12396aba42
Update PacketSource and RtpFileSource
...
The NextPacket method should now return NULL when the end of the
source was reached. In the RtpFileSource, this means that when
the end of file is reached, NULL is returned. Also, when an RTCP
packet is encountered, the next packet will be read from file
immediately.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6479 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-18 12:20:31 +00:00
d8de0669c9
Revert "Restore ptypes.txt file"
...
This reverts r6460. It turns out the file was no longer needed after
all.
BUG=2996
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6478 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-18 11:09:53 +00:00
d42da54768
Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..."
...
> Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling.
>
> TEST=passed_all_trybots
> R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/16619005
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6462 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 09:50:12 +00:00
2ca2188906
Restore ptypes.txt file
...
The file was lost when the neteq folders where moved and renamed.
BUG=2996
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6460 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 08:51:01 +00:00
8f8503d947
Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling.
...
TEST=passed_all_trybots
R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16619005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6458 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 08:02:05 +00:00
721f970cba
common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16
...
We should avoid macros in general (see style guide) and the shift ones are particular dangerous since they assume that the user apply a non-negative shift.
Related CL: https://webrtc-codereview.appspot.com/16669004
BUG=3348,3353
TESTED=trybots and manually on linux
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6444 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 10:30:14 +00:00
63e46077a3
Add thread annotations to parts of ACMGenericCodec
...
This change adds annotations to all member variables that could be
annotated without acquiring any new locks, or changing the lock
structure in any other way.
BUG=3041
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6441 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 08:02:25 +00:00
a1bfc50a72
Pass GYP DEPTH variable to isolate.
...
Similar change to https://codereview.chromium.org/322403003/
This will make it possible to handle different
directory levels for special builds of WebRTC, without
breaking GYP when the .isolate files are processed and
their contents is verified.
Also update all our .isolate files to use the <(DEPTH)
variable.
BUG=343106
TEST=Successful compile+test on Linux using:
ninja -C out/Release
tools/swarming_client/isolate.py run -s out/Release/tools_unittests.isolated
Also trybots passing all tests.
R=pbos@webrtc.org
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6427 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-13 09:02:15 +00:00
5b111b06fa
Re-land "Create a joint encoder/decoder wrapper for iSAC in ACM"
...
The change was reverted since it was thought to cause a flaky test.
But the test kept flaking after the change was reverted.
This effectively reverts r6394, relanding r6377.
BUG=3496
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6404 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 14:37:21 +00:00
c726b1fc33
Revert r6377 "Create a joint encoder/decoder wrapper for iSAC in ACM"
...
BUG=3469
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6394 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 08:35:53 +00:00
18026abd82
common_audio/signal_processing: Removes macro WEBRTC_SPL_RSHIFT_U16
...
This macro is only used at a few places and implies a cast to uint16_t before right shifting. All places already use uint16_t. Further, the amount of shifts applied in the macro has no sanity check for negativity, makes the macro dangerous to use.
BUG=3348,3353
TESTED=trybots and manually
R=kwiberg@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6393 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 06:53:20 +00:00
782978cfcb
common_audio/signal_processing: Moves WEBRTC_SPL_UMUL_16_16_RSFT16 to iSAC fix
...
This macro is only used by the fixed point version of iSAC. Replacing the (five) locations in arith_routines_logist.c, where it is used, with the actual operation.
BUG=3348,3353
TESTED=trybots and manually
R=kwiberg@webrtc.org , tina.legrand@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6392 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 06:39:03 +00:00
fd59c39caa
Delete last file in neteq4 folder
...
The .isolate file can now be safely removed, since issue 3462 is
resolved.
BUG=2996
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6388 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10 20:26:27 +00:00
919914d71b
MIPS optimizations for ISAC (patch #1 )
...
Implemented functions:
- WebRtcIsacfix_AutocorrMIPS
- WebRtcIsacfix_FilterArLoop
- WebRtcIsacfix_FilterMaLoopMIPS
- WebRtcIsacfix_AllpassFilter2FixDec16MIPS (only MIPS DSP)
- WebRtcIsacfix_PitchFilterCore (only MIPS DSPR2)
Gain achieved: from aprox. 15% (MIPS32) up to aprox. 40% (MIPS DSPR2)
R=andrew@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17559005
Patch from Ljubomir Papuga <lpapuga@mips.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6387 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10 18:13:15 +00:00
7b82c18979
Add kjellander@webrtc.org as OWNER for *.isolate
...
This should make project-wide changes for isolate files
easier and make it more obvious who's a suitable reviewer
for them.
BUG=
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6379 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10 05:42:53 +00:00
620048172c
Create a joint encoder/decoder wrapper for iSAC in ACM
...
This CL extends the ACMISAC wrapper class to inherit from AudioDecoder
as well (the type of object that NetEq uses). The class has it's own
lock protecting the iSAC instance. This way, we can remove the
neteq_decode_lock_ (a.k.a. decoder_lock_) in a later CL.
The old AcmAudioDecoderIsac class is deleted.
R=kwiberg@webrtc.org , tina.legrand@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6377 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09 18:39:00 +00:00
a90abdef62
Add thread annotations to AcmReceiver
...
