| 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/opus/audio_encoder_opus.cc
|
| 25702cb1628941427fa55e528f53483f239ae011 |
|
08-Jan-2016 |
pkasting <pkasting@chromium.org> |
Misc. small cleanups.
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests). Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks
All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1534193008
Cr-Commit-Position: refs/heads/master@{#11191}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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/opus/audio_encoder_opus.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/opus/audio_encoder_opus.cc
|
| 74640895fafbb90a6630a6a91b80da0a7cff229c |
|
29-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
audio_coding: rename interface -> include
BUG=webrtc:5095
R=henrik.lundin@webrtc.org
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417173004 .
Cr-Commit-Position: refs/heads/master@{#10444}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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/opus/audio_encoder_opus.cc
|
| 3f5f1c2ad305a665fb2ecd3e31c57d405e19af97 |
|
09-Sep-2015 |
kwiberg <kwiberg@webrtc.org> |
Change return type of AudioEncoder::SetMaxPlaybackRate to void
There's no point in returning a status code, since the max playback rate
is only a suggestion that the encoder is free to disregard.
Review URL: https://codereview.webrtc.org/1332573003
Cr-Commit-Position: refs/heads/master@{#9900}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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/opus/audio_encoder_opus.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/opus/audio_encoder_opus.cc
|
| b297c5a01f88219da26cffe433804963d1b70f0f |
|
23-Jul-2015 |
pkasting <pkasting@chromium.org> |
Miscellaneous changes split from https://codereview.webrtc.org/1230503003 .
These are mostly trivial changes and are separated out just to reduce the
diff on that change to the minimum possible.
Note explanatory comments on patch set 1.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1235643003
Cr-Commit-Position: refs/heads/master@{#9617}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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/opus/audio_encoder_opus.cc
|
| bba780707848ad36066739c60d7b28cd752fb92f |
|
12-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Reland "Upconvert various types to int.", misc. codecs portion.
This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which
reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24. Specifically, the
files in webrtc/modules/audio_coding/codecs/ that are not in ilbc/ or isac/, as
well as webrtc/modules/audio_coding/main/test/opus_test.cc, 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/1179093003
Cr-Commit-Position: refs/heads/master@{#9424}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
|
| 728d9037c016c01295177fa700fc7927f0bb80bb |
|
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Reformat existing code. There should be no functional effects.
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
|
| b7e5054414ff524f9db81dab7917729b8c4c8bcb |
|
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Match existing type usage better.
This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones. For example:
* Change a few type declarations to better match how the majority of code uses those objects.
* Eliminate "< 0" check for unsigned values.
* Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar.
* Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects.
* Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t.
* Similarly, add casts when passing a larger type to a function taking a smaller one.
* Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar.
* Use "false" instead of "0" for setting a bool.
* Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps. For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t.
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=andrew, asapersson, henrika
Review URL: https://codereview.webrtc.org/1168753002
Cr-Commit-Position: refs/heads/master@{#9419}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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/opus/audio_encoder_opus.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/opus/audio_encoder_opus.cc
|
| 092041c1cdadeb82463ee79dfc291d60b41d35ef |
|
11-May-2015 |
Minyue Li <minyue@webrtc.org> |
Setting OPUS_SIGNAL_VOICE when enable DTX.
A better solution than forcing OPUS_APPLICATION_VOIP when enabling DTX has been found, which is to set OPUS_SIGNAL_VOICE.
This reduces the uncertainty of entering DTX over silence period of audio.
This CL contains the setup of OPUS_SIGNAL_VOICE and decoupling opus application mode with DTX.
BUG=4559
R=henrik.lundin@webrtc.org, henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46959004
Cr-Commit-Position: refs/heads/master@{#9168}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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/opus/audio_encoder_opus.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/opus/audio_encoder_opus.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/opus/audio_encoder_opus.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/opus/audio_encoder_opus.cc
|
| e16bfde5124fc75dcd294ae1856da820923e18a4 |
|
12-Mar-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Adding flag to force Opus application and DTX while toggling.
Currently, we only allow Opus DTX in combination with Opus kVoip mode. When one of them is toggled, the other might need to change as well. This CL is to introduce a flag to force a co-config.
BUG=
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40159004
Cr-Commit-Position: refs/heads/master@{#8698}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8698 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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/opus/audio_encoder_opus.cc
|
| c86bbbaa9348b868e94c021426abcc2f5e0144b0 |
|
04-Mar-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Add speech flag to EncodedInfo
The flag indicates if the encoded bitstream is speech or comfort noise.
COAUTHOR=kwiberg@webrtc.org
R=jmarusic@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42629004
Cr-Commit-Position: refs/heads/master@{#8598}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8598 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
|
| 0561716ae262461eaa3fe5291f4626c76822108a |
|
03-Mar-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Adding Opus DTX support in ACM.
This solution does not use the existing VAD/DTX logic of ACM, since Opus DTX is codec feature, while ACM VAD/DTX is mainly for setting the WebRTC VAD/DTX.
During the development of this CL, two old bugs were found and are fixed in this CL too.
They are in
webrtc/modules/audio_coding/test/Channels.cc
and webrtc/modules/audio_coding/main/acm2/acm_opus_unittest.cc
respectively.
BUG=webrtc:1014
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38469004
Cr-Commit-Position: refs/heads/master@{#8573}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8573 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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/opus/audio_encoder_opus.cc
|
| 13ca5f6db2db66e34907141f151dac780ae2411d |
|
24-Feb-2015 |
jmarusic@webrtc.org <jmarusic@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
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| 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/opus/audio_encoder_opus.cc
|
| 6c68c85b46f4394efe64c6914496af3e662e96ae |
|
11-Feb-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
|
| 13980253f0a76511b67a99315c0f7d050a830cb1 |
|
29-Jan-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Add new members to AudioEncoderOpus::Config
Adding fec_enabled and max_playback_rate_hz.
BUG=3926
COAUTHOR:kwiberg@webrtc.org
R=minyue@webrtc.org, tina.legrand@webrtc.org
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39659004
Cr-Commit-Position: refs/heads/master@{#8207}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8207 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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/opus/audio_encoder_opus.cc
|
| b6fab2b1cdcc8fd93ab8ac3dad19ee213a31a89e |
|
26-Jan-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Introduce rtc::CheckedDivExact
Use the new method to replace local ones in AudioEncoder{Opus,Isac}.
COAUTHOR:kwiberg@webrtc.org
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8148 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
|
| 7dba7860c79652593f0a643fc81fe35f8707e0db |
|
20-Jan-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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/opus/audio_encoder_opus.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/opus/audio_encoder_opus.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/opus/audio_encoder_opus.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/opus/audio_encoder_opus.cc
|
| 1db20a418031935595dd66f9f0deb94a07cb8f1f |
|
01-Dec-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
|
| decd9306ae02f157628075311079df30d5e39c1f |
|
29-Oct-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
|
| 663fdd02fde854b9765c500effd6b306681398f7 |
|
29-Oct-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
|