409 Commits

Author SHA1 Message Date
henrik.lundin
da8bbf6e3c NetEq: Change member variables for current RTP types to rtc::Optionals
With this change, the value 0xFF is no longer used to flag that the RTP
type is unknown. Instead, an empty value for the rtc::Optional is used.

Review-Url: https://codereview.webrtc.org/2290153002
Cr-Commit-Position: refs/heads/master@{#13989}
2016-08-31 10:14:18 +00:00
kwiberg
b8e56ee320 Fix Chromium clang plugin warnings
This adds a new file, webrtc/modules/audio_coding/neteq/tools/packet_source.cc, so that I'll have somewhere to put the new non-inlined methods.

NOTRY=true
BUG=webrtc:163

Review-Url: https://codereview.webrtc.org/2290593002
Cr-Commit-Position: refs/heads/master@{#13956}
2016-08-29 13:37:42 +00:00
ossu
09f1560f81 TimestampScaler no longer depends on NetEqDecoder to figure out scaling.
This is still a tiny lie, since it checks for kCodecArbitrary to avoid
scaling a codec with an external decoder, because of missing information
in that case. The main point is still true, though. Once the next CL is
in, removing NetEqDecoder usage completely in DecoderDatabase, another
workaround will be in place for external decoders until we can fully
deprecate them.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2270063006
Cr-Commit-Position: refs/heads/master@{#13952}
2016-08-29 10:59:09 +00:00
ossu
84bc98509b Removed virtual from several methods in DecoderDatabase to minimize
the number of points that need to be mocked for testing.

For the now non-virtual methods, DecoderDatabase now does a lookup
through GetDecoderInfo and then delegates to the appropriate method in
the DecoderInfo object, if one is found.

A few other methods were also changed to look up through GetDecoderInfo.

Also moved the audio decoder factory into DecoderInfo, so that
DecoderInfo::GetDecoder can be used directly.

Review-Url: https://codereview.webrtc.org/2276913002
Cr-Commit-Position: refs/heads/master@{#13933}
2016-08-26 12:41:30 +00:00
henrik.lundin
c76680419e NetEq: Update CNG code to accommodate 48 kHz sample rate
Checksums were updated for NetEq and ACM bitexactness tests (after
verifying the audio quality).

BUG=webrtc:5447

Review-Url: https://codereview.webrtc.org/2266293005
Cr-Commit-Position: refs/heads/master@{#13928}
2016-08-26 06:53:43 +00:00
henrik.lundin
8a6a600c16 Make neteq_rtpplay parse RTP header extensions
This removes the warning printouts about unknown header extensions.

BUG=webrtc:2692

Review-Url: https://codereview.webrtc.org/2266403005
Cr-Commit-Position: refs/heads/master@{#13912}
2016-08-25 07:46:41 +00:00
henrik.lundin
549d80b979 NetEq: only update current_rtp_payload_type_ when validated
The current_rtp_payload_type_ should only be updated when the packet is
actually inserted into the packet buffer, since then the payload type
has been validated. This CL removes an unvalidated setting of this value
that happened after SSRC change or upon first packet.

BUG=webrtc:5447

Review-Url: https://codereview.webrtc.org/2270793003
Cr-Commit-Position: refs/heads/master@{#13910}
2016-08-25 07:44:32 +00:00
henrik.lundin
5fac3f0892 NetEq: Don't check sample rate and frame size upon error
If an error happens in the GetAudio call, for instance when corrupt
payloads are inserted, GetAudio wil return an error. In this case, the
audio frame is not always correctly populated, which is to be expected.

BUG=webrtc:5447

Review-Url: https://codereview.webrtc.org/2272963002
Cr-Commit-Position: refs/heads/master@{#13902}
2016-08-24 18:18:54 +00:00
henrik.lundin
d1a10a0f77 Make FakeDecodeFromFile handle codec-internal CNG
This implementation interprets payloads of size 1 as codec-internal SID
frames, marking the start of a CNG period. Changes were made to other
parts of the test payload chain, since it had to make use of the virtual
payload size in the case of header-only RTP files.

BUG=webrtc:2692

Review-Url: https://codereview.webrtc.org/2275903002
Cr-Commit-Position: refs/heads/master@{#13901}
2016-08-24 17:59:00 +00:00
henrik.lundin
b3f1c5d2fe Add NetEq::FilteredCurrentDelayMs() and use it in VoiceEngine
The new method returns the current total delay (packet buffer and sync
buffer) in ms, with smoothing applied to even out short-time
fluctuations due to jitter. The packet buffer part of the delay is not
updated during DTX/CNG periods.

