8da96ac0f6
Switch to using AudioEncoderIlbc instead of ACMILBC
...
This change switches from the old codec wrapper ACMILBC to the new
AudioEncoderIlbc wrapped in an ACMGenericCodecWrapper.
BUG=4228
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40699004
Cr-Commit-Position: refs/heads/master@{#8314}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8314 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-10 15:34:38 +00:00
648f5d6dc7
pcm16b: Make input arrays const and use uint8_t[] for byte arrays
...
There were both uint8 and uint16 versions of the pcm16b encode and
decode functions; this patch removes the latter.
BUG=909
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34139004
Cr-Commit-Position: refs/heads/master@{#8309}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8309 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-10 09:19:09 +00:00
c11348b5d7
Fixing a bug in expand_rate calculation for stereo signal.
...
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41849004
Cr-Commit-Position: refs/heads/master@{#8307}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8307 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-10 08:36:07 +00:00
e01bae24a5
Fixing a nit
...
This is a follow-up for https://webrtc-codereview.appspot.com/33209004/
where a post-commit nit was provided.
R=tommi@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35039004
Cr-Commit-Position: refs/heads/master@{#8295}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8295 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 13:21:44 +00:00
1c6239a3b6
G711: Make input arrays const and use uint8_t[] for byte arrays
...
BUG=909
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39809004
Cr-Commit-Position: refs/heads/master@{#8294}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8294 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 12:56:16 +00:00
2b69eab077
Restructure GYP for vp9, opus and direct trace
...
This is needed to make the build more flexible for some use cases.
BUG=4185
R=andresp@webrtc.org , stefan@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34099004
Cr-Commit-Position: refs/heads/master@{#8290}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8290 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 10:01:40 +00:00
751a36590a
Switch to using AudioEncoderPcmU/A instead of ACMPCMU/A
...
This change switches from the old codec wrappers ACMPCMU and ACMPCMA
to the new AudioEncoderPcmU and AudioEncoderPcmA wrapped in an
ACMGenericCodecWrapper. RED and CNG is also switched to using their
AudioEncoder implementations (AudioEncoderCopyRed and AudioEncoderCng,
respectively), when RED and/or CNG is combined with PCM u/A.
This is the first in a series of changes that will switch all codecs
to use the new AudioEncoder interface.
BUG=4228
COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33209004
Cr-Commit-Position: refs/heads/master@{#8268}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8268 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-06 14:03:41 +00:00
74d27884af
Remove defined(__cplusplus) tests in C++ code.
...
This header is a C++ header (it contains keywords such as 'class'
and 'public'). It is not necessary to test defined(__cplusplus).
That test is appropriate in a C header that may be included by C++
code.
R=henrik.lundin@webrtc.org , jan.skoglund@webrtc.org , sprang@webrtc.org
BUG=none
Review URL: https://webrtc-codereview.appspot.com/38899004
Cr-Commit-Position: refs/heads/master@{#8256}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8256 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-05 19:18:21 +00:00
f45c8ca88b
Reland r8248 "Introduce ACMGenericCodecWrapper"
...
This effectively reverts r8249.
This new class inherits from ACMGenericCodec. The purpose is to wrap
AudioEncoder objects into an ACMGenericCodec interface. This is a
temporary construction that will be used during the ACM redesign work.
BUG=4228
COAUTHOR=kwiberg@webrtc.org
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38919004
Cr-Commit-Position: refs/heads/master@{#8255}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8255 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-05 18:30:16 +00:00
3a87630629
Revert r8248 "Introduce ACMGenericCodecWrapper"
...
This reverts r8248 due to some build bot failures.
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40649004
Cr-Commit-Position: refs/heads/master@{#8249}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8249 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-05 09:37:11 +00:00
af8c13f2a1
Introduce ACMGenericCodecWrapper
...
This new class inherits from ACMGenericCodec. The purpose is to wrap
AudioEncoder objects into an ACMGenericCodec interface. This is a
temporary construction that will be used during the ACM redesign work.
