| 6955870806624479723addfae6dcf5d13968796c |
|
13-Jan-2016 |
Peter Kasting <pkasting@google.com> |
Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 3c652b67468d182bd36aee4c31557621be50cc92 |
|
18-Nov-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
modules/audio_coding: Remove some codec include dirs
Also clean up some include_dir entries and update the few
references to them with absolute include paths instead.
Finally fixed a few lint errors and invalid header guards.
None of these are used downstream.
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1438663003 .
Cr-Commit-Position: refs/heads/master@{#10700}
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 288886b2ec9a2dac730f115e9c3079d8439efe60 |
|
06-Nov-2015 |
kwiberg <kwiberg@webrtc.org> |
Pass audio to AudioEncoder::Encode() in an ArrayView
Instead of in separate pointer and size arguments.
Review URL: https://codereview.webrtc.org/1418423010
Cr-Commit-Position: refs/heads/master@{#10535}
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 12cfc9b4dacd6942377df1f29a64bdbec591920e |
|
08-Sep-2015 |
kwiberg <kwiberg@webrtc.org> |
Fold AudioEncoderMutable into AudioEncoder
It makes more sense to combine the two interfaces, since there wasn't
a clear line separating them. The result is a combined interface with
just over a dozen methods, half of which need to be implemented by
every subclass, while the other half have sensible (and trivial)
default implementations and are implemented only by the few subclasses
that need non-default behavior.
Review URL: https://codereview.webrtc.org/1322973004
Cr-Commit-Position: refs/heads/master@{#9894}
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.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/g722/audio_encoder_g722.cc
|
| 3e89dbf45835896c8fd89f235f396d03bc2e6065 |
|
18-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Add AudioEncoder::GetTargetBitrate
The GetTargetBitrate implementation will return the
target bitrate of the codec. This may differ from the
desired target bitrate, as set by SetTargetBitrate, depending on implementation.
Tests are updated to exercise the new functionality.
R=kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1184313002.
Cr-Commit-Position: refs/heads/master@{#9461}
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| f045e4da43e671ae511aa1d9b6ef2968256a745d |
|
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Prepare to convert various types to size_t.
This makes some behaviorally-invariant changes to make certain code that
currently only works correctly with signed types work safely regardless of the
signedness of the types in question. This is preparation for a future change
that will convert a variety of types to size_t.
There are also some formatting changes (e.g. converting "enum hack" usage to real consts) to make it simpler to just change "int" to "size_t" in the future to change the types of those constants.
BUG=none
R=andrew@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=ajm
Review URL: https://codereview.webrtc.org/1174813003
Cr-Commit-Position: refs/heads/master@{#9413}
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| dcccab3ebb623df74fbb1425da2cb9d9a42439fa |
|
07-May-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
New interface: AudioEncoderMutable
With implementations for all codecs. It has no users yet. This new
interface is the same as AudioEncoder (in fact it is a subclass) but
it allows changing some parameters after construction.
COAUTHOR=henrik.lundin@webrtc.org
BUG=4228
R=jmarusic@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51679004
Cr-Commit-Position: refs/heads/master@{#9149}
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 92f9eacd1353f04df61c9b52046fb34572742f6a |
|
23-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
g722 and red encoders: Use rtc::Buffer instead of scoped_ptr<uint8_t[]>
It's a win for red, and a toss-up for g722 since it never resizes its
buffer.
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45219005
Cr-Commit-Position: refs/heads/master@{#9067}
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 9afaee74ab1ef36c8b4ea4c22f4c5aebf2359da2 |
|
19-Mar-2015 |
jmarusic@webrtc.org <jmarusic@webrtc.org> |
Reland 8749: AudioEncoder: return EncodedInfo from Encode() and EncodeInternal()
Old review at:
https://webrtc-codereview.appspot.com/43839004/
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45769004
Cr-Commit-Position: refs/heads/master@{#8788}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8788 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 019955d77015fed0b2dcec0cc62a8bdd63e0481e |
|
18-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Revert 8749 "We changed Encode() and EncodeInternal() return typ..."
The reason is that this cl adds a static initializer so we can't roll webrtc into Chromium.
See audio_encoder.cc and 'sizes' regression here:
http://build.chromium.org/p/chromium/builders/Linux%20x64/builds/186
> We changed Encode() and EncodeInternal() return type from bool to void in this issue:
> https://webrtc-codereview.appspot.com/38279004/
> Now we don't have to pass EncodedInfo as output parameter, but can return it instead. This also adds the benefit of making clear that EncodeInternal() needs to fill in this info.