This change adds thread annotations to AcmReceiver. These are the
annotations that could be added without changing acquiring the locks in
more locations, or changing the lock structure.
BUG=3401
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6376 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09 18:35:11 +00:00
a1a2c0c190
Multi-threaded unit test for Audio Coding Module using iSAC
...
This test extends AudioCodingModuleTest and AudioCodingModuleMtTest
to using iSAC as codec.
R=kwiberg@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6369 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09 09:37:17 +00:00
9c55f0f957
Rename neteq4 folder to neteq
...
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for
a hard-coded reference to change.
This CL effectively reverts r6257 "Rename neteq4 folder to neteq".
BUG=2996
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09 08:10:28 +00:00
9221ab420d
Re-enable AudioCodingModuleMtTest again
...
Increase timeout and decrease test length.
BUG=3426
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15679006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6365 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-08 21:43:45 +00:00
94454b71ad
Fix the chain that propagates the audio frame's rtp and ntp timestamp including:
...
* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.
Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.
Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.
BUG=3111
R=henrik.lundin@webrtc.org , turaj@webrtc.org , xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc
Review URL: https://webrtc-codereview.appspot.com/14559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 20:34:08 +00:00
65d61c3924
Opus send rate overflows if over 65 kbps
...
The member holding the send rate for Opus had too low resolution for rates above ~65 kbps.
I've added a test that checks if the average rate in a Opus test is in the right range. The test fails before my fix, and now passes.
BUG=3267
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 13:42:51 +00:00
2f816bbae7
NetEq: Add thread annotation to const scoped_ptrs
...
Since the objects pointed to are not const, only the pointer to them,
they too must be accessed under lock.
Move the crit_sect to above the variables it is protecting.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12679006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6340 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 10:37:13 +00:00
aafd7a88c5
The correct fix of workaround in r6261.
...
The CL also includes same changes to filterbanks.c in iSAC fix and aecm_core_c.c
BUG=3370,3395,3439
TESTED=trybots
R=fdegans@chromium.org , glaznev@webrtc.org , kwiberg@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6337 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 08:53:51 +00:00
edbe886a0b
common_audio/signal_processing: Removed macro WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND
...
This macro was only used at two places in fixed point iSAC, where it has been replaced with the operation.
BUG=3348,3353
TESTED=trybots
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6336 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 08:42:53 +00:00
e6e139159f
Android: cleanup gtest_target_type conditions.
...
Ever since crrev.com/133053 OS==android implies:
gtest_target_type=shared_library
Similar to Chromium's crrev.com/271222 where base.gyp's conditions are changed
(which the affected conditions in this cl comes from).
R=henrike@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6332 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 20:46:50 +00:00
ddc6bc9347
Revert 6312 "Re-enable AudioCodingModuleMtTest"
...
An example of botbreakage is http://chromegw.corp.google.com/i/client.webrtc/builders/Linux%20Memcheck/builds/1807
> Re-enable AudioCodingModuleMtTest
>
> Increase timeout and decrease test length. Also fixing a bug in the
> test, and make sure the test aborts if fatal failure occurrs.
>
> BUG=3426
> R=kwiberg@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/13579005
TBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6314 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-03 15:25:34 +00:00
8d13cd1956
Re-enable AudioCodingModuleMtTest
...
Increase timeout and decrease test length. Also fixing a bug in the
test, and make sure the test aborts if fatal failure occurrs.
BUG=3426
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13579005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6312 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-03 12:53:21 +00:00
a28c697d93
- Get rid of 'using' from .h
...
- Add parenthesis to make order of evaluation clearer.
BUG=
R=minyue@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6304 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-02 15:22:33 +00:00
fe41a8f68d
Adding thread annotations to parts of Audio Coding Module
...
Picking some low-hanging fruit. Add annotations for acm_crit_sect_ that
do not require lock changes. Also adding annotations for callbacks.
BUG=3401
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12579005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6299 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-02 11:45:26 +00:00
af48aaadf4
Disable AudioCodingModuleMtTest due to memcheck and tsan failures.
...
This is a new test; the failures are not due to a change in underlying code.
TBR=henrik.lundin
BUG=3426
Review URL: https://webrtc-codereview.appspot.com/19589005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6288 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 17:11:15 +00:00
288bd15db8
Multi-threaded test for Audio Coding Module
...
This CL adds a basic multi-threaded extention of the ACM unit test.
The test has three threads. One thread adds raw audio to the sender
side and encodes it. The next thread adds encoded RTP packets to the
receiver. The last thread pulls decoded audio out of the receiver.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6286 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 13:00:35 +00:00
a816180f93
Fixing a bug regarding VOE packet loss rate feedback to ACM
...
Phenomenon:
When packet loss rate was fed to a codec that does not implement packet loss adaptive encoding, VoE logs an error.
Reason:
The ACM function SetPacketLossRate(int rate) return -1 unnecessarily too often. It was intended for more severe errors like
1. codec is not ready
2. input rate is out of range
BUG=webrtc:3413
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6283 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 09:28:07 +00:00
0aa3ee661c
Better buffer size estimation in NetEq for redundant packets
...
