8328e7c44d
Revert "Revert part of r7561, "Refactor audio conversion functions.""
...
This restores the conversion changes to AudioProcessing originally
added in r7561, with minor alterations to ensure it passes all tests.
TBR=kwiberg
Review URL: https://webrtc-codereview.appspot.com/28899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7574 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 04:58:14 +00:00
d0cf68ee37
Add 15 fps support for Android devices with missing 15 fps
...
camera mode.
Some latest Android devices support only 30 fps for front camera,
but HW VP8 encoder performance is not enough for 720p 30 fps
encoding. Add 15 fps support for these devices by allowing
frame drop in Android camera wrapper.
BUG=
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7571 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 18:38:26 +00:00
bcfb4d0403
Revert part of r7561, "Refactor audio conversion functions."
...
Specifically, revert this part:
"Remove hacks in AudioBuffer intended to maintain bit-exactness with
the float path. The conversions etc. are now all natural, and
instead we enforce close but not bit-exact output between the two
paths."
But keep the conversion function rename, since that doesn't seem to be
causing problems.
R=tina.legrand@webrtc.org , bjornv@webrtc.org
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7569 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 11:16:06 +00:00
4fc4addc81
Refactor audio conversion functions.
...
Use a consistent naming scheme that can be understood at the callsite
without having to refer to documentation.
Remove hacks in AudioBuffer intended to maintain bit-exactness with the
float path. The conversions etc. are now all natural, and instead we
enforce close but not bit-exact output between the two paths.
Output of ApmTest.Process:
https://paste.googleplex.com/5931055831842816
R=aluebs@webrtc.org , bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7561 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 03:40:10 +00:00
776e6f289c
Use external VideoDecoders in VideoReceiveStream.
...
Removes direct VideoCodec use from the new API, exposes VideoDecoders
through webrtc/video_decoder.h similar to VideoEncoders.
Also includes some preparation for wiring up external decoders in
WebRtcVideoEngine2 by adding AllocatedDecoders that specify whether they
were allocated internally or externally.
Additionally addresses a data race in VideoReceiver that was exposed with this change.
R=mflodman@webrtc.org , stefan@webrtc.org
TBR=pthatcher@webrtc.org
BUG=2854,1667
Review URL: https://webrtc-codereview.appspot.com/27829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7560 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 15:28:39 +00:00
2dd3134e50
Add stats for duplicate sent and received NACK requests.
...
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7559 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 12:42:30 +00:00
decd9306ae
AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket
...
Rename this accessor function to reflect its new, slightly changed
meaning. The reason for the change is that some codecs (iSAC) vary the
number of 10 ms frames from packet to packet, and so can't return a
truly constant value.
BUG=3926
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7556 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 08:38:50 +00:00
663fdd02fd
Make an AudioEncoder subclass for Opus
...
BUG=3926
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7552 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 07:28:36 +00:00
ffeaeed8c1
Make NSinst_t* const and rename to self in ns_core
...
This is only to make the code more readable and maintainable.
It generates a bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7550 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 22:52:09 +00:00
8b1b23f8f8
Make local functions static and dropWebRtcNs_ in ns_core
...
This is only to make the code more readable and maintainable.
It generates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7548 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 21:06:57 +00:00
28b54671cb
Make all comments whole sentences in ns_core
...
This is done to make the code more readable.
It generates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7547 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 20:56:53 +00:00
a296725d0e
audio_coding/codecs/isac/fix: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>"
...
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7544 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 13:05:43 +00:00
67ca26e087
common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16
...
The macro made a cast to uint16_t before a plain multiplication. At the few places where it was used the variables were already uint16_t.
Affected components:
* isac/fix
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7543 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 13:03:10 +00:00
ff8a98e352
Use neteq_unittest_tools in audio_decoder_unittests
...
With the recent move of RtpFileReader to the rtp_test_utils target
(in r7536), it is now possible to let audio_decoder_unittests depend
on neteq_unittest_tools without breaking the Android build.
BUG=2692
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7542 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 09:47:13 +00:00
820efd5b55
Fix double backslashes in incoming_video_stream.cc
...
Originally uploaded in https://codereview.appspot.com/149160043/ .