This CL also pipes the new metric through ACM and uses it in
VoiceEngine. It replaces the previous method of estimating the buffer
delay (where an inserted packet's RTP timestamp was compared with the
last played timestamp from NetEq). The new method works better under
periods of DTX/CNG.

Review-Url: https://codereview.webrtc.org/2262203002
Cr-Commit-Position: refs/heads/master@{#13855}
2016-08-22 22:40:00 +00:00
ehmaldonado
6c46eaa544 Add gtest as a dependency for neteq_quality_test_support.
Was removed in Patch Set 5 of https://codereview.webrtc.org/2252413002
but shouldn't have been, since it's actually required.

https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h?l=17

BUG=webrtc:6228
NOTRY=True

Review-Url: https://codereview.webrtc.org/2262173003
Cr-Commit-Position: refs/heads/master@{#13851}
2016-08-22 16:48:11 +00:00
ehmaldonado
861da3c662 Refactor neteq_test_support.
Take 'tools/neteq_quality_test.cc' and 'tools/neteq_quality_test.h' outside of neteq_test_support into their own target, neteq_quality_test_support.

BUG=webrtc:6228
NOTRY=True

Review-Url: https://codereview.webrtc.org/2252413002
Cr-Commit-Position: refs/heads/master@{#13831}
2016-08-19 14:02:31 +00:00
henrik.lundin
38d840c35a NetEq: Changing checked_cast to saturated_cast
The cast involves packet_len_samp, which is derived from the timestamps
and sequence numbers of incoming packets. Being values from the outside,
these should be treated as if any value is possible, making a
checked_cast unsuitable. Changing instead to saturated_cast to avoid
overflow with out-of-bounds values.

Review-Url: https://codereview.webrtc.org/2243403007
Cr-Commit-Position: refs/heads/master@{#13815}
2016-08-18 10:49:41 +00:00
flim
64a7eab891 Update tests and DTX check for Opus 1.1.3.
DTX is now indicated by packets that may have a size of up to 2 bytes.
Ref: https://git.xiph.org/?p=opus.git;a=commit;h=1c311423c86b89eba27a494e17c79fefd7d75ab0

BUG=

Review-Url: https://codereview.webrtc.org/2158293003
Cr-Commit-Position: refs/heads/master@{#13736}
2016-08-12 11:36:14 +00:00
aleloi
0e0be0a2f5 Migrated GN target :neteq_ilbc_quality_test
Migrated GN target :neteq_ilbc_quality_test from
webrtc/modules/audio_coding/neteq/neteq_tests.gypi

NOTRY=True

BUG=webrtc:6190

Review-Url: https://codereview.webrtc.org/2221193004
Cr-Commit-Position: refs/heads/master@{#13708}
2016-08-10 11:55:29 +00:00
aleloi
6391012514 Migrated GN target :audio_classifier_test
Migrated GN target :audio_classifier_test from
webrtc/modules/audio_coding/neteq/neteq_tests.gypi

NOTRY=True

BUG=webrtc:6190

Review-Url: https://codereview.webrtc.org/2225333002
Cr-Commit-Position: refs/heads/master@{#13706}
2016-08-10 11:41:23 +00:00
aleloi
116fd61599 Migrated GN target :neteq_speed_test
Migrated GN target :neteq_speed_test from
webrtc/modules/audio_coding/neteq/neteq.gypi

NOTRY=True

BUG=webrtc:6190

Review-Url: https://codereview.webrtc.org/2223343004
Cr-Commit-Position: refs/heads/master@{#13704}
2016-08-10 11:16:45 +00:00
aleloi
630c6d58d3 Migrated GN target :neteq_opus_quality_test
Migrated GN target :neteq_opus_quality_test from
webrtc/modules/audio_coding/neteq/neteq_tests.gypi

NOTRY=True

BUG=webrtc:6190, webrtc:2692

Review-Url: https://codereview.webrtc.org/2223933004
Cr-Commit-Position: refs/heads/master@{#13701}
2016-08-10 09:11:49 +00:00
aleloi
e5106e75da Removed target rtp_to_text. The code does not even run.
NOTRY=True

BUG=webrtc:6162, webrtc:2692

Review-Url: https://codereview.webrtc.org/2199803002
Cr-Commit-Position: refs/heads/master@{#13692}
2016-08-09 13:56:16 +00:00
aleloi
17dfa7425f Gyp->GN migration of RTPjitter.
BUG=6162

NOTRY=True

Review-Url: https://codereview.webrtc.org/2196283002
Cr-Commit-Position: refs/heads/master@{#13620}
2016-08-03 08:17:29 +00:00
aleloi
5556dcb24c Migrated GN target rtpcat.
BUG=6162

