets.
This is essentially reverting https://webrtc-codereview.appspot.com/5789004
and moving these sources back to the right target.

Add missing foreman_cif.yuv resource needed for these tests.

For MIPS, a compile error was surfacing for logcat_trace_context.h when
flipping bot to GN, which was fixed.

BUG=webrtc:5949
NOTRY=True

Review-Url: https://codereview.webrtc.org/2267113002
Cr-Commit-Position: refs/heads/master@{#13860}
2016-08-23 11:52:19 +00:00
b3f1c5d2fe Add NetEq::FilteredCurrentDelayMs() and use it in VoiceEngine
The new method returns the current total delay (packet buffer and sync
buffer) in ms, with smoothing applied to even out short-time
fluctuations due to jitter. The packet buffer part of the delay is not
updated during DTX/CNG periods.

This CL also pipes the new metric through ACM and uses it in
VoiceEngine. It replaces the previous method of estimating the buffer
delay (where an inserted packet's RTP timestamp was compared with the
last played timestamp from NetEq). The new method works better under
periods of DTX/CNG.

Review-Url: https://codereview.webrtc.org/2262203002
Cr-Commit-Position: refs/heads/master@{#13855}
2016-08-22 22:40:00 +00:00
6c46eaa544 Add gtest as a dependency for neteq_quality_test_support.
Was removed in Patch Set 5 of https://codereview.webrtc.org/2252413002
but shouldn't have been, since it's actually required.

https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h?l=17

BUG=webrtc:6228
NOTRY=True

Review-Url: https://codereview.webrtc.org/2262173003
Cr-Commit-Position: refs/heads/master@{#13851}
2016-08-22 16:48:11 +00:00
d48717b455 Fix issue where the number of packets reported in tests/simulations sometimes are negative.
BUG=webrtc:6159

Review-Url: https://codereview.webrtc.org/2223033002
Cr-Commit-Position: refs/heads/master@{#13850}
2016-08-22 15:50:36 +00:00
4ec01d9c9d Fix trivial lint errors in FileRecorder and FilePlayer
Mostly, it's about replacing mutable reference arguments with pointer
arguments, and replacing C style casts with C++ style casts.

Review-Url: https://codereview.webrtc.org/2056653002
Cr-Commit-Position: refs/heads/master@{#13849}
2016-08-22 15:43:58 +00:00
853ecb21f7 Style cleanup in UpdateTmmbr:
function names style updated,
unused return type removed.
Comment style fixed, redundant comments removed.
pass-by-pointer parameter changed to pass-by-value because can't be nullptr any more.

NOTRY=true
BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2258523005
Cr-Commit-Position: refs/heads/master@{#13848}
2016-08-22 15:26:22 +00:00
7f82fc988d WebRtcIlbcfix_Smooth: Fix UBSan fuzzer bug (left shift of 1 by 31 overflows)
scale1 == 31 if and only if w10 == 0. So even though 1 << scale1
overflows, we know that the result of the multiplication should be 0.
Handle that case.

BUG=chromium:615818

Review-Url: https://codereview.webrtc.org/2258543002
Cr-Commit-Position: refs/heads/master@{#13847}
2016-08-22 14:43:50 +00:00
642e3bc75b [rtcp] TransportFeedback adjusted to match other rtcp packets:
Derived from rtcp::Rtpfb instead of directly from RtcpPacket
Does not depend on RTCPUtility.
Parse function takes CommonHeader.
TransportFeedback::BlockLength fixed to match size used by Create

BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/1847973003
Cr-Commit-Position: refs/heads/master@{#13846}
2016-08-22 14:37:00 +00:00
49810511c9 [Reland] Cleanup of the AudioDeviceBuffer class.
See https://codereview.webrtc.org/2256833003/

Contains a minor change to ensure that an external client builds.