TEST=passed_all_trybots
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15579005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6260 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 07:48:01 +00:00
1b9df05c85
Revert 6257 "Rename neteq4 folder to neteq"
...
> Rename neteq4 folder to neteq
>
> BUG=2996
> R=turaj@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/12569005
TBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6259 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 07:33:39 +00:00
a90f6d67f7
Rename neteq4 folder to neteq
...
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12569005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6257 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 06:23:34 +00:00
74767401f2
Fix a bug preventing FilePlayer from playing encoded wav files
...
A bug in ACM2 prevented decoding and playout of wav files where the
audio data was encoded (i.e., not just linear PCM 16 bit data).
This CL fixes the issue, and adds a unit test for the FilePlayer.
BUG=3386
R=henrike@webrtc.org , tina.legrand@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21499005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6248 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-26 13:37:45 +00:00
546961a9d3
Avoid reading uninitialized values (outside baundary) in DFT arithmatic decoder of iSAC-fix.
...
Arithmetic encoder does not right the last 2 or 3 bytes of |streamval| when terminating the bit-stream. Perhaps the last bytes makes no difference in decoding the stream. However, the decoder reads full |streamval| (int16_t) going out of boundary and reading uninitialized values. This avoids this problem. by inserting zero-bytes whenever decoder intends to read outside boundary.
BUG=1353,chrome373312,b/13468260
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16499005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6234 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 17:14:29 +00:00
aa5ea1c0f9
1. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED
...
2. Add two new APIs to configure codec internal FEC
3. Add a test and listened to results. This is based modifying EncodeDecodeTest and deriving a new class from it.
New ACM gives good result.
Old ACM does not use NetEq 4, so FEC won't be decoded.
BUG=
R=tina.legrand@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6233 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 15:16:51 +00:00
88fbb2d86b
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
...
Same as https://webrtc-codereview.appspot.com/19519004 . The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux ...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing ...
(tested locally).
BUG=3380
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17619005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 21:18:46 +00:00
2fa7f79094
Revert 6202 "Switch to using base/constructormagic.h and remove ..."
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> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
>
> BUG=N/A
> R=andrew@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/19519004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14579007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 11:07:29 +00:00
48438c2c90
Enabling NetEq bit-exactness test for Win x64
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A new reference file (neteq4_universal_ref_win_64.pcm) was generated and
uploaded.
Also removing the old hack to have different reference files
for different version of Visual Studio. The test is now only supporting
VS 2012 and later (_MSC_VER >= 1700). This makes the windows 32-bit
output identical to the generic reference file
(neteq4_universal_ref.pcm), so the specialized one
(neteq4_universal_ref_win_32.pcm) could have been removed. However,
since the resources sync mechanism does not include removing of old
files, a client could pick up the old reference and fail. Therefore,
this cl also updates neteq4_universal_ref_win_32.pcm to be identical to
neteq4_universal_ref.pcm.
BUG=1458
R=kjellander@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14569005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6204 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 16:07:43 +00:00
125ffd709d
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
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BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 15:20:44 +00:00
cb711f77d2
Add interface to propagate audio capture timestamp to the renderer.
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BUG=3111
R=andrew@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 17:39:11 +00:00
a3b5673879
common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16
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This macro was only used on two lines in iSACfix and I replaced those with the operations the macro performed.
BUG=3348
TESTED=trybots, manual unittests
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6184 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 12:11:20 +00:00
1b21a57902
common_audio/signal_processing: Removed macro WEBRTC_SPL_SUB_SAT_W16
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Macro was only mapping a function used in one place.
BUG=3348
TESTED=trybots, unittests
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6180 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 06:40:31 +00:00
b4e80e095f
Re-enable almost all NetEqDecodingTests for Android
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All but three tests in NetEqDecodingTest could be re-enabled without
any changes. Also making sure that the TestNetworkStatistics test exits
on first diff. (Otherwise, the log output gets flooded with error
messages.)
The tests that are still disabled are:
NetEqDecodingTest.TestBitExactness
NetEqDecodingTest.TestNetworkStatistics
NetEqDecodingTest.DecoderError
BUG=3343
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6168 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-15 07:14:00 +00:00
a36ad6929d
Add webrtc field trials API.
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From now on it is expected that code linking system_wrappers.gyp:system_wrappers
provides an implementation for field_trial API or links with the default one in
system_wrappers.gyp:field_trial_default.
Note: Since there is no use of webrtc::field_trial API inside webrtc this CL on
itself does not forces the clients to actually define it. It however lays the
API and updates the gyp rules to link with so that it is ready to use.
Tested: Introduced a use of field trial in system wrappers and make sure all
bots were building successfully.
BUG=crbug/367114
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6147 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 12:24:04 +00:00
2fa17015d1
Re-enable NetEqExternalDecoderTest for Android
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The test runs without problems now.
BUG=3343
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16519005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6144 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 11:45:22 +00:00
bf93fb3176
Re-enable NetEQ DecoderDatabase test for Android
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The test was failing because iLBC is not enabled on Android. Now, the
test is using PCM16B instead.
BUG=3343
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6143 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 10:42:03 +00:00