Review-Url: https://codereview.webrtc.org/2202193002
Cr-Commit-Position: refs/heads/master@{#13605}
2016-08-02 11:27:32 +00:00
aleloi
8266773f18 Migrated target RTPencode to GN
TBR=henrik.lundin@webrtc.org

BUG=6162

Review-Url: https://codereview.webrtc.org/2195353002
Cr-Commit-Position: refs/heads/master@{#13599}
2016-08-02 08:45:58 +00:00
flim
1cd46643d7 Disable some Opus tests pending an update
These tests will be reenabled and fixed after Opus 1.1.3 has landed in
Chromium and is rolled into WebRTC.

BUG=

Review-Url: https://codereview.webrtc.org/2185673002
Cr-Commit-Position: refs/heads/master@{#13534}
2016-07-26 16:01:45 +00:00
ossu
108ecec51c Removed LEGACY_BITEXACT from neteq_impl.cc and updated the ACM unit tests.
I'll be rewriting AcmReceiver soon and am trying to reduce the amount of
old stuff that needs to be supported.

I've manually checked the outputs of the AcmReceiver bitexactness
tests with this change. A large part of the tests are still bitexact,
with one section only differing slightly in timings. Nothing audible
unless playing the old and new versions back simultaneously.

The output of NetEqDecoderTest were also changed due to this CL, although only on android. I built and ran the test locally and compared the audio output manually - the changes were the same as for the other tests; i.e. very slight timing changes for a part of the output.

I updated the network stats checksum for android without analyzing it further. I expect it goes hand-in-hand with the changes to the output; i.e. the changes in it are fine because the audio output is fine. Likely, the stats will show changes in the usage of CNG, since that is what the code changes.

BUG=webrtc:1361

Review-Url: https://codereview.webrtc.org/2117763002
Cr-Commit-Position: refs/heads/master@{#13415}
2016-07-08 15:45:26 +00:00
henrik.lundin
d5f50a1b53 NetEq: Fix a bug in DelayPeakDetector causing asserts to trigger
In some situation, typically when incoming packets were reordered, the
DelayPeakDetector::Update method may be called twice (or more) with
non-zero inter_arrival_time argument, but without the TickTimer object
being updated in between (i.e., packets coming in more or less at the
same time). In these situations, a delay peak may be registered with
zero peak period. This could eventually trigger the DCHECK in
DelayPeakDetector::MaxPeakPeriod().

With this fix, the problem is solved by not registering peaks for which
the TickTimer object has not moved since the last peak.

The problem was originally introduced with
https://codereview.webrtc.org/1921163003.

BUG=webrtc:6021

Review-Url: https://codereview.webrtc.org/2085233002
Cr-Commit-Position: refs/heads/master@{#13257}
2016-06-22 16:07:07 +00:00
henrik.lundin
e8a77e3309 Refactor neteq_rtpplay
This change is a major refactoring of the neteq_rtpplay tool. It
consists of the following parts:

- NetEqTest class: Breaks out the main simulation loop from
  neteq_rtpplay into a separate class with well defined inputs and
  outputs.
- NetEqInput: Interface class for the input to NetEqTest.
- NetEqPacketSourceInput: Implementation of NetEqInput that provides a
  PacketSource objects with a NetEqInput interface. This has two
  subclasses; one for RtpFileSource and one for RtcEventLogSource.
- NetEqReplacementInput: An object that modifies the packets provided by
  another NetEqInput object, and replaces the packet payloads with meta
  data readable by a FakeDecodeFromFile decoder.
- FakeDecodeFromFile: An AudioDecoder implementation that produces
  "decoded" data by reading from an audio file.

BUG=webrtc:2692, webrtc:5447

Review-Url: https://codereview.webrtc.org/2020363003
Cr-Commit-Position: refs/heads/master@{#13252}
2016-06-22 13:34:08 +00:00
kwiberg
342f74005f NetEq: Ask AudioDecoder for sample rate instead of passing it as an argument
BUG=webrtc:5801
NOTRY=true

Review-Url: https://codereview.webrtc.org/2027993002
Cr-Commit-Position: refs/heads/master@{#13162}
2016-06-16 10:18:09 +00:00
kwiberg
347d35129e AudioDecoder: Remove the default implementation of SampleRateHz
And implement SampleRateHz in a bunch of mocks.