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7541 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 08:47:16 +00:00
aca5803b19
Move (test) RtpFileReader to a lightweight target.
...
Moves RtpFileReader to rtp_packet_parser and renames it to
rtp_test_utils which is allowed to rely on rtp_rtcp.
R=andrew@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/24979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7536 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 18:01:03 +00:00
a37f1dd6b8
Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile
...
This CL contains some cleaning up and refactoring of
audio_decoder_test.cc. A new class ResampleInputAudioFile is created
and used in the tests.
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7528 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 12:58:18 +00:00
0552356fda
isacfix: Refactor big-endian reading and writing
...
Make subroutines for encoding and decoding arrays of 16-bit big-endian
integers, and in the process fix a bug: When decoding an odd number of
bytes from be16, the least significant byte of the last int16 in the
array was properly taken to be zero instead of actually being read
(since it's outside the array). However, when encoding an odd number
of bytes, the least significant byte of the last int16 in the array
was written to the output as-is instead of being taken to be zero;
thus, we encoded one byte more than we should. This was probably not
harmful, and the value was dropped at decoding anyway; nevertheless,
writing a constant zero is the safe thing to do, and this patch does
so.
R=aluebs@webrtc.org , bjornv@webrtc.org , henrik.lundin@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7527 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 11:25:37 +00:00
7e19a11a71
Break out WebRtcNs_ComputeDdUpdate function in ns_core
...
This is done in order to make the code more readible and maintainable.
It generates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7514 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 19:54:33 +00:00
f8ea0d5518
Break out WebRtcNs_UpdateNoise function in ns_core
...
This is done in order to make the code more readible and maintainable.
It generates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7513 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 19:49:42 +00:00
799e88ae19
Break out FFT function in ns_core
...
This is done in order to make the code more readible and maintainable.
This introduces an error of only +1 and -1.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7512 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 19:36:42 +00:00
8454ad88ed
Break out ComputeSnr function in ns_core
...
This is done in order to make the code more readible and maintainable.
The output is bit-exact.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7511 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 19:34:14 +00:00
580d367b14
Add macros and APIs for webrtc histograms.
...
BUG=crbug/419657
Code that links system_wrappers.gyp:system_wrappers should either:
- provide implementations for the APIs, or
- link with default implementations in system_wrappers.gyp:system_wrappers_default.
R=andresp@webrtc.org , kjellander@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7508 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 12:57:56 +00:00
82462aade0
Adds support for sending first set of packets at increasingly higher bitrates to probe the link and faster ramp up to a high bitrate.
...
Also wires up a finch experiment to control this.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7505 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 11:57:05 +00:00
ad553a2731
Let video_loopback use internal VCM capturers.
...
R=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7503 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 08:24:02 +00:00
3f8f5554a0
Disable TestVp8Impl.BaseUnitTest on MSan.
...
MemorySanitizer uses generic (non-optimized) libvpx which is not bit
exact. This may be a bug in upstream libvpx, but we're forced to disable
it now as it blocks launching the MSan bot.
R=stefan@webrtc.org
TBR=marpan@webrtc.org
BUG=3904
Review URL: https://webrtc-codereview.appspot.com/24089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7491 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 10:30:30 +00:00
76960d5f74
For FIR packet, payload length is zero, so SendToNetwork function is failing.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7490 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 09:47:14 +00:00
67cf1d742b
Break out WebRtcNs_Windowing function in ns_core
...
This is done in order to make the code more readible and maintainable.
This introduces only +1 and -1 errors.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7488 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 22:35:40 +00:00
0e7099244c
Break out WebRtcNs_Energy function in ns_core
...
This is done in order to make the code more readible and maintainable.
This generates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7487 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 22:14:10 +00:00
7634c09406
Break out WebRtcNs_IFFT function in ns_core
...
This is done in order to make the code more readible and maintainable.
This creates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7486 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 21:27:00 +00:00
333e2556ed
Break out WebRtcNs_UpdateBuffer function in ns_core
...
This is done in order to make the code more readible and maintainable.
It generates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7483 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 20:33:09 +00:00
def1e97ed2
Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests
...