TBR=magjed
BUG=NONE

Review-Url: https://codereview.webrtc.org/2269553004
Cr-Commit-Position: refs/heads/master@{#13845}
2016-08-22 12:56:17 +00:00
83d79cd4a2 Revert of Add pps id and sps id parsing to the h.264 depacketizer. (patchset #5 id:80001 of https://codereview.webrtc.org/2238253002/ )
Reason for revert:
Breaks some h264 bitstream tests downstream. Reverting for now.

Original issue's description:
> Add pps id and sps id parsing to the h.264 depacketizer.
>
> BUG=webrtc:6208
>
> Committed: https://crrev.com/abcc3de169d8896ad60e920e5677600fb3d40180
> Cr-Commit-Position: refs/heads/master@{#13838}

TBR=sprang@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6208

Review-Url: https://codereview.webrtc.org/2265023002
Cr-Commit-Position: refs/heads/master@{#13844}
2016-08-22 12:34:43 +00:00
e5b4141746 Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData
BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2249223005
Cr-Commit-Position: refs/heads/master@{#13842}
2016-08-22 10:39:31 +00:00
abcc3de169 Add pps id and sps id parsing to the h.264 depacketizer.
BUG=webrtc:6208

Review-Url: https://codereview.webrtc.org/2238253002
Cr-Commit-Position: refs/heads/master@{#13838}
2016-08-22 08:20:43 +00:00
8177452698 iOS H264VideoToolBoxEncoder: Stop scaling native CVPixelBuffers
If the input to H264VideoToolBoxEncoder is a native CVPixelBuffer and
the quality scaler requests scaling, we fall back to a slow path where
the buffer is converted from NV12 to I420 on the CPU and then uploaded
to a native CVPixelBuffer again. It turns out this scaling is not needed
and that the H264VideoToolBoxEncoder can handle the scaling internally.

BUG=b/30939444

Review-Url: https://codereview.webrtc.org/2258103003
Cr-Commit-Position: refs/heads/master@{#13835}
2016-08-20 17:53:32 +00:00
d7a89dbe8b Revert of Cleanup of the AudioDeviceBuffer class (patchset #6 id:100001 of https://codereview.webrtc.org/2256833003/ )
Reason for revert:
Seems to break an external client.

Original issue's description:
> Cleanup of the AudioDeviceBuffer class.
>
> WebRTC works on 10ms buffer sizes in both directions but this class has contained
> support for any size (with some limits) and for changes on the fly. It makes no sense to maintain such code and we have no tests to test it. This CL ensures that only 10ms audio buffers are supported and that nothing can be changed on the fly.
>
> It also updates the style to follow the Google C++ style guide.
>
> Finally, I remove very old (not tested and not maintained) support for file
> handling since the code is never used. It was more or less dead code.
>
> BUG=NONE
> R=magjed@webrtc.org
>
> Committed: https://crrev.com/cf327b45b9f5738950d4fca2b6a7b6030d508cdf
> Cr-Commit-Position: refs/heads/master@{#13833}

TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=NONE

Review-Url: https://codereview.webrtc.org/2260183002
Cr-Commit-Position: refs/heads/master@{#13834}
2016-08-19 15:09:29 +00:00
cf327b45b9 Cleanup of the AudioDeviceBuffer class.
WebRTC works on 10ms buffer sizes in both directions but this class has contained
support for any size (with some limits) and for changes on the fly. It makes no sense to maintain such code and we have no tests to test it. This CL ensures that only 10ms audio buffers are supported and that nothing can be changed on the fly.

It also updates the style to follow the Google C++ style guide.

Finally, I remove very old (not tested and not maintained) support for file
handling since the code is never used. It was more or less dead code.

BUG=NONE
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/2256833003 .

Cr-Commit-Position: refs/heads/master@{#13833}
2016-08-19 14:38:07 +00:00
da161d795c Reformat rtcp_receiver
git cl format --full

BUG=webrtc:5565
NOTRY=true

Review-Url: https://codereview.webrtc.org/2259213002
Cr-Commit-Position: refs/heads/master@{#13832}
2016-08-19 14:29:51 +00:00
861da3c662 Refactor neteq_test_support.
Take 'tools/neteq_quality_test.cc' and 'tools/neteq_quality_test.h' outside of neteq_test_support into their own target, neteq_quality_test_support.

