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
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
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
abe3f1879c
Checking whether ACM uses codec internal or WebRTC DTX.
...
It was not clear how one could know if ACM is using DTX from WebRTC or codec internal DTX.
This CL makes better use of IsInternalDTXReplacedWithWebRtc() which was designed for G.729 to export such information.
Before
IsInternalDTXReplacedWithWebRtc() gives true only if codec == G729 and G729's internal DTX is replaced with WebRTC DTX.
Now
IsInternalDTXReplacedWithWebRtc() gives true also when codec does not have internal DTX, i.e., must use WebRTC DTX, which is much more logical.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7870 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 08:53:21 +00:00
4591fbd09f
Use size_t more consistently for packet/payload lengths.
...
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
4f6f22f0c6
Reland (rev 7259) "Convert AcmReceiverTest to new AudioCoding interface"
...
Was reverted by mistake in 7260. Actual culprit was 7258.
BUG=3520
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7272 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 11:37:57 +00:00
c570761288
Revert "Convert AcmReceiverTest to new AudioCoding interface" (rev 7258).
...
Breaks android modules_unittests tests by crashing on AcmReceiverBitExactness.8kHzOutput
Was already visible on "git cl try" before submitting on https://webrtc-codereview.appspot.com/23719004/#
BUG=3520
R=kwiberg@webrtc.org , henrik.lundin@webrtc.org
TBR=kwiberg@webrtc.org , henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7260 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 13:18:34 +00:00
cfe073539c
Convert AcmReceiverTest to new AudioCoding interface
...
In order to maintain test coverage for the old API (AudioCodingModule)
during the transition period, the old test was copied to
AcmReceiverTestOldApi.
Modified and extended AudioCoding and the implementation to make the
test compile and run.
Created a converter method from new to old config struct
BUG=3520
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7259 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 12:10:44 +00:00
8f073c5054
Create a new interface for AudioCodingModule
...
This is a first draft of the interface, and is work in progress.
BUG=3520
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7085 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 13:16:23 +00:00
adee8f9242
Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate
...
This is to maintain the consistency with the Opus codec option "maxplaybackrate" defined in http://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03
BUG=
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7038 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 12:28:06 +00:00
6aac93bd9c
Adding SetOpusMaxBandwidth in VoE and ACM
...
This is a step to solve
https://code.google.com/p/webrtc/issues/detail?id=1906
In particular, we add an API in VoE and ACM to call Opus's API of setting maximum bandwidth.
TEST = added a test in voe_cmd_test and listened to the result
BUG=
R=henrika@google.com , henrika@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6869 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-12 08:13:33 +00:00
ceb5a1d724
Annotating the rest of AudioCodingModuleImpl
...
A few extra locks had to be acquired as a result of the annotation.
BUG=3401
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6524 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:52:27 +00:00
9c55f0f957
Rename neteq4 folder to neteq
...
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for
a hard-coded reference to change.
This CL effectively reverts r6257 "Rename neteq4 folder to neteq".
BUG=2996
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09 08:10:28 +00:00
1b9df05c85
Revert 6257 "Rename neteq4 folder to neteq"
...
> Rename neteq4 folder to neteq
>
> BUG=2996
> R=turaj@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/12569005
TBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6259 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 07:33:39 +00:00
a90f6d67f7
Rename neteq4 folder to neteq
...
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12569005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6257 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 06:23:34 +00:00
aa5ea1c0f9
1. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED
...
2. Add two new APIs to configure codec internal FEC
3. Add a test and listened to results. This is based modifying EncodeDecodeTest and deriving a new class from it.
New ACM gives good result.
Old ACM does not use NetEq 4, so FEC won't be decoded.
BUG=
R=tina.legrand@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6233 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 15:16:51 +00:00
70e53fa34d
Remove ACM1 and NetEq3 related targets from modules.gyp
...
Make necessary changes to compile.
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6008 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 08:58:46 +00:00
fdf2053787
Remove AudioCodingModuleFactory
...
These were no longer used anywhere in the code.
BUG=2996
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6007 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 08:22:14 +00:00
0bc9b5a5a7
Add clock to ACM config struct
...
The purpose is to clean up the ACM interface a bit. This is a
follow-up of a comment in http://review.webrtc.org/13379004/ .
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16389005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6006 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 08:09:31 +00:00
e772c71743
Introduce a config struct for AudioCoding module
...
The config struct currently contains the module ID, and the NetEq
config struct, but will be extended in the future. The purpose of this
change is to expose certain NetEq settings to the ACM interface.
BUG=3083
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5993 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 10:16:57 +00:00
24301a67c6
Update talk to 58174641 together with http://review.webrtc.org/4319005/ .
...
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5287 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 19:17:43 +00:00
1e8c93c953
Distinguish instances of ACM1 from ACM2 by a version string. This is fpr testing purposes and will be removed when the experiment is done and ACM1 is fade out.
...
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4069006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5161 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-22 17:04:49 +00:00
eb524d997b
Remove deprecated AudioCodingModule::Destroy.
...
Have Channel hold a pointer rather than reference, and shorten the name.