BUG=4228
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34939004
Cr-Commit-Position: refs/heads/master@{#8248}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8248 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-05 09:20:18 +00:00
cf7efeba37
Add new AudioEncoderOpusTest
...
This test will replace AcmOpusTest when ACMOpus is removed. The old
AcmOpusTest also contains tests for setting and updating the
"application" setting in Opus. However, in the new AudioEncoderOpus
class, the application is trivially set in the Config struct at
construction, wherefore a test is no longer needed.
BUG=3926
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37929004
Cr-Commit-Position: refs/heads/master@{#8244}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8244 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-04 15:34:40 +00:00
0e81fdf5d2
Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting.
...
BUG=chromium:82439
TEST=none
R=henrik.lundin@webrtc.org , mflodman@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40569004
Cr-Commit-Position: refs/heads/master@{#8229}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8229 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-02 23:54:40 +00:00
c420a86f4c
Change name for local CriticalSectionScoped variable
...
Tools were complaining about (harmless) shadowing of variable names.
This is a follow-up to
https://webrtc-codereview.appspot.com/41659004/#msg8
BUG=3926
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37099004
Cr-Commit-Position: refs/heads/master@{#8225}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8225 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-02 10:36:39 +00:00
a1dfbf1e5c
WebRtcG722_Decode: Input array should be const uint8_t[]
...
BUG=909
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38799004
Cr-Commit-Position: refs/heads/master@{#8224}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8224 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-02 08:58:39 +00:00
026b892e72
Using << on an int8_t or uint8_t will output a character rather than a number.
...
Places that do this need to cast to int to get the desired behavior.
BUG=none
TEST=none
R=henrik.lundin@webrtc.org , pthatcher@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40579004
Cr-Commit-Position: refs/heads/master@{#8223}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8223 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 19:54:19 +00:00
05db352f56
Fix a bug in ACM test channel
...
The test code could read outside the allocated memory. The bug could up
until now not be triggered by the production code, but coming changes
would uncover it.
COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34929004
Cr-Commit-Position: refs/heads/master@{#8216}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8216 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 13:04:16 +00:00
3154a1cf9d
Reland r8210 "Add a new parameter to ACMGenericCodec constructor""
...
This effectively reverts r8211.
The problem with r8210 was that the change in constructor signature was not done for other codec selections that then default one. That is, some code that was hidden under #ifdef did not get updated. This is now fixed.
BUG=4228
COAUTHOR=kwiberg@webrtc.org
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37879004
Cr-Commit-Position: refs/heads/master@{#8215}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8215 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 12:30:22 +00:00
4455f6243a
WebRtcIsacfix_Time2SpecNeon and _Spec2TimeNeon: Fix stack alignment
...
The ABI
(http://infocenter.arm.com/help/topic/com.arm.doc.ihi0042e/IHI0042E_aapcs.pdf )
says to 8-byte-align stack frames. That means we have to push an even
number of registers on function entry if we want to be able to make
subroutine calls without adjusting the stack first.
BUG=4177
R=bjornv@webrtc.org , henrik.lundin@webrtc.org , zhongwei.yao@arm.com
Review URL: https://webrtc-codereview.appspot.com/33149004
Cr-Commit-Position: refs/heads/master@{#8214}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8214 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 11:58:15 +00:00
6752b85ff7
Revert r8210 "Add a new parameter to ACMGenericCodec constructor"
...
The change failed to compile on some bots.
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34949004
Cr-Commit-Position: refs/heads/master@{#8211}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8211 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 06:36:41 +00:00
c3643f2fe3
Add a new parameter to ACMGenericCodec constructor
...
Adding the same parameter to the constructors in all subclasses.
This change is in preparation for changes to come where this will be
needed.
BUG=4228
COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34849004
Cr-Commit-Position: refs/heads/master@{#8210}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8210 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 06:15:18 +00:00
13980253f0
Add new members to AudioEncoderOpus::Config
...
Adding fec_enabled and max_playback_rate_hz.