BUG=webrtc:6228
NOTRY=True

Review-Url: https://codereview.webrtc.org/2252413002
Cr-Commit-Position: refs/heads/master@{#13831}
2016-08-19 14:02:31 +00:00
bcba64a0fa GN: Add "//build/config/sanitizers:deps" as a dependency to executable targets.
When the sanitizer bots are switched to GN, this needs to be included as a dependency so that the executables can be compiled.

BUG=webrtc:6215
NOTRY=True

Review-Url: https://codereview.webrtc.org/2250893003
Cr-Commit-Position: refs/heads/master@{#13829}
2016-08-19 09:11:15 +00:00
b5b30908dc Corrected the testvectors for the level controller
bitexactness test. The activation of the test will
be done in another CL.

BUG=

Review-Url: https://codereview.webrtc.org/2257733002
Cr-Commit-Position: refs/heads/master@{#13822}
2016-08-18 16:47:52 +00:00
8df4d0e426 Add playout_delay_oracle_unittest as gn target
BUG=

Review-Url: https://codereview.webrtc.org/2256743002
Cr-Commit-Position: refs/heads/master@{#13821}
2016-08-18 14:53:44 +00:00
3a11933a63 Remove audio_device_test_func.
This code does not work and hasn't been used in a long time. It also
lacks a GN target. There's no reason to save it.

BUG=none

Review-Url: https://codereview.webrtc.org/2255173002
Cr-Commit-Position: refs/heads/master@{#13820}
2016-08-18 14:20:48 +00:00
644fa96886 Added recording of the configuration for the AudioFrame API call
BUG=webrtc:6227

Review-Url: https://codereview.webrtc.org/2252043003
Cr-Commit-Position: refs/heads/master@{#13819}
2016-08-18 13:48:38 +00:00
2b616397de Remove TMMBRSet class
by cleaning RTCPReceiveInfo class
and following cleaning of RTCPReceiver::BoundingSet function.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2254703003
Cr-Commit-Position: refs/heads/master@{#13817}
2016-08-18 13:17:48 +00:00
38d840c35a NetEq: Changing checked_cast to saturated_cast
The cast involves packet_len_samp, which is derived from the timestamps
and sequence numbers of incoming packets. Being values from the outside,
these should be treated as if any value is possible, making a
checked_cast unsuitable. Changing instead to saturated_cast to avoid
overflow with out-of-bounds values.

Review-Url: https://codereview.webrtc.org/2243403007
Cr-Commit-Position: refs/heads/master@{#13815}
2016-08-18 10:49:41 +00:00
e9a6acfbf5 Added missing unittest to the modules/BUILD.gn build file
NOTRY=True

BUG=

Review-Url: https://codereview.webrtc.org/2255093002
Cr-Commit-Position: refs/heads/master@{#13813}
2016-08-18 09:41:51 +00:00
cb2d701946 Add kjellander as BUILD.gn OWNER in webrtc/modules
NOTRY=True

Review-Url: https://codereview.webrtc.org/2258593003
Cr-Commit-Position: refs/heads/master@{#13812}
2016-08-18 09:39:14 +00:00
71fead2146 Reland of StartTimestamp generated randomly in RtpSender constructor (patchset #1 id:1 of https://codereview.webrtc.org/2248413002/ )
Reason for revert:
Reland: downstream code expectation about rtp_sender timestamp adjusted.