>
> R=kwiberg@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/43839004
TBR=jmarusic@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49449004
Cr-Commit-Position: refs/heads/master@{#8772}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8772 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 0cb612b43bc1ef42cde8cb3887dc48917d5a58dd |
|
17-Mar-2015 |
jmarusic@webrtc.org <jmarusic@webrtc.org> |
We changed Encode() and EncodeInternal() return type from bool to void in this issue:
https://webrtc-codereview.appspot.com/38279004/
Now we don't have to pass EncodedInfo as output parameter, but can return it instead. This also adds the benefit of making clear that EncodeInternal() needs to fill in this info.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43839004
Cr-Commit-Position: refs/heads/master@{#8749}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8749 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 51ccf376387266225cd8c78e63238b725860f0af |
|
10-Mar-2015 |
jmarusic@webrtc.org <jmarusic@webrtc.org> |
AudioEncoder: add method MaxEncodedBytes
Added method AudioEncoder::MaxEncodedBytes() and provided implementations in derived encoders. This method returns the number of bytes that can be produced by the encoder at each Encode() call.
Unit tests were updated to use the new method.
Buffer allocation was not changed in AudioCodingModuleImpl::Encode(). It will be done after additional investigation.
Other refactoring work that was done, that may not be obvious why:
1. Moved some code into AudioEncoderCng::EncodePassive() to make it more consistent with EncodeActive().
2. Changed the order of NumChannels() and RtpTimestampRateHz() declarations in AudioEncoderG722 and AudioEncoderCopyRed classes. It just bothered me that the order was not the same as in AudioEncoder class and its other derived classes.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40259005
Cr-Commit-Position: refs/heads/master@{#8671}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8671 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| b1f0de30be3397eba3d423b71abc5c50db2a1665 |
|
26-Feb-2015 |
jmarusic@webrtc.org <jmarusic@webrtc.org> |
AudioEncoder: change Encode and EncodeInternal return type to void
After code cleanup done on issues:
https://webrtc-codereview.appspot.com/34259004/
https://webrtc-codereview.appspot.com/43409004/
https://webrtc-codereview.appspot.com/34309004/
https://webrtc-codereview.appspot.com/34309004/
https://webrtc-codereview.appspot.com/36209004/
https://webrtc-codereview.appspot.com/40899004/
https://webrtc-codereview.appspot.com/39279004/
https://webrtc-codereview.appspot.com/42099005/
and the similar work done for AudioEncoderDecoderIsacT, methods AudioEncoder::Encode and AudioEncoder::EncodeInternal will always succeed. Therefore, there is no need for them to return bool value that represents success or failure.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38279004
Cr-Commit-Position: refs/heads/master@{#8518}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8518 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| f3a306b5bcecc2719d86091631fadc11d145a5b3 |
|
23-Feb-2015 |
jmarusic@webrtc.org <jmarusic@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 50604128db3387cae72b847c4b2ec14f87a4e425 |
|
19-Feb-2015 |
jmarusic@webrtc.org <jmarusic@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 05211277798ca4791fbdc508e24d7fd06d5ee6ff |
|
18-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 478cedc055f95bd160b53a4d7b69d8b3dd023ec7 |
|
27-Jan-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Add new methods to AudioEncoder interface
The following three methods are added:
rtp_timestamp_rate_hz()
SetTargetBitrate()
SetProjectedPacketLossRate()
Default implementations are provided, and a few overrides are
implemented. AudioEncoderCopyRed and AudioEncoderCng propagate the new
methods to the underlying speech codec.
BUG=3926
COAUTHOR:kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34049004
Cr-Commit-Position: refs/heads/master@{#8171}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8171 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 3b79daff14127f3adb19b16d94336d44ff49e841 |
|
12-Dec-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 8911bc52f14636bd98ab516f01629624aff72009 |
|
08-Dec-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Add AudioEncoder::Max10MsFramesInAPacket
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7834 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 8dc21dc238020afd93a367f741823f2f3d0bec93 |
|
03-Dec-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Rename internal AudioEncoder::Encode method to EncodeInternal
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7801 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 7f1dfa5b61f526badbccf1e0a250acee033dd3db |
|
02-Dec-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|
| 0cd5558f2b9357914873479e7901de6adc44609c |
|
02-Dec-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
|