BUG=3926
COAUTHOR:kwiberg@webrtc.org
R=minyue@webrtc.org , tina.legrand@webrtc.org
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39659004
Cr-Commit-Position: refs/heads/master@{#8207}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8207 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 16:09:08 +00:00
a33f05e8d7
Re-land "Remove <(webrtc_root) from source file entries."
...
Changes differing from https://webrtc-codereview.appspot.com/37859004 :
* I put the include_tests==1 stuff of audio_coding.gypi in its
own audio_coding_tests.gypi file, including the Android and isolate
targets which were incorrectly located in the previous CL
* I moved the bwe utilities in remote_bitrate_estimator.gypi
into include_tests==1 since they depend on test.gyp after I
cleaned up the duplicated inclusion of rtp_file_reader.cc
R=stefan@webrtc.org
TBR=tina.legrand@webrtc.org
TESTED=Passing gyp and compile using:
webrtc/build/gyp_webrtc -Dinclude_tests=1
webrtc/build/gyp_webrtc -Dinclude_tests=0
I also setup a Chromium checkout with my checkout mounted in
third_party/webrtc and ran build/gyp_chromium successfully.
BUG=4185
Review URL: https://webrtc-codereview.appspot.com/33159004
Cr-Commit-Position: refs/heads/master@{#8205}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8205 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 14:30:41 +00:00
bdebccf384
Fix a number of things in AudioEncoderDecoderIsac*
...
- Add max_bit_rate and max_payload_size_bytes to config structs.
- Fix support for 48 kHz sample rate.
- Fix iSAC-RED.
- Add method UpdateDecoderSampleRate().
- Update locking structure with a separate lock for local member
variables used by the encoder methods.
BUG=3926
COAUTHOR:kwiberg@webrtc.org
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41659004
Cr-Commit-Position: refs/heads/master@{#8204}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8204 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 14:11:09 +00:00
4161715e3f
Remove ChangeUniqueID.
...
This fixes a two year old TODO of deleting dead code :)
In cases where the _id or id_ member variable is being used for tracing,
I changed the member to at least be const.
It doesn't look like id's are that useful anymore so maybe the next step is to get rid of them.
BUG=
R=henrika@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37849004
Cr-Commit-Position: refs/heads/master@{#8201}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8201 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 12:14:13 +00:00
1ece0cbbec
Revert "Remove <(webrtc_root) from source file entries."
...
And the follow-up fix in r8198 that was not sufficient.
Reason: breaks Chromium bots runhooks (GYP).
I will have to try some more to make sure I don't
include test code, since include_tests==0 in Chromium.
TBR=andrew@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37039004
Cr-Commit-Position: refs/heads/master@{#8200}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8200 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 12:02:42 +00:00
a87c398a41
Move audio_codec_speed_tests into include_tests==1 condition.
...
I made a mistake in https://webrtc-codereview.appspot.com/37859004
and moved this target out of the include_tests==1 condition.
This moves it back in.
TBR=tina.legrand@webrtc.org
BUG=4185
Review URL: https://webrtc-codereview.appspot.com/33139004
Cr-Commit-Position: refs/heads/master@{#8198}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8198 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 10:39:45 +00:00
2d2a1f9f05
Remove <(webrtc_root) from source file entries.
...
This required to move the AGC tools source files
into webrtc/tools and create a new agc_test_utils target.
Since audio_codec_speed_tests.gypi referenced sources above,
the best approach I could come up with was to add an audio_coding.gypi
file at a higher level and move the targets in there (+ the includes from
modules.gyp which is an improvement IMO).
I also added a PRESUBMIT.py check to prevent new source
entries being added with <(webrtc_root) in the path.
BUG=4185
R=andrew@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37859004
Cr-Commit-Position: refs/heads/master@{#8197}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8197 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 10:24:44 +00:00
7d2b6a9346
Enable Clang warning implicit-fallthrough and annotate the code.
...
BUG=4242
R=henrik.lundin@webrtc.org , stefan@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34899004
Cr-Commit-Position: refs/heads/master@{#8187}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8187 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 18:38:13 +00:00
664ccb7d8d
Reland r8125: Modify some tests to never use DTX disable mode
...
