Commit Graph

758 Commits

Author SHA1 Message Date
e9bdfd859c Added ACM_dump protobuf, class for reading/writing and unittest.
This adds a class to read and write ACM_dump protobuf files. In this CL
it is not hooked up to actually store any packets or debug events.
The unittest writes two dummy RTP packets to disk and reads them to see
if they contain the expected data.

BUG=webrtc:4741
R=andrew@webrtc.org, henrik.lundin@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52059005

Cr-Commit-Position: refs/heads/master@{#9460}
2015-06-18 11:04:35 +00:00
1d34fe979c Adds support for webrtc::test::ResourcePath on iOS
BUG=webrtc:4752
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1178843002.

Cr-Commit-Position: refs/heads/master@{#9445}
2015-06-16 08:04:24 +00:00
ac81163011 iSAC: Move global trig tables into the codec instance
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}
2015-06-15 22:02:45 +00:00
d10cd97ad3 Make global constants 'const'
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1188533002.

Cr-Commit-Position: refs/heads/master@{#9438}
2015-06-15 13:07:11 +00:00
a6aa6d96f8 Fix a data race in AudioEncoderMutableImpl and derived classes
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}
2015-06-15 11:46:24 +00:00
01bbe3eb8c Fix AppRTCDemo crash under iOS armv7 devices
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}
2015-06-15 06:57:00 +00:00
36b7cc3264 Reland "Upconvert various types to int.", neteq portion.
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}
2015-06-12 02:57:28 +00:00
bba7807078 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}
2015-06-12 02:02:58 +00:00
a8b335c709 Reland "Upconvert various types to int.", ilbc portion.
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}
2015-06-12 01:51:33 +00:00
aba07ef6d9 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}
2015-06-12 01:19:37 +00:00
728d9037c0 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}
2015-06-11 21:31:48 +00:00
b7e5054414 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}
2015-06-11 19:56:03 +00:00
cb180976dd 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}
2015-06-11 19:42:42 +00:00
f045e4da43 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}
2015-06-11 04:15:51 +00:00
786dbdcc38 Rename targets to use lower case format.
It makes writing a build script for merging libraries
across architectures easier. See talk/build/build_ios_libs.sh.

BUG=
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1171793002.

Cr-Commit-Position: refs/heads/master@{#9412}
2015-06-10 20:45:12 +00:00
a2c79405b4 Ensures that modules_unittests runs on iOS
BUG=4752
R=tkchin@chromium.org

Review URL: https://codereview.webrtc.org/1171033002.

Cr-Commit-Position: refs/heads/master@{#9408}
2015-06-10 11:24:58 +00:00
2a10087d5e Manual cleanups following clang-formatting.
This primarily addresses two things:
* Tab characters still present, mostly in comments
* printfs split across multiple lines in a suboptimal way

Along the way this fixes a few spelling errors and other minor changes.

BUG=none
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52689004

Cr-Commit-Position: refs/heads/master@{#9406}
2015-06-10 00:26:48 +00:00
83ad33a8ae 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}
2015-06-10 00:20:09 +00:00
248b0b0790 Run clang-format --style=Chromium on four files I'm otherwise touching.
The existing style in these files is pretty inconsistent and wildly divergent
from most of WebRTC/Chromium; clang-formatting them not only makes them easier
to read, it makes me see fewer presubmit errors when I try to touch the files to
make other changes.

BUG=none
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52019004

Cr-Commit-Position: refs/heads/master@{#9364}
2015-06-03 19:32:55 +00:00
5abd3e1f98 Revert r9359 "Implement NetEq's CurrentDelay function"
This reverts commit d8a03facf6986a011c8f889c63d87f9216a1e912, since it
broke the Chrome build. Will have to swap to using base/logging.h in
neteq_impl.cc before re-landing this change.

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50219004

Cr-Commit-Position: refs/heads/master@{#9360}
2015-06-03 10:58:52 +00:00
d8a03facf6 Implement NetEq's CurrentDelay function
This was not implemented before. It returns the current total delay (packet buffer and sync buffer) of NetEq. This is the same information that was already available in NetEqNetworkStatistics::current_buffer_size_ms, that can be obtained through NetEq::NetworkStatistics(). But, since the current delay is a key metric of NetEq, it is convenient to have it available in a simpler way.

