2c1bcf2cb4
Adding decoded_fec_rate to NetEQ Network Statistics.
...
A statistic is introduced to reflect the actual benefits of Opus FEC. It shows what percentage of samples in the rendered audio come from FEC data.
BUG=3867
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34969004
Cr-Commit-Position: refs/heads/master@{#8384}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8384 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-17 10:17:48 +00:00
34509d9f33
Fix an issue with comfort noise in ACMGenericCodecWrapper
...
In some cases it was not possible to set another payload type for CNG
than the default one. This CL fixes this. The problem was also
dependent on whether the comfort noise codec was registered before or
after the speech codec.
A test is implement to expose the bug, registering comfort noise at a
non-default payload type, and both before and after the speech codec.
BUG=4228
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35199004
Cr-Commit-Position: refs/heads/master@{#8380}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8380 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-16 16:02:40 +00:00
fbc347f2ef
Re-land r8342 "Switch to using AudioEncoderIsac instead of ACMISAC""
...
This reverts r8372, with a bug fix: allowing zero rate in
AudioEncoderIsac::Config. Without this fix, setting the rate to zero
triggered a CHECK. Existing callers assumed that zero was a valid
value. Setting the rate to zero will result in the default rate 32000
being set.
BUG=4228,chromium:458638
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org
TBR=tina.legrand@webrtc.org
CC=jmarusic@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39159004
Cr-Commit-Position: refs/heads/master@{#8378}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8378 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-16 14:28:45 +00:00
ce22f13f0e
GN: Changes for vp9, opus and direct trace
...
Corresponding GN changes for
https://webrtc-codereview.appspot.com/34099004/
BUG=4185
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/40669004
Cr-Commit-Position: refs/heads/master@{#8377}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8377 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-16 12:47:45 +00:00
e35fa96cbe
Move isacfix.gypi and isac.gypi
...
Move webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.gypi
and webrtc/modules/audio_coding/codecs/isac/main/source/isac.gypi to
webrtc/modules/audio_coding/codecs/isac to pass recently
added _CheckNoSourcesAboveGyp presubmit rule.
BUG=4002
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37269004
Cr-Commit-Position: refs/heads/master@{#8376}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8376 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-16 12:47:22 +00:00
0f7f161ed6
Add audio_coding module OWNERS file.
...
It should simplify things to have an
OWNERS file at the top level of audio_coding, in addition
to the lower ones.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39149004
Cr-Commit-Position: refs/heads/master@{#8373}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8373 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-16 09:53:47 +00:00
4dc0003bed
Revert r8342 "Switch to using AudioEncoderIsac instead of ACMISAC"
...
BUG=chromium:458638
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33349004
Cr-Commit-Position: refs/heads/master@{#8372}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8372 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-14 09:42:41 +00:00
a8cc3440b1
Allowing RED decoding for Opus.
...
BUG=4247
TEST=reproduced and fixed the bug
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41809004
Cr-Commit-Position: refs/heads/master@{#8364}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8364 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-13 14:02:17 +00:00
ba97ea69f0
audio_coding/codec/ilbc: Removed usage of macro WEBRTC_SPL_MUL_16_16
...
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
Some other minor code cleanup also exists.
BUG=3348, 3353
TESTED=locally on Mac and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34179004
Cr-Commit-Position: refs/heads/master@{#8358}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8358 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-13 09:52:42 +00:00
bb1219eca3
Add a unit test for callbacks with empty frames and fix bug in code
...
This change adds a couple of new tests that verify that callbacks
with frame type kFrameEmpty are sent in between comfort noise packets.
This used to be the case until r8268, and with the fix included in
this CL is once again so.
COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37229004
Cr-Commit-Position: refs/heads/master@{#8353}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8353 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-12 15:53:55 +00:00
76b4ac96cd
Switch to using AudioEncoderIsac instead of ACMISAC
...
This change switches from the old codec wrapper ACMISAC to the new
AudioEncoderIsac wrapped in an ACMGenericCodecWrapper.
This is also the CL where the old codec for producing redundancy (RED)
is inactivated. All RED payloads are now produces through the
AudioEncoderCopyRed or AudioEncoderIsacRed classes.
BUG=4228
TEST=Please, try the iSAC codec extensively.
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33249005
Cr-Commit-Position: refs/heads/master@{#8342}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8342 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-11 21:37:26 +00:00
6c68c85b46
Switch to using AudioEncoderOpus instead of ACMOpus
...
This change switches from the old codec wrapper ACMOpus to the new
AudioEncoderOpus wrapped in an ACMGenericCodecWrapper.
BUG=4228
TEST=Please, try the Opus codec extensively.
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33259004
Cr-Commit-Position: refs/heads/master@{#8341}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8341 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-11 21:34:06 +00:00
fddeaf5daa
Switch to using AudioEncoderG722 instead of ACMG722
...
This change switches from the old codec wrapper ACMG722 to the new
AudioEncodeG722 wrapped in an ACMGenericCodecWrapper.
BUG=4228
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39879004
Cr-Commit-Position: refs/heads/master@{#8330}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8330 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-11 13:28:44 +00:00
c2d0473320
Switch to using AudioEncoderPcm16B instead of ACMPCM16B
...
This change switches from the old codec wrapper ACMPCM16B to the new
AudioEncoderPcm16B wrapped in an ACMGenericCodecWrapper.
BUG=4228
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33249004
Cr-Commit-Position: refs/heads/master@{#8324}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8324 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-11 08:25:44 +00:00
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