Original issue's description:
> Revert of StartTimestamp generated randomly in RtpSender constructor (patchset #4 id:60001 of https://codereview.webrtc.org/2241193002/ )
>
> Reason for revert:
> Breaks downstream code.
>
> Original issue's description:
> > StartTimestamp generated randomly in RtpSender constructor
> > instead of not-randomly at SetSendingState(true)
> > Renamed to timestamp_offset_ to better match meaning of the variable.
> >
> > R=asapersson@webrtc.org, terelius@webrtc.org
> >
> > Committed: https://crrev.com/4466782ae43e1b1125a55ee7e18abd10dd37cede
> > Cr-Commit-Position: refs/heads/master@{#13796}
>
> TBR=asapersson@webrtc.org,terelius@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/86c96948e340cf8b879bddb0c7293f3b5ad4dad4
> Cr-Commit-Position: refs/heads/master@{#13798}

TBR=asapersson@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2257083002
Cr-Commit-Position: refs/heads/master@{#13811}
2016-08-18 09:02:16 +00:00
d4e9f62ea7 Updated AudioDecoderFactory to list AudioCodecSpecs instead of SdpAudioFormats.
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2123923004
Cr-Commit-Position: refs/heads/master@{#13810}
2016-08-18 09:02:15 +00:00
235020dba6 Roll chromium_revision 915e47250f..e3860bd297 (412201:412289)
Change log: 915e47250f..e3860bd297
Full diff: 915e47250f..e3860bd297

No dependencies changed.
No update to Clang.

NOTRY=TRUE
TBR=
BUG=webrtc:6219

Review-Url: https://codereview.webrtc.org/2253973002
Cr-Commit-Position: refs/heads/master@{#13809}
2016-08-18 08:45:53 +00:00
6a35590d14 Add code for dummy file audio to fallback to dummy audio.
BUG=

Review-Url: https://codereview.webrtc.org/2250853002
Cr-Commit-Position: refs/heads/master@{#13804}
2016-08-17 22:19:55 +00:00
d8a72f0ab2 Close input file in FileAudioDevice::StopRecording.
Also added some more logging, to help track down start/stop, start
failure, and the name of the file used.

BUG=

Review-Url: https://codereview.webrtc.org/2253763002
Cr-Commit-Position: refs/heads/master@{#13802}
2016-08-17 22:14:57 +00:00
d22854bf7d FilePlayer: Remove unused default values for arguments
The functions in question were virtual, so we would've wanted to get
rid of the default values even if callers had relied on them.

Review-Url: https://codereview.webrtc.org/2045943004
Cr-Commit-Position: refs/heads/master@{#13800}
2016-08-17 16:27:08 +00:00
4a42900540 Removes redundant log warning in WebRtcAudioManager.
Trivial patch which avoids logs that are of no value.

BUG=NONE

Review-Url: https://codereview.webrtc.org/2250403002
Cr-Commit-Position: refs/heads/master@{#13799}
2016-08-17 15:43:59 +00:00
86c96948e3 Revert of StartTimestamp generated randomly in RtpSender constructor (patchset #4 id:60001 of https://codereview.webrtc.org/2241193002/ )
Reason for revert:
Breaks downstream code.

Original issue's description:
> StartTimestamp generated randomly in RtpSender constructor
> instead of not-randomly at SetSendingState(true)
> Renamed to timestamp_offset_ to better match meaning of the variable.
>
> R=asapersson@webrtc.org, terelius@webrtc.org
>
> Committed: https://crrev.com/4466782ae43e1b1125a55ee7e18abd10dd37cede
> Cr-Commit-Position: refs/heads/master@{#13796}

TBR=asapersson@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2248413002
Cr-Commit-Position: refs/heads/master@{#13798}
2016-08-17 15:12:27 +00:00
5a25d9504a FileRecorder + FilePlayer: Let Create functions return unique_ptr
Because passing ownership in raw pointers makes kittens cry.

This also means we can ditch the Destroy functions and the protected
destructors. (Well, almost. We need to keep the old CreateFilePlayer
and DestroyFilePlayer around for a little while longer because of an
external caller.)

Review-Url: https://codereview.webrtc.org/2049683003
Cr-Commit-Position: refs/heads/master@{#13797}
2016-08-17 14:31:18 +00:00