TESTED=trybots
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2256004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4820 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-23 23:02:24 +00:00
532f3dc548
Compile ACM2 and ACM1.
...
First patch set is the same as patch set 3 of http://review.webrtc.org/2237004/
-Make ACM1 to depend on ACM2.
-Remove APIs to set and get background noise mode. There is no VoE call to these
APIs.
-Remove APIs to set and get receive side VAD mode. There is no VoE call to these
APIs, and NetEq 4, doesn't support them.
-Remove callback for in-band DTMF detection. ACM doesn't support in-band DTMF
detection.
-Use acm_common_defs.h everywhere required.
-Complete ACM factory method.
-Update ACMCodecDatabase of ACM2. CNG full-band need to be define-guarded.
Remove dynamic payload-type assignment.
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4785 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-19 00:12:23 +00:00
1c77dfd521
Revert r4772 "Compile ACM1 and ACM2."
...
Breaks Android build.
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2244004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4777 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-18 12:34:05 +00:00
367baa6eb3
Compile ACM1 and ACM2.
...
-Make ACM1 to depend on ACM2.
-Remove APIs to set and get background noise mode. There is no VoE call to these APIs.
-Remove APIs to set and get receive side VAD mode. There is no VoE call to these APIs, and NetEq 4, doesn't support them.
-Remove callback for in-band DTMF detection. ACM doesn't support in-band DTMF detection.
-Use acm_common_defs.h everywhere required.
-Complete ACM factory method.
-Update ACMCodecDatabase of ACM2. CNG full-band need to be define-guarded. Remove dynamic payload-type assignment.
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2237004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4772 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-18 00:36:11 +00:00
e509f943ed
This issue is related to
...
https://chromereviews.googleplex.com/9908014/
I was thinking about shipping ACM2 from the signal repository. There seems to be too many changes in one CL.
BUG=
R=andrew@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2171004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4733 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 17:03:00 +00:00
89df092807
Make the destructor of AudioCodingModule public.
...
This allows the type to be used with a scoped_ptr. Remove all calls to
the deprecated Destroy() from tests.
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2200006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4731 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 01:27:43 +00:00
401ef361ac
Added configuration of max delay to ACM and NetEq
...
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1964004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4499 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-06 21:01:36 +00:00
a305e9612a
Nack for audio.
...
R=stefan@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1507004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4188 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-06 19:00:09 +00:00
e46c8d3875
API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay.
...
TEST=unit-test, manual, trybots.
R=henrik.lundin@webrtc.org , henrika@webrtc.org , mflodman@webrtc.org , mikhal@webrtc.org , stefan@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1384005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4087 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-22 20:39:43 +00:00
0946a56023
WebRtc_Word32 => int32_t etc. in audio_coding/
...
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1271006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-09 00:28:06 +00:00
7a7a008031
Changing non-const reference arguments to pointers, ACM
...
Part of refactoring of ACM, and recent lint-warnings.
This CL changes non-const references in the ACM API to pointers.
BUG=issue1372
Committed: https://code.google.com/p/webrtc/source/detail?r=3543
Review URL: https://webrtc-codereview.appspot.com/1103012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3555 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-21 10:27:48 +00:00
eb7ebf20ed
Revert 3543
...
> Changing non-const reference arguments to pointers, ACM
>
> Part of refactoring of ACM, and recent lint-warnings.
> This CL changes non-const references in the ACM API to pointers.
>
> BUG=issue1372
>
> Review URL: https://webrtc-codereview.appspot.com/1103012
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1116004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3544 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-20 15:57:31 +00:00
374aa49e1a
Changing non-const reference arguments to pointers, ACM
...
Part of refactoring of ACM, and recent lint-warnings.
This CL changes non-const references in the ACM API to pointers.
BUG=issue1372
Review URL: https://webrtc-codereview.appspot.com/1103012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3543 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-20 15:22:23 +00:00
6388c3e2fd
Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM.
...
TEST=ACM unit test is added, also a manual integration test is writen.
Review URL: https://webrtc-codereview.appspot.com/1097009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3506 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-12 21:42:18 +00:00
b8ba4d8109
Add number of inserted samples to NetEq statistics.
...
BUG=
Review URL: https://webrtc-codereview.appspot.com/964030
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3289 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-14 00:06:18 +00:00
c454fab03b
Reformatting ACM. All changes are bit-exact in this CL.
...
TEST=VoE auto-test, audio_coding_module_test;
only 15 ms of teststereo_out_1.pcm is not bit-exact with output file of the head revision
Review URL: https://webrtc-codereview.appspot.com/937035
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3287 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-13 22:46:43 +00:00
b718619f0a
Expose NetEq playout mode off through VoiceEngine.
...
BUG=
Review URL: https://webrtc-codereview.appspot.com/971016
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3272 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-12 21:59:14 +00:00
226db898f7
Dual-stream implementation, not including VoE APIs.
...
Review URL: https://webrtc-codereview.appspot.com/933015
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3230 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03 22:13:31 +00:00
14b43beb7c
Move src/ -> webrtc/
...
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00