R=kwiberg@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51149004

Cr-Commit-Position: refs/heads/master@{#9359}
2015-06-03 09:55:53 +00:00
f69f1fbc98 Testing and improving NADA algorithm.
A modified operation mode was added, holding:
--- Stricter conditions for AcceleratedRampUp.
--- Smoother GradualRateUpdate adjustments.
--- New AcceleratedRampDown update mode.
This mode reduces significantly the delay for bitrates around its minimum bound.

Several NADA unittests and a few simulations were added.

Fixed LinkedSet bug.
Fixed IsNewerSequenceNumber/IsNewerTimestamp bug.

BUG=4550
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54399004

Cr-Commit-Position: refs/heads/master@{#9340}
2015-05-30 15:49:32 +00:00
8bb6ea3da9 Reset speech encoder before hooking it up to RED or CNG
Commit 7e0c7d49 ("Add support for external encoders in ACM") changed
things around so that we no longer recreate the speech encoder when
adding CNG or RED to an existing encoder. This isn't correct, since
those two expect to be in sync with the speech encoder they work with.
Solve the problem by resetting the speech encoder before hooking in
RED or CNG.

BUG=crbug/490368
R=jmarusic@webrtc.org
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53589004

Cr-Commit-Position: refs/heads/master@{#9307}
2015-05-28 11:37:27 +00:00
92fbbb21f8 Switch acm_receiver over to using base/logging.h
BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/57439004

Cr-Commit-Position: refs/heads/master@{#9298}
2015-05-27 20:07:46 +00:00
cf808d2366 Add new fast mode for NetEq's Accelerate operation
This change instroduces a mode where the Accelerate operation will be
more aggressive. When enabled, it will allow acceleration at lower
correlation levels, and possibly remove multiple pitch periods at
once.

The feature is enabled through NetEq::Config, and is off by
default. This means that bit-exactness tests are currently not
affected.

A unit test was added for the Accelerate class, with and without fast
mode enabled.

BUG=4691
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50039004

Cr-Commit-Position: refs/heads/master@{#9295}
2015-05-27 12:33:39 +00:00
c065cc797d Clarify boolean flags in neteq_opus_quality_test.
Note that the use of boolean flags in gflags is a bit unnatural. For setting a boolean flag to false: putting "no" in front of its name (see http://gflags.github.io/gflags/)

We make this clearer by defaulting boolean flags to false, and clarifying it in the description.

BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53539004

Cr-Commit-Position: refs/heads/master@{#9293}
2015-05-27 08:01:18 +00:00
c13cacbb39 Remove an unused method in NetEq::Expand
TBR=ivoc@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53549004

Cr-Commit-Position: refs/heads/master@{#9292}
2015-05-27 07:23:53 +00:00
de4703c5d1 Refactor common_audio/vad: Create now returns the handle directly instead of an error code
Changed the WebRtcVad_Create() function to the more conventional format of returning the handle directly instead of an error code to take care of.
In addition NULL was changed to nullptr in the files where it applied.

Affected components:
* AGC
* VAD
* NetEQ

BUG=441, 3347
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51919004

Cr-Commit-Position: refs/heads/master@{#9291}
2015-05-27 05:23:11 +00:00
905495cfaa Introduce NetEq::Config::ToString and use it in NetEq's constructor
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54559004

Cr-Commit-Position: refs/heads/master@{#9279}
2015-05-25 14:58:46 +00:00
d8399e630f Also provide sample rate when registering decoders
This replaces the old practice of looking up the sample rate in a
table, which won't work when we add support for external decoders.

COAUTHOR=henrik.lundin@webrtc.org
BUG=4474
R=jmarusic@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54469004

Cr-Commit-Position: refs/heads/master@{#9276}
2015-05-25 12:40:05 +00:00
323b132f5e Protect ACM decoder buffer in stereo.
In https://code.google.com/p/webrtc/source/detail?r=8730, I did a protection on ACM decoder buffer from being overflow.

