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e102e8147b
|
Enable the iSACfix AudioDecoder test (and make it work again)
As far as I can tell, the test should have been enabled again once
https://code.google.com/p/webrtc/issues/detail?id=1353 was fixed, but
it wasn't, and has rotted a bit as a result. I'm not sure why the
number of encoded bytes have changed, but the output seems to be
correct (EncodeDecodeTest encodes, decodes, and compares the result
with the original).
The DecodePlc change is necessary because r7912 added support for that
to the iSACfix AudioDecoder.
BUG=1353, 3926
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7927 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-12-17 07:30:23 +00:00 |
|
|
88bdec8c3a
|
AudioEncoder subclass for iSACfix
This patch refactors AudioEncoderDecoderIsac so that it can share
almost all code with the very similar AudioEncoderDecoderIsacFix.
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7912 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-12-16 12:49:37 +00:00 |
|
|
3b79daff14
|
Moving encoded_bytes into EncodedInfo
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7883 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-12-12 13:31:24 +00:00 |
|
|
817e50dd7d
|
Make an AudioEncoder subclass for PCM16B
The implementation depends on AudioEncoderPcm to reduce code
duplication.
BUG=3926
R=kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7872 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-12-11 10:47:19 +00:00 |
|
|
b3ad8cf6ca
|
Make an AudioEncoder subclass for iSAC
BUG=3926
Previously committed: https://code.google.com/p/webrtc/source/detail?r=7675
and reverted: https://code.google.com/p/webrtc/source/detail?r=7676
R=henrik.lundin@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7871 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-12-11 10:08:19 +00:00 |
|
|
d8ca723de7
|
Remove CELT support from audio_coding.
R=henrik.lundin@webrtc.org, juberti@webrtc.org
TBR=kjellander@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/33579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7864 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-12-10 11:49:13 +00:00 |
|
|
e04a93bcf5
|
Move the AudioDecoder interface out of NetEq
It belongs with the codecs, next to the AudioEncoder interface.
R=andrew@webrtc.org, henrik.lundin@webrtc.org, kjellander@webrtc.org
Previously committed here: https://code.google.com/p/webrtc/source/detail?r=7798
and reverted here: https://code.google.com/p/webrtc/source/detail?r=7799
Review URL: https://webrtc-codereview.appspot.com/27309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7839 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-12-09 10:12:53 +00:00 |
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|
130fef89dd
|
Bugfix in AudioDecoderTest
When the encoded frame size (L ms) was larger than 10 ms, the test would
repeat the first 10 ms L/10 times for each encoded frame. This is now
fixed.
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7833 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-12-08 21:07:59 +00:00 |
|
|
cb858ba397
|
Make an AudioEncoder subclass for iLBC
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@google.com
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32649005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7828 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-12-08 17:11:44 +00:00 |
|
|
3800e13a3a
|
Revert r7798 ("Move the AudioDecoder interface out of NetEq")
Apparently, it caused all sorts of problems I don't have time to
straighten out right now.
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7799 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-12-03 16:28:17 +00:00 |
|
|
00ba1a7dfd
|
Move the AudioDecoder interface out of NetEq
It belongs with the codecs, next to the AudioEncoder interface.
R=henrik.lundin@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7798 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-12-03 14:23:23 +00:00 |
|
|
7f1dfa5b61
|
Adding a payload type to AudioEncoder objects
The type is set in the Config struct and is provided in the EncodedInfo
output struct from each Encode() call. The audio_decoder_unittest is
updated to verify correct propagation of the payload type.
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7780 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-12-02 12:08:39 +00:00 |
|
|
0cd5558f2b
|
AudioEncoder subclass for G722
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7779 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-12-02 11:45:51 +00:00 |
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1db20a4180
|
Adding EncodedInfo struct to AudioEncoder::Encode
This struct will be expanded in future changes.