DTX disable mode will be removed as a part of the ACM redesign work.
This CL effectively reverts r8129, and relands r8125, but now using
assert instead of DCHECK.
COAUTHOR:kwiberg@webrtc.org
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37839004
Cr-Commit-Position: refs/heads/master@{#8185}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8185 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 14:49:12 +00:00
4aecd008dd
Add support for 40 and 60 ms frames to AudioEncoderIlbc
...
BUG=3926
COAUTHOR:kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37789004
Cr-Commit-Position: refs/heads/master@{#8182}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8182 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 13:16:44 +00:00
8bb32d600b
Minor updates to AudioEncoderCng
...
Removing sample_rate_hz_ from AudioEncoderCng and from the config
struct. The sample rate will now be read from the underlying speech
codec.
BUG=3926
COAUTHOR:kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40559004
Cr-Commit-Position: refs/heads/master@{#8173}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8173 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 20:54:22 +00:00
478cedc055
Add new methods to AudioEncoder interface
...
The following three methods are added:
rtp_timestamp_rate_hz()
SetTargetBitrate()
SetProjectedPacketLossRate()
Default implementations are provided, and a few overrides are
implemented. AudioEncoderCopyRed and AudioEncoderCng propagate the new
methods to the underlying speech codec.
BUG=3926
COAUTHOR:kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34049004
Cr-Commit-Position: refs/heads/master@{#8171}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8171 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 18:25:40 +00:00
4dba2e98a2
Consolidate anonymous namespace content and file-static methods to all be in the
...
anonymous namespace, in preparation for refactoring a few of the functions a
little.
No code change.
BUG=none
TEST=none
R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8155 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 19:59:32 +00:00
b6fab2b1cd
Introduce rtc::CheckedDivExact
...
Use the new method to replace local ones in AudioEncoder{Opus,Isac}.
COAUTHOR:kwiberg@webrtc.org
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8148 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 11:08:53 +00:00
ff108fe508
Revert 8125 "Modify some tests to never use DTX disable mode"
...
Broke compile on the Chromium FYI bots:
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/3483
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac/builds/16028
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/14293
Error:
In file included from ../../third_party/webrtc/voice_engine/channel.cc:13:
In file included from ../../third_party/webrtc/base/checks.h:22:
In file included from ../../third_party/webrtc/overrides/webrtc/base/logging.h:35:
../../base/logging.h:367:9:error: 'LOG' macro redefined [-Werror,-Wmacro-redefined]
#define LOG(severity) LAZY_STREAM(LOG_STREAM(severity), LOG_IS_ON(severity))
^
../../third_party/webrtc/system_wrappers/interface/logging.h:123:9: note: previous definition is here
#define LOG(sev) \
^
In file included from ../../third_party/webrtc/voice_engine/channel.cc:13:
In file included from ../../third_party/webrtc/base/checks.h:22:
../../third_party/webrtc/overrides/webrtc/base/logging.h:189:9:error: 'LOG_V' macro redefined [-Werror,-Wmacro-redefined]
#define LOG_V(sev) DIAGNOSTIC_LOG(sev, NONE, 0)
^
../../third_party/webrtc/system_wrappers/interface/logging.h:129:9: note: previous definition is here
#define LOG_V(sev) \
^
2 errors generated.
> Modify some tests to never use DTX disable mode
>
> DTX disable mode will be removed as a part of the ACM redesign work.
>
> COAUTHOR:kwiberg@webrtc.org
>
> R=henrika@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/34769004
TBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8129 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 19:02:03 +00:00
043db24767
Modify some tests to never use DTX disable mode
...
DTX disable mode will be removed as a part of the ACM redesign work.
COAUTHOR:kwiberg@webrtc.org
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8125 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 13:30:58 +00:00
fbd37bd737
Make iSAC SWB own its decoder
...
