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Author SHA1 Message Date
henrik.lundin@webrtc.org
fa58745445 Delete all codec-specific subclasses of ACMGenericCodec 
They have all been replaced by AudioEncoder subclasses, accessed throgh
ACMGenericCodecWrapper objects. After this change, the only subclass of
ACMGenericCodec is ACMGenericCodecWrapper. (The two will be consolidated
in a future cl.)

This CL also deletes acm_opus_unittest.cc. This test file was already
replaced audio_encoder_opus_unittest.cc	in r8244.

BUG=4228
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40729004

Cr-Commit-Position: refs/heads/master@{#8457}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8457 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-23 09:26:51 +00:00
jmarusic@webrtc.org
2a5cfc2167 Replaced unnecessary check with an explicit CHECK. 
WebRtcIlbcfix_Encode method that is called returns an error code only if a packet with more than 3 frames is passed, which is illegal.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36209004

Cr-Commit-Position: refs/heads/master@{#8456}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8456 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-23 08:53:15 +00:00
kwiberg@webrtc.org
be96bfb179 Re-land "Switch to using AudioEncoderIsac instead of ACMISAC" 
It should work now, after the fix in r8431.

Previously committed in r8342, reverted in r8372, committed in r8378,
and reverted in r8412.

COAUTHOR=henrik.lundin@webrtc.org
BUG=4228
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34279004

Cr-Commit-Position: refs/heads/master@{#8433}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8433 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-19 15:10:49 +00:00
henrik.lundin@webrtc.org
287755246a Fix a problem with reading uninitialized memory in ACM 
When an "empty frame" was produced by ACMGenericCodecWrapper::Encode,
the timestamp value was not set. This is now fixed, and the first byte
of the bitstream is set to something as well to avoid similar problems.

BUG=chromium:459483
R=jmarusic@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34269004

Cr-Commit-Position: refs/heads/master@{#8431}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8431 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-19 13:56:15 +00:00
jmarusic@webrtc.org
50604128db Method WebRtc_g722_encode that is eventually called always returns non-negative integer (internal counter) 
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34259004

Cr-Commit-Position: refs/heads/master@{#8428}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8428 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-19 12:16:49 +00:00
minyue@webrtc.org
c0bd7be0df Adding two new stats to VoiceReceiverInfo 
There have been requests of two new stats namely

speech_expand_rate and secondary_decoded_rate.

BUG=3867
R=henrik.lundin@webrtc.org, henrika@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40789004

Cr-Commit-Position: refs/heads/master@{#8415}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8415 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 15:24:39 +00:00
jmarusic@webrtc.org
b255865e6e The PCM codecs can never fail, so we don't need to check the return value 
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37329004

Cr-Commit-Position: refs/heads/master@{#8413}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8413 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 15:02:44 +00:00
henrik.lundin@webrtc.org
78619e2714 Revert of r8378 "Switch to using AudioEncoderIsac instead of ACMISAC" 
This is a speculative revert to try to isolate a memory issue.

BUG=chromium:459483,4228
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39189004

Cr-Commit-Position: refs/heads/master@{#8412}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8412 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 14:51:15 +00:00
henrik.lundin@webrtc.org
635838bd9b Re-implementing AcmOpusTest as AcmGenericCodecOpusTest 
The old AcmOpusTest depends on the ACMOpus class, but this class was
obsoleted by AudioEncoderOpus. In this CL, the test code is re-written
to use AudioEncoderOpus and ACMGenericCodecWrapper instead of
ACMOpus. Most of the test functionality is preserved, except for the
packet loss rate tests, which where already transferred to
AudioEncoderOpusTest in r8244.

R=kwiberg@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40029004

Cr-Commit-Position: refs/heads/master@{#8410}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 13:15:45 +00:00
kwiberg@webrtc.org
0521127779 AudioEncoder: Rename virtual accessors to CamelCase 
Although sample_rate_hz(), num_channels(), and rtp_timestamp_rate_hz()
are simple accessors for almost all implementations of AudioEncoder,
they are virtual and not guaranteed to be just simple accessors. Thus,
it makes more sense to use the normal CamelCase naming scheme.

BUG=4235
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34239004

Cr-Commit-Position: refs/heads/master@{#8407}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8407 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 12:01:13 +00:00
minyue@webrtc.org
7d721eea14 Adding speech_expand_rate to NetEQ Network Statistics. 
There have been requests for separating rate of expanded speech samples from noise samples.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37309004

Cr-Commit-Position: refs/heads/master@{#8404}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8404 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 10:02:20 +00:00
jmarusic@webrtc.org
71b35a4ce4 iLBC: Use uint8_t[] for byte arrays 
BUG=909

This is the same as https://review.webrtc.org/41779004/ with the review comments addressed.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40769004

Cr-Commit-Position: refs/heads/master@{#8394}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8394 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 16:02:46 +00:00
minyue@webrtc.org
f9b5c1b3d0 Removing CELT. 
CELT is not supported in WebRTC/Libjingle. There are a few left-over in our code base. They are cleaned up in this CL.

BUG=
R=pbos@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36099004

Cr-Commit-Position: refs/heads/master@{#8385}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 12:37:14 +00:00
minyue@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
kjellander@webrtc.org
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
kjellander@webrtc.org
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
kjellander@webrtc.org
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
henrik.lundin@webrtc.org
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
minyue@webrtc.org
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
bjornv@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
kwiberg@webrtc.org
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
minyue@webrtc.org
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
henrik.lundin@webrtc.org
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
kwiberg@webrtc.org
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
kjellander@webrtc.org
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
henrik.lundin@webrtc.org
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
jan.skoglund@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
pkasting@chromium.org
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
henrik.lundin@webrtc.org
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
kwiberg@webrtc.org
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
pkasting@chromium.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
kjellander@webrtc.org
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