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7771 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-12-01 14:44:50 +00:00 |
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1431e4dd1c
|
Revert 7675 "Make an AudioEncoder subclass for iSAC"
Above CL did not compile on Android. Followings are links to Android builds
http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20Builder%20%28dbg%29/builds/2648
http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20Clang%20%28dbg%29/builds/2369
http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20ARM64%20%28dbg%29/builds/1320
> Make an AudioEncoder subclass for iSAC
>
> BUG=3926
> R=henrik.lundin@webrtc.org, kjellander@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/25019004
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7676 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-11-11 01:44:13 +00:00 |
|
|
05feff013e
|
Make an AudioEncoder subclass for iSAC
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7675 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-11-10 23:53:08 +00:00 |
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721ef633d0
|
Remove the codec_type_ member from AudioDecoder
It isn't actually required, as evidenced by the comparative ease with
which it can be removed.
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7606 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-11-04 11:51:46 +00:00 |
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decd9306ae
|
AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket
Rename this accessor function to reflect its new, slightly changed
meaning. The reason for the change is that some codecs (iSAC) vary the
number of 10 ms frames from packet to packet, and so can't return a
truly constant value.
BUG=3926
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7556 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-10-29 08:38:50 +00:00 |
|
|
663fdd02fd
|
Make an AudioEncoder subclass for Opus
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7552 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-10-29 07:28:36 +00:00 |
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|
a37f1dd6b8
|
Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile
This CL contains some cleaning up and refactoring of
audio_decoder_test.cc. A new class ResampleInputAudioFile is created
and used in the tests.
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7528 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-10-27 12:58:18 +00:00 |
|
|
def1e97ed2
|
Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests
BUG=3926
R=kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7481 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-10-21 12:48:29 +00:00 |
|
|
7ee24a7906
|
WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
We have to fix both at once, since there's a macro that calls one of
them or the other.
BUG=909
R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7266
Review URL: https://webrtc-codereview.appspot.com/19229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7285 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-09-24 10:31:02 +00:00 |
|
|
a3c4d4dd2c
|
Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..."
This was causing apparently legitimate failures on the following bots:
http://chromegw/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/2599
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28KK%20Nexus5%29%28dbg%29/builds/2023
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28JB%20Nexus7.2%29%28dbg%29/builds/1825
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28KK%20Nexus5%29/builds/2013
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28JB%20Nexus7.2%29/builds/1795
> WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
>
> We have to fix both at once, since there's a macro that calls one of
> them or the other.
>
> BUG=909
> R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/19229004
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7267 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-09-23 01:32:57 +00:00 |
|
|
8c5740b485
|
WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
We have to fix both at once, since there's a macro that calls one of
them or the other.
BUG=909
R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7266 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-09-22 23:04:14 +00:00 |
|
|
3c0aae17f0
|
Change gflags and gmock includes to be full paths.
This will fix PRESUBMIT warnings developers will get due to
r7014 and r7020.
Also some minor style cleanup in:
webrtc/modules/audio_coding/main/test/RTPFile.cc
webrtc/modules/audio_coding/neteq/test/RTPjitter.cc
BUG=
R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7058 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-09-04 09:55:40 +00:00 |
|
|
f563e85ab0
|
This is to re-open an earlier CL
https://webrtc-codereview.appspot.com/16619005/
which is reverted due to an issue in audio conference mixer.
This issue has been solved in
https://webrtc-codereview.appspot.com/20779004/
BUG=webrtc:3155
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18819005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6736 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-07-18 21:11:27 +00:00 |
|
|
1ed1af9b31
|
Remove payload duplication in AudioDecoderTest
This hack was made to come around issue 845. Now that is solved, and
the test code can be cleaned up.
BUG=845
R=kwiberg@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6534 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-06-25 07:59:40 +00:00 |
|
|
d42da54768
|
Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..."
> Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling.
>
> TEST=passed_all_trybots
> R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/16619005
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6462 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-06-17 09:50:12 +00:00 |
|
|
8f8503d947
|
Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling.
TEST=passed_all_trybots
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16619005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6458 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-06-17 08:02:05 +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 |
|