A bug in the ACM codec database caused iSAC-swb to behave differently
from iSAC-wb and -fb. With this fix, all iSAC codecs behave the same
with respect to decoder ownership.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8120 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 08:16:29 +00:00
11af039590
Disable AcmSenderBitExactnessOldApi.Opus_stereo_20ms_voip on ARM64.
...
BUG=4199
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8114 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 14:22:39 +00:00
7dba7860c7
Setting Opus target application.
...
This CL is to allow to set Opus target application at the creation of an encoder.
According to Opus spec, there are three applications:
OPUS_APPLICATION_VOIP
OPUS_APPLICATION_AUDIO
OPUS_APPLICATION_RESTRICTED_LOWDELAY
BUG=
R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8103 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 16:01:50 +00:00
a32d15448d
Disable tests failing on Android ARM64 (Nexus9).
...
BUG=4198,4199,4200
TBR=andrew@webrtc.org
TESTED=Printed using #pragma message to check that the define was properly used.
Review URL: https://webrtc-codereview.appspot.com/33919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8090 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 12:46:01 +00:00
1f67b53c88
Remove dual stream functionality in ACM
...
This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. With this change, there is no longer need for the
ProcessDualStream method, which is removed. Consequently, the method
ProcessSingleStream is renamed to Process.
BUG=3520
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8074 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 09:36:30 +00:00
6c3855258d
Add WebRtcIsacfix_AllpassFilter2FixDec16Neon()'s intrinsics version.
...
This intrinsics version gives bit-exact result as the current C
code. And the performance is 14% better than current assembly
neon version, 3.4 times faster than current C version. The test runs
under Cortex-a53 aarch32 mode, other cpu should give similar performance
result.
Change-Id: Icce5eaf2e17790ce44513d52b53b9f600cc16f96
BUG=4002
R=andrew@webrtc.org , jridges@masque.com
Review URL: https://webrtc-codereview.appspot.com/36689004
Patch from Zhongwei Yao <zhongwei.yao@arm.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8070 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 02:56:06 +00:00
2ebfac5649
Remove COMPILE_ASSERT and use static_assert everywhere
...
COMPILE_ASSERT is no longer needed now that we have C++11's
static_assert.
R=aluebs@webrtc.org , andrew@webrtc.org , hellner@chromium.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8058 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 10:51:54 +00:00
86e1e487e7
Move system_wrappers.gyp files to the proper directory.
...
Build targets should not refer to non-subpaths as was happening before when
source/system_wrappers.gyp refers to ../interface/ files.
R=kjellander@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 09:30:52 +00:00
3df38b442f
Unify the two copies of compile_assert.h
...
This patch basically deletes webrtc/base/compile_assert.h (which is
the more outdated copy) and moves
webrtc/system_wrappers/source/compile_assert.h to take its place.
R=aluebs@webrtc.org , andrew@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8048 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 11:37:48 +00:00
16825b1a82
Use int64_t more consistently for times, in particular for RTT values.
...
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t. Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org , holmer@google.com , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 21:51:21 +00:00
2a26734f04
Partial revert of r7396
...
This change reverts a small part of what was done in r7396. It seems
like that change uncovered another issue with NEON.
BUG=4177,chrome-os-partner:31534
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8043 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 20:52:21 +00:00
c14e3572c6
common_audio: Made input signal const in WebRtcSplFilterMAFastQ12()
...
BUG=3353, 1133
TESTED=locally on Mac and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8037 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 05:50:52 +00:00
f3fd8e7cdf
Add NEON intrinsics version for transform_neon
...
WebRtcIsacfix_Time2SpecNeon and WebRtcIsacfix_Spec2TimeNeon are added.
TransformTest in modules_unittests is passed on ARM32/ARM64 platform.
Initially reviewed here:
https://webrtc-codereview.appspot.com/36449004/
BUG=4002
R=andrew@webrtc.org , jridges@masque.com
Change-Id: I0920ff66a0a0f529707fd7e6619f91e271a47019
Review URL: https://webrtc-codereview.appspot.com/31309004
Patch from Yang Zhang <yang.zhang@arm.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8030 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 18:29:37 +00:00