However, the I misunderstood the return unit for PacketDuration(), and therefore, stereo decoders are not well protected.

This CL fixed this.

BUG=4361
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47289004

Cr-Commit-Position: refs/heads/master@{#9275}
2015-05-25 11:49:45 +00:00
57e5fd2e60 PRESUBMIT: Improve PyLint check and add GN format check.
Add pylintrc file based on
https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc
bit tightened up quite a bit (the one in depot_tools is far
more relaxed).

Remove a few excluded directories from pylint check and fixed/
suppressed all warnings generated.

Add GN format check + formatted all GN files using 'gn format'.
Cleanup redundant rules in tools/PRESUBMIT.py

TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations.
Ran it again with a modification in webrtc/build/webrtc.gni, formatted
all the GN files and ran it again.

R=henrika@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50069004

Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 10:55:50 +00:00
367c868c99 AudioEncoderCng: Handle case where speech encoder is reset
Previously, AudioEncoderCng required the speech encoder to not change
its mind regarding the number of 10 ms frames in the next packet
between calls to AudioEncoderCng::EncodeInternal()---specifically, it
could handle an upward but not a downward adjustment. With this patch,
it can handle a downward adjustment too, by simply saving the
overshoot data for the next call to EncodeInternal().

It will still not handle the case where the encoder's reported number
of 10 ms frames in the next packet is inconsistent with the behavior
of its Encode() function when called with no intervening changes to
the encoder.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53469005

Cr-Commit-Position: refs/heads/master@{#9261}
2015-05-22 13:13:24 +00:00
f761d10393 Update NetEq Quality Test.
1. move channel number of input file to the base class

2. limit channel number to be 1, since the resampler support only mono at the moment

3. adding a logging function

4. adding more switch to neteq_opus_quality_test

BUG=2692
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47239004

Cr-Commit-Position: refs/heads/master@{#9260}
2015-05-22 09:21:58 +00:00
cb7f8ce2df Clear ARM NEON flag
Merge WEBRTC_ARCH_ARM64_NEON and WEBRTC_ARCH_ARM_NEON into one
WEBRTC_HAS_NEON.
Replace WEBRTC_DETECT_ARM_NEON by WEBRTC_DETECT_NEON.
Replace WEBRTC_ARCH_ARM by WEBRTC_ARCH_ARM64 for arm64 cpu.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org

Change-Id: I870a4d0682b80633b671c9aab733153f6d95a980

Review URL: https://webrtc-codereview.appspot.com/49309004

Cr-Commit-Position: refs/heads/master@{#9228}
2015-05-20 05:20:04 +00:00
7e0c7d49ea Add support for external encoders in ACM
Also introduce tests using external (mock) encoders, both for
CodecOwner and for AudioCodingModule.

Support for external decoders is still missing.

COAUTHOR=henrik.lundin@webrtc.org
BUG=4474
R=jmarusic@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49939004

Cr-Commit-Position: refs/heads/master@{#9206}
2015-05-18 12:52:13 +00:00
bd1bc47395 Restructure decoder registration in ACM
Before this change, a decoder was registered into ACMReceiver through
the CodecOwner; the CodecOwner had to have a pointer back to the
AudioCodingModuleImpl object to make this call. With this change, the
AudioCodingModuleImpl object asks the CodecOwner for a decoder pointer
instead, making the chain of calls more straightforward.

COAUTHOR=henrik.lundin@webrtc.org
BUG=4474
R=jmarusic@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52439004

Cr-Commit-Position: refs/heads/master@{#9204}
2015-05-18 10:18:44 +00:00
8171735b0c Add NetEqIlbcQualityTest
This is virtually the same as NetEq{Isac,Opus}QualityTest but for iLBC.

BUG=2692
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50909004

Cr-Commit-Position: refs/heads/master@{#9178}
2015-05-12 13:04:29 +00:00
e5ff00a1c6 Add NetEqPcmuQualityTest
This is virtually the same as NetEq{Isac,Opus}QualityTest but for PCMu.

