Commit Graph

626 Commits

Author SHA1 Message Date
ccd7c7c45d Remove more unused code in ACM
This CL removes a lot of unused code in AudioCodingModuleImpl and
ACMGenericCodec.

BUG=4228
COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40889004

Cr-Commit-Position: refs/heads/master@{#8470}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8470 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-24 12:02:18 +00:00
13ca5f6db2 AudioEncoderOpus: CHECK that encode call doesn't fail
WebRtcOpus_Encode will only ever fail if fed bad input, and since we don't do that, we can CHECK that it doesn't fail instead of having code that tries to handle failure.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40899004

Cr-Commit-Position: refs/heads/master@{#8469}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8469 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-24 09:57:18 +00:00
d324546ced Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :
* Move constants into the files/functions that use them
* Declare variables in the narrowest scope possible
* Use correct (expected, actual) order for gtest macros
* Remove unused functions
* Untabify
* 80-column limit
* Avoid C-style casts
* Prefer true typed constants to "enum hack" constants
* Print size_t using the right format macro
* Shorten and simplify code
* Other random cleanup bits and style fixes

BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36179004

Cr-Commit-Position: refs/heads/master@{#8467}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 21:29:45 +00:00
722739108a Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530)
Includes GN changes from
https://webrtc-codereview.appspot.com/39249004/

Android changes for JNI were required due to
https://codereview.chromium.org/843103003

Other relevant changes:
* src/buildtools: 5c5e924..93b3d0a
* src/third_party/boringssl/src: d306f16..b180ee9
* src/third_party/icu: 4e3266f..2081ee6
* src/third_party/libvpx: 5cdd302..33bbffe
* src/third_party/usrsctp/usrsctplib: 190c8cb..13718c7
* src/tools/gyp: 4d7c139..3464008
* src/tools/swarming_client: bdad118..1b7bfec
Details: b0c3ed3..2c3ffb2/DEPS

Clang version was not updated in this roll.

R=dpranke@chromium.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40079004

Cr-Commit-Position: refs/heads/master@{#8466}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8466 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 19:09:22 +00:00
829a6f4ac2 Merge ACMGenericCodec and ACMGenericCodecWrapper
ACMGenericCodecWrapper was the only remaining subclass of
ACMGenericCodec, and was the only class that was ever instantiated.
This CL merges the two, essentially keeping the function implementations
from ACMGenericCodecWrapper except where the base class's code was
invoked.

As it turns out, a lot of functions were never used, but in some cases
they were refernced in AudioCodingModuleImpl. In these cases, the
referencing code is commented out and marked FATAL(). This will be
further cleaned up in follow-up CLs.

BUG=4228
COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38209004

Cr-Commit-Position: refs/heads/master@{#8463}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8463 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 16:33:49 +00:00
f3a306b5bc g722: Enhanced documentation. Added CHECK.
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43409004

Cr-Commit-Position: refs/heads/master@{#8462}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8462 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 15:41:49 +00:00
2acec4cc32 Enhanced documentation. Replaced DCHECK with CHECK.
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34309004

Cr-Commit-Position: refs/heads/master@{#8461}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8461 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 15:28:14 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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