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 ..."
...
> 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
...
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.
...
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.
...
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
...
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
...
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
...
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.
...
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
...
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
...
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
5c49c64de5
Remove all use of AudioFrame::energy_ from AudioCodingModule
...
Since r6117, the energy is always calculated in the mixer module,
regardless of the value that ACM sets for energy_.
This part of the the aftermath of issue 3255.
BUG=3255
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6140 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 09:06:52 +00:00
c3e8abda7c
Deleting all NetEq3 files
...
NetEq3 is deprecated and replaced by NetEq4
(webrtc/modules/audio_coding/neteq4/).
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14469007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6118 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 10:40:52 +00:00
3a5825909d
Deleting all ACM1 files
...
ACM1 is deprecated and replaced by ACM2
(webrtc/modules/audio_coding/acm2/).
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18429005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6115 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 08:08:56 +00:00
b9863ce6ba
One of the NetEq methods needs to be virtual.
...
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6099 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-10 00:58:49 +00:00
17bf9a2c5e
Modifying neteq.gyp
...
|codecs| list is introduced, which is used in both |neteq_dependencies| and AudiDecoder unittests dependencies. This way, AudioDecoder unittests depend only on |codecs| and not on the entire |neteq_dependencies|, which is unnecessary.
TEST=trybots
BUG=
R=andrew@webrtc.org , henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6094 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-09 18:04:50 +00:00
9bd49becc1
Fix a data race in ACM1 when audio is pulled.
...
BUG=chromium:348511
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6026 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 20:27:45 +00:00
f2aafe4355
Added include of assert.h for files calling assert but missing the include.
...
BUG=N/A
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19409005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6022 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 17:54:17 +00:00
97e67cb476
Fix iOS assembly compile error.
...
In the roll of
https://webrtc-codereview.appspot.com/13369007
the fix in transform_neon.S was incorrectly removed
assuming it was only affecting Android when rolling to
265795. This CL fixes the iOS build when rolled to
266514.
Error looks like:
[893/2157] CC obj/webrtc/modules/audio_coding/codecs/isac/main/source/iSAC.entropy_coding.o
FAILED: /Volumes/data/b/build/goma/gomacc ../../third_party/llvm-build/Release+Asserts/bin/clang -MMD -MF obj/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_neon.transform_neon.o.d -DV8_DEPRECATION_WARNINGS -DBLINK_SCALE_FILTERS_AT_RECORD_TIME -DDISABLE_NACL -DCHROMIUM_BUILD -DUSE_LIBJPEG_TURBO=1 -DENABLE_CONFIGURATION_POLICY -DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY -DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE -DENABLE_EGLIMAGE=1 -DCLD_VERSION=1 -DENABLE_SPELLCHECK=1 -DDISABLE_FTP_SUPPORT=1 -DWEBRTC_RESTRICT_LOGGING -DWEBRTC_MODULE_UTILITY_VIDEO -DWEBRTC_ARCH_ARM -DWEBRTC_ARCH_ARM_V7 -DWEBRTC_ARCH_ARM_NEON -DWEBRTC_POSIX -DWEBRTC_MAC -DWEBRTC_IOS -D__STDC_CONSTANT_MACROS -D__STDC_FORMAT_MACROS -DNDEBUG -DNVALGRIND -DDYNAMIC_ANNOTATIONS_ENABLED=0 -DNS_BLOCK_ASSERTIONS=1 -D_FORTIFY_SOURCE=2 -I../.. -I../.. -I../../webrtc -I../../webrtc/common_audio/resampler/include -I../../webrtc/common_audio/signal_processing/include -I../../webrtc/common_audio/vad/include -isysroot /Applications/Xcode.app/Contents/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS7.1.sdk -Os -gdwarf-2 -fvisibility=hidden -Werror -Wnewline-eof -miphoneos-version-min=6.0 -arch armv7 -Wall -Wendif-labels -Wextra -Wno-unused-parameter -Wno-missing-field-initializers -Wheader-hygiene -Wno-c++11-narrowing -Wno-char-subscripts -Wno-unneeded-internal-declaration -Wno-covered-switch-default -Wstring-conversion -Wno-deprecated-register -Wno-absolute-value -Wno-selector-type-mismatch -std=c99 -fcolor-diagnostics -c ../../webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S -o obj/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_neon.transform_neon.o
../../webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S:45:11: error: immediate expression for mov requires :lower16: or :upper16
mov r6, #(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
^
../../webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S:458:11: error: immediate expression for mov requires :lower16: or :upper16
mov r2, #(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
in
http://build.chromium.org/p/client.webrtc/builders/iOS%20Release/builds/911/steps/compile/logs/stdio
TBR=ajm
TEST=ios trybots passing tryjob based on r6010.
BUG=
Review URL: https://webrtc-codereview.appspot.com/12439005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 10:25:30 +00:00
acf15dc90f
Remove Version method from ACM1
...
BUG=2996
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6009 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 09:25:21 +00:00
70e53fa34d
Remove ACM1 and NetEq3 related targets from modules.gyp
...
Make necessary changes to compile.
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6008 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 08:58:46 +00:00
fdf2053787
Remove AudioCodingModuleFactory
...
These were no longer used anywhere in the code.
BUG=2996
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6007 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 08:22:14 +00:00
0bc9b5a5a7
Add clock to ACM config struct
...
The purpose is to clean up the ACM interface a bit. This is a
follow-up of a comment in http://review.webrtc.org/13379004/ .
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16389005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6006 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 08:09:31 +00:00
e772c71743
Introduce a config struct for AudioCoding module
...
The config struct currently contains the module ID, and the NetEq
config struct, but will be extended in the future. The purpose of this
change is to expose certain NetEq settings to the ACM interface.
BUG=3083
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5993 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 10:16:57 +00:00
12a34247a4
Fix the NetEq build
...
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5990 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 08:36:35 +00:00
116ed1d4f0
Include buffer size limits in NetEq config struct
...
This change includes max_packets_in_buffer and max_delay_ms in the
NetEq config struct. The packet buffer is also no longer limited in
terms of payload sizes (bytes), only number of packets.
The old constants governing the packet buffer limits are deleted.
BUG=3083
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5989 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 08:20:04 +00:00
b08bbf57a6
Add henrik.lundin as owner in AudioCoding module
...
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5988 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 08:15:35 +00:00