BUG=3926
R=kjellander@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7481 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 12:48:29 +00:00
78ea06dd34
audio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
...
Removed usage of trivial macro.
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7480 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 07:17:24 +00:00
913f7b8d5e
Fix for glitches in ACM when switching desired output sample rate
...
The problem was that if the output sample rate is changed such from one
where no resampling is needed to a rate that requires resampling, the
first output from the resampler will contain an onset period. The
solution provided in this CL is to keep a copy of the last output frame
in ACM, and if the resampler is engaged, it will be primed with this
old frame before resampling the current frame.
BUG=3919
R=bjornv@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7479 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 06:54:23 +00:00
2abebe7baf
audio_coding/codecs/isac/main: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
...
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7474 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-20 08:26:41 +00:00
a5ce7bbe17
audio_coding/neteq: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
...
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7473 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-20 08:24:54 +00:00
b1dac33cac
Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
...
BUG=3932
R=marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/27779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7470 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 18:54:46 +00:00
f7030d4ed7
Query Android device orientation on every camera frame received.
...
Remove orientation listener from Android camera, since device
orientation change events are not well synchronized with actual
device display orientation. Plus these event may not be delivered
at all if device is in stationary position causing initial camera
frames appear rotated.
BUG=
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7467 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 16:25:06 +00:00
264e66f7a5
Add encoded_timestamp to AudioEncoder base class
...
BUG=3926
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7464 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-16 21:16:07 +00:00
9ea6f8a84d
New interface class AudioEncoder
...
This class will be the base for new C++ wrapper classes for all
encoders.
BUG=3926
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7463 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-16 11:26:24 +00:00
f87c0aff7f
audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
...
Also includes a typo in a comment.
Affects
* aecm
* hpf
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7456 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 12:51:23 +00:00
f02ba9be54
audio_processing/agc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
...
Affects AGC only.
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7455 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 11:16:48 +00:00
8dc00d76af
audio_processing/ns: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
...
Affects fixed point version of Noise Suppression.
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7454 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 09:31:40 +00:00
99e561f6a6
Extend AcmSwitchingOutputFrequencyOldApi with more frequencies
...
Also reducing test duration, since the issue is triggered anyway.
The tests that are not failing are now enabled.
BUG=3919
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7453 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 08:50:00 +00:00
eb24b04f16
Add periodic logging of received RTP headers and estimated clock offsets for e2e delay.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7444 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 11:40:13 +00:00
81a78930ee
New ACM test to trigger audio glitch when switching output sample rate
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This CL implements a new unit test. The test is designed to trigger
a problem in ACM where switching the desired output frequency creates
a short discontinuity in the output audio. The problem itself is not
solved in this CL, but the failing test is disabled for now.
BUG=3919
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7443 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 10:49:58 +00:00
a57678a70e
Workarounds for a bug in VS2013.3 linker when PGO is turned on.
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See crbug.com/421607 for more details about this. This CL solve a linker bug when the PGO is turned on, without changing the behaviour or the performances.
BUG=crbug.com/421607
R=kwiberg@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26789005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7441 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 09:40:04 +00:00
b6af4283ca
Adjust speech probability in NS when echo
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The average speech probability for the higher band is multiplied by the quotient of the process and analyze powers, to avoid thinking that suppressed echo is speech. In order to do this both magnitudes, alanyze and process, needed to be stored. This also was used to calculate different previous STSA estimates for analyze and process.
This CL was tested on two long team member recordings (bjornv and kwiberg) and the noisiest (5) recordings from the QA set.
BUG=webrtc:3763
R=andrew@webrtc.org , bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7437 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 20:48:05 +00:00
bc1a4578e0
common_audio: Removed macro WEBRTC_SPL_RSHIFT_W16
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Replaced the trivial right shift macro at remaining 4 places and removed from signal_processing.
Affected components:
* vad
* aecm
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7434 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 14:00:43 +00:00
a3722b643d
iSAC tests: Type buffers as uint8_t[] to avoid casts
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The iSAC interface functions now expect uint8_t arrays, so change some
arrays to be of that type instead of casting at each point of use.
R=bjornv@webrtc.org , henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7433 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 13:29:04 +00:00
