| 4376648df021fd82f25a38694e33678f802d06ea |
|
27-Aug-2015 |
Karl Wiberg <kwiberg@google.com> |
AudioDecoder: Replace Init() with Reset()
The Init() method was previously used to initialize and reset
decoders, and returned an error code. The new Reset() method is used
for reset only; the constructor is now responsible for fully
initializing the AudioDecoder.
Reset() doesn't return an error code; it turned out that none of the
functions it ended up calling could actually fail, so this CL removes
their error return codes as well.
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1319683002 .
Cr-Commit-Position: refs/heads/master@{#9798}
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
|
| dce40cf804019a9898b6ab8d8262466b697c56e0 |
|
24-Aug-2015 |
Peter Kasting <pkasting@google.com> |
Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
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| aba07ef6d92bf1ded7ad1af49b54a8e6652dfcbb |
|
12-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Reland "Upconvert various types to int.", isac portion.
This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which
reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24. Specifically, the
files in webrtc/modules/audio_coding/codecs/isac/ are relanded.
The original commit message is below:
Upconvert various types to int.
Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t.
Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C."
This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t). Other locations will be converted to size_t in a separate change.
BUG=none
TBR=kwiberg
Review URL: https://codereview.webrtc.org/1179093002
Cr-Commit-Position: refs/heads/master@{#9422}
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
|
| cb180976dd0e9672cde4523d87b5f4857478b5e9 |
|
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Revert "Upconvert various types to int."
This reverts commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24.
BUG=499241
TBR=hlundin
Review URL: https://codereview.webrtc.org/1179953003
Cr-Commit-Position: refs/heads/master@{#9418}
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
|
| 83ad33a8aed1fb00e422b6abd33c3e8942821c24 |
|
10-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Upconvert various types to int.
Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t.
Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C."
This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t). Other locations will be converted to size_t in a separate change.
BUG=none
R=andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54629004
Cr-Commit-Position: refs/heads/master@{#9405}
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
|
| 396a5e00012ea505a58447e9378a64227e74346b |
|
13-Oct-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
WebRtcIsac_Decode et al.: Type encoded data as uint8[], not uint16[]
This patch changes WebRtcIsac_Decode, WebRtcIsac_DecodeRcu, and
WebRtcIsacfix_Decode so that they read the encoded data from a uint8
array instead of a uint16 array.
BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7431 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
|
| 3f7f899a15c2685a8e45484f7b2c540771d28d90 |
|
13-Oct-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
WebRtcIsac_UpdateBwEstimate et al.: Type byte streams as uint8, not uint16
This patch changes the signature of WebRtcIsac_UpdateBwEstimate,
WebRtcIsacfix_UpdateBwEstimate, and WebRtcIsacfix_UpdateBwEstimate1 so
that they expect the encoded data to be uint8 arrays, not uint16,
which is more natural. The implementations of the functions are left
unchanged for now.
BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7430 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
|
| 1172988c794d15706b4c951dcbaa57b11221d225 |
|
13-Oct-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Some WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[]
The affected functions are
WebRtcIsacfix_ReadFrameLen
WebRtcIsacfix_GetNewBitStream
WebRtcIsacfix_ReadBwIndex
and
WebRtcIsac_ReadFrameLen
WebRtcIsac_GetNewBitStream
WebRtcIsac_ReadBwIndex
WebRtcIsac_GetRedPayload
BUG=909
R=aluebs@webrtc.org, henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7429 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
|
| 7ee24a79065a655dcc62a27fd22e0cc77fee6d68 |
|
24-Sep-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
|
| a3c4d4dd2cece2cfbbd687eb76da833c37fbde3c |
|
23-Sep-2014 |
andrew@webrtc.org <andrew@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
|
| 8c5740b48507e8fbb2c56c7dd52a1197ebb5d20d |
|
23-Sep-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
|
| 0946a56023d821e0deca04029bb016ae1f23aa82 |
|
09-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
|
| b0dff12d2bfd2be52c07b0bcce5a36938ea4f491 |
|
03-Dec-2012 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
48 kHz extension to iSAC.
Test:
-manual test with voe_cmd_test.
-manual test with RTPEncode & NetEqRTPPlay.
-manual test with simpleKenny.
-Bit-exact test of iSAC-swb and iSAC-wb with head revision of trunk. The bit-exactness is confirmed on all files generated by running webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACLongtest.txt
Review URL: https://webrtc-codereview.appspot.com/937025
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3226 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
|
| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
|
22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
|