All ownership is now handled by the top-level OWNERS file in
modules/audio_coding.
NOTRY=True
Review URL: https://codereview.webrtc.org/1212133005
Cr-Commit-Position: refs/heads/master@{#9512}
This change introduces the sub-class ChangeLogger in AudioCodingModuleImpl. The class writes values to the named UMA histogram, but only if the value has changed since the last time (and always for the first call). This is to avoid the problem with audio codecs being registered but never used. Before this change, these codecs' bitrate was also logged, even though they were never used.
BUG=chromium:488124
R=kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1203803004
Cr-Commit-Position: refs/heads/master@{#9506}
This change includes several improvements:
* VP8 configured with new rate control
* Detection of frame dropping, with qp bump for next frame
* Increased target and TL0 bitrates
* Reworked rate control (TL allocation) in screenshare_layers
A note on performance: PSNR and SSIM is expected to get slightly worse with this cl. Frame drops and delays should however improve.
BUG=4171
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1193513006.
Cr-Commit-Position: refs/heads/master@{#9495}
Revert of original: https://codereview.webrtc.org/1187033005/
Changes in original:
- Added files to gyp and BUILD
- Made minor fixes to get everything to compile
and intelligibility_proc to run
- Added comments
- Auto-reformatting
New Changes:
- Added <numeric> header to intelligibility_enhancer.cc to address buildbot errors
- Switched to use WAV for i/o in intelligibility_proc.cc to address windows errors
- clean up
Note: Patch 1 duplicates Patch 7 of https://codereview.webrtc.org/1182323005/R=andrew@webrtc.org
Review URL: https://codereview.webrtc.org/1190733004.
Cr-Commit-Position: refs/heads/master@{#9486}
This CL logs the target audio bitrate to a UMA histogram called
WebRTC.Audio.TargetBitrateInKbps. It logs the rate when a codec is
created, and when the target is explicitly updated. Note that since
each codec implementation is free to change or ignore the target
value, there is no guarantee that the logged value will actually be
used as the target.
BUG=chromium:488124
Review URL: https://codereview.webrtc.org/1178053002
Cr-Commit-Position: refs/heads/master@{#9484}
This is just https://webrtc-codereview.appspot.com/53629004/
Remove a constructor of VCMJitterBuffer.
Remove unnecessary factory use
Comment Fix
Move frame incoming simulation to the clock
DCHECK typo fix
Coding Style Fix
Rephrased some comments, and removed some virtual for override function.
Coding Style Fix
Coding Style Fix
Add a unittest for VCMReceiver::FrameForDecoding. Mainly test the time control algorithm.
BUG=
TBR=holmer@chromium.org
Review URL: https://codereview.webrtc.org/1173253008.
Cr-Commit-Position: refs/heads/master@{#9470}
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}
Original review at https://codereview.webrtc.org/1180423006
SystemDelayTests was not updated w.r.t. extended_filter mode and some tests were disabled on Android since DA-AEC is automatically set.
All tests have now been updated for both extended_filter mode as well as DA-AEC, hence are now enabled on Android.
Also
* Moves default settings of extended_filter and DA-AEC form Init() to Create() to avoid unintentional loss of state during a reset.
* Fixes a potential bug of starting from scratch in extended_filter mode + DA-AEC.
This reverts commit 01c9b012e9171c813ace9e405c32fc75f4262bf6.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1187943005.
Cr-Commit-Position: refs/heads/master@{#9458}
The code only affects DA-AEC, but since DA-AEC is the default AEC if run on Android tests failed. Reverting to fix that test.
This reverts commit 9002cc426dab7a576f5247f45ba888cd081a39f0.
BUG=
TBR=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1183243003.
Cr-Commit-Position: refs/heads/master@{#9453}
We've seen that if we get a buffer underrun followed by a sudden buffer build up the DA-AEC can't really catch up even though it should be possible to estimate the upcoming difference. We have a feature for this already, but that is only used in the regular AEC. This CL turns that feature on also for DA-AEC.
- Adds a helper function MoveFarReadPtrWithoutSystemDelayUpdate()
- Only apply conservative correction for positive delays, where we can put the AEC into a non-causal state
- Stuff the farend buffer if we don't have enough data to process w.r.t. to current nearend buffer.
- Always run delay estimation based on reported delays to catch buffer starvation.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1180423006.
Cr-Commit-Position: refs/heads/master@{#9452}
This is a follow-up to r9401, where the configuration DelayCorrection
was replaced by ExtendedFilter.
This change also removes the media constraint
kExperimentalEchoCancellation which was replaced by
kExtendedFilterEchoCancellation in the same CL.
Both settings that are now being removed were kept in the code to avoid
API breakages. In https://codereview.chromium.org/1167343004,
depending code has been updated to avoid breakages.
BUG=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1181413004.
Cr-Commit-Position: refs/heads/master@{#9444}
These tables are constant, so it makes sense for all encoders to share
one copy---but it was initialized in a racy way, and there's no
appealing way to fix that without adding dependencies on locking
functions. So we simply give each codec instance its own copy, which
costs 8 * (240 + 240 + 120 + 120) = 5760 bytes apiece.
As noted in the TODO comment, the size of the tables could be reduced,
and they could be filled in at compile-time, but that would make the
encoder output slightly different, which would mess with our tests.
R=henrik.lundin@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1177993003.
Cr-Commit-Position: refs/heads/master@{#9442}
Before this change, it could happen that a caller would get a pointer
to the encoder_ but not use it before another thread called the
Reconstruct method, changing the pointer. This of course resulted in
bad access crashes. With this change, each use of the pointer acquired
from the encoder() method is protected by the same lock that is
required to update the pointer. Note that this fix is probably too
aggressive, since it also affects the Opus implementation; the crash
has so far only been seen for iSAC.
Also adding a test to trigger the problem. The test did not trigger
the problem deterministically, but out would typically find it in less
than 1000 runs.
BUG=chromium:499468
R=jmarusic@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1176303004.
Cr-Commit-Position: refs/heads/master@{#9436}
Fix AppRTCDemo crash under iOS due to the unaligned access in vld1
instruction in iSACFix codec, which is not allowed in iOS build.
BUG=4717
R=andrew@webrtc.org, jridges@masque.com
TEST=Run the AppRTCDemo
Change-Id: Ie5fbc9b8ae88cd00b243a8e65cab95b00362a9da
Review URL: https://codereview.webrtc.org/1182493006.
Cr-Commit-Position: refs/heads/master@{#9432}
This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which
reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24. Specifically, the
files in webrtc/modules/audio_coding/neteq/ 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/1181073002
Cr-Commit-Position: refs/heads/master@{#9427}
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}
This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which
reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24. Specifically, the
files in webrtc/modules/audio_coding/codecs/ilbc/ 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/1184643002
Cr-Commit-Position: refs/heads/master@{#9423}
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}
The SW_SHOWXXX values are not bitfields, so using "&" is incorrect.
BUG=none
TEST=Verify that Chrome's WebRTC window-capture feature can capture an unoccluded, non-maximized window correctly under Windows 8 or above.
R=wez@chromium.org
Review URL: https://codereview.webrtc.org/1180673009
Cr-Commit-Position: refs/heads/master@{#9421}
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}
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}
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}