BUG=2692
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54379004

Cr-Commit-Position: refs/heads/master@{#9176}
2015-05-12 10:09:53 +00:00
075bb8d125 Fix race in AudioCodingModuleImpl::Add10MsData()
AudioCodingModuleImpl::Add10MsData() calls two private methods that
together do all the work: Add10MsDataInternal() and Encode(). They
each took locks internally in order to protect access to, among other
things, codec_manager_.

This turned out to be inadequate. Add10MsDataInternal() calls
codec_manager_.CurrentEncoder()->SampleRateHz() in order to be able to
resample the audio data to what the current encoder wants. When the
resampled data is fed to the encoder deep inside the Encode() call,
that sample rate must still be correct, but occasionally it wasn't,
which triggered a CHECK. (The specific test that failed was the
voe_auto_test subtest
CodecTest.OpusMaxPlaybackRateCannotBeSetForNonOpus, which changes the
current encoder while encoding is in progress.)

This CL solves the problem by covering all of
AudioCodingModuleImpl::Add10MsData() in a single critical section, so
that the sample rate obtained in Add10MsDataInternal() is guaranteed
to still be valid during the Encode() call.

BUG=4644
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52459004

Cr-Commit-Position: refs/heads/master@{#9174}
2015-05-12 08:09:58 +00:00
cb3e8fe492 Increase the tolerance in NetEq's DelayManagerTest a notch
This change is to make the test pass on Samsung devices.

BUG=4426
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52449004

Cr-Commit-Position: refs/heads/master@{#9172}
2015-05-11 13:15:49 +00:00
64dad838e6 Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
The original change was reverted due to a breakage in the chrome build.
This change includes a fix for this.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49329004

Cr-Commit-Position: refs/heads/master@{#9169}
2015-05-11 10:44:20 +00:00
092041c1cd 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}
2015-05-11 10:19:36 +00:00
1f629232d5 Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit fd32f35aff8fc28ec084bddc274de284e0422a57.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55369004

Cr-Commit-Position: refs/heads/master@{#9165}
2015-05-10 09:06:20 +00:00
fd32f35aff Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit cdb47a4533b7b1e29e803ed6591a68bb1a4f1692.

Contains a tentative fix to the chrome build breakage caused by the
original change.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47139004

Cr-Commit-Position: refs/heads/master@{#9164}
2015-05-10 09:03:00 +00:00
cdb47a4533 Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit 208a2294cde839025318f1b3d57559cb0611a4e7.
Breaks the Chrome build.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53399004

Cr-Commit-Position: refs/heads/master@{#9161}
2015-05-08 12:03:46 +00:00
208a2294cd Adding a new constraint to set NetEq buffer capacity from peerconnection
This change makes it possible to set a custom value for the maximum
capacity of the packet buffer in NetEq (the audio jitter buffer). The
default value is 50 packets, but any value can be set with the new
functionality.

R=jmarusic@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50869004

Cr-Commit-Position: refs/heads/master@{#9159}
2015-05-08 10:58:51 +00:00
83b5c053b9 Modify NetEqQualityTest
- Take input sample rate as parameter - provides resampling when needed.
- Add support for wav output.

BUG=2692
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49699004

Cr-Commit-Position: refs/heads/master@{#9158}
2015-05-08 08:34:00 +00:00
2ea71c3279 Replace ACMGenericCodec with CodecOwner and AudioEncoderMutable
CodecOwner is introduced here; AudioEncoderMutable was introduced in a
previous commit, but had no users until now. The only remaining task
for ACMGenericCodec was to construct and maintain the stack of speech,
CNG, and RED encoders. This task is now handled by the CodecOwner,
which is owned and used by the CodecManager.

COAUTHOR=henrik.lundin@webrtc.org
BUG=4228
R=jmarusic@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43189004

Cr-Commit-Position: refs/heads/master@{#9152}
2015-05-07 13:49:24 +00:00
dcccab3ebb 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}
2015-05-07 10:35:18 +00:00