BUG=webrtc:5801
NOTRY=true

Review-Url: https://codereview.webrtc.org/2029543002
Cr-Commit-Position: refs/heads/master@{#13161}
2016-06-16 08:59:13 +00:00
henrik.lundin
919518613f NetEq: Rename Nack to NackTracker to avoid name collisions in GN
BUG=webrtc:5949
NOTRY=True
R=kjellander@webrtc.org

Review-Url: https://codereview.webrtc.org/2045243002
Cr-Commit-Position: refs/heads/master@{#13069}
2016-06-08 13:43:49 +00:00
kwiberg
c0f2dcf9ed NetEq decoder database: Don't keep track of sample rate for builtin decoders
This allows us to get rid of the function that computes it, which gets
us one step closer to getting rid of the NetEqDecoder type.

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/2021063002
Cr-Commit-Position: refs/heads/master@{#12974}
2016-05-31 13:28:09 +00:00
kwiberg
6c2eab34f8 AudioDecoder: New method SampleRateHz, + implementations for our codecs
This will let NetEq (and the factory, and every layer in between) keep
track of just the decoder, instead of decoder and sample rate.

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/2024633002
Cr-Commit-Position: refs/heads/master@{#12968}
2016-05-31 09:46:28 +00:00
henrik.lundin
303d3e1782 Fixing neteq_rtpplay
A regression happened in https://codereview.webrtc.org/2006723002,
causing neteq_rtpplay not to work. The problem was that when the main
code was moved inside of the webrtc::test namespace, it was no longer
visible to the linker. Meanwhile, the dependency on test_support_main
rather than test_support caused the executable to be a gtest.

In this fix, the gyp dependencies are corrected, and a main method is
added outside of the namespaces.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2018473002
Cr-Commit-Position: refs/heads/master@{#12918}
2016-05-26 12:56:07 +00:00
henrik.lundin
612c25e7af NetEq: Fix stats counting in muted mode
The NetEqNetworkStatistics::expand_rate was not incremented during muted
state, which caused under-reporting of that metric. This change fixes
that.

BUG=chromium:613321, webrtc:5608

Review-Url: https://codereview.webrtc.org/2003203004
Cr-Commit-Position: refs/heads/master@{#12894}
2016-05-25 15:21:09 +00:00
ossu
e352578bc8 Moved injection of AudioDecoderFactory into voe::Channel.
Channel's API remains unchanged, but the creation of a BuiltinAudioDecoderFactory is now in Channel. The next step would be to amend Channel's API (through CreateChannel, I believe) to allow an AudioDecoderFactory to be sent along.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/1992763002
Cr-Commit-Position: refs/heads/master@{#12893}
2016-05-25 14:37:47 +00:00
henrik.lundin
46ba49c622 Let PacketSource::NextPacket() return an std::unique_ptr
The return type of PacketSource::NextPacket() is changed from a naked
pointer to an std::uniqe_ptr. The interface contract was and still is
that the ownership is passed from the callee to the caller, but a
unique_ptr makes this explicit.

BUG=webrtc:2692

Review-Url: https://codereview.webrtc.org/2005873002
Cr-Commit-Position: refs/heads/master@{#12884}
2016-05-25 05:50:54 +00:00
henrik.lundin
ce5570e54e Move neteq_rtpplay.cc inside webrtc::test namespace
This simplifies the code.

BUG=webrtc:2692

Review-Url: https://codereview.webrtc.org/2006723002
Cr-Commit-Position: refs/heads/master@{#12873}
2016-05-24 13:15:03 +00:00
ossu
e725f7c73e Turned AudioDecoderFactory into a RefCounted thing to use with scoped_refptr.
First step of AudioDecoderFactory injection CLs. AudioDecoderFactories will be shared, and shared_ptr is currently off the table, so this CL changes the current uses of AudioDecoderFactory from std::unique_ptr to rtc::scoped_refptr.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/1990803004
Cr-Commit-Position: refs/heads/master@{#12815}
2016-05-19 17:48:12 +00:00
terelius
d5c1a0bd5d New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs.
Changed rtpdump converter and neteq tool to use new parser, but still aborting if the file is corrupt.

Review-Url: https://codereview.webrtc.org/1768773002
Cr-Commit-Position: refs/heads/master@{#12714}
2016-05-13 07:43:04 +00:00
henrik.lundin
7a926812d8 NetEq: Implement muted output
This CL implements the muted output functionality in NetEq. Tests are
added. The feature is currently off by default, and AcmReceiver makes
sure that the muted state is not engaged.

BUG=webrtc:5608

Review-Url: https://codereview.webrtc.org/1965733002
Cr-Commit-Position: refs/heads/master@{#12711}
2016-05-12 20:51:37 +00:00
minyue-webrtc
79553cb66e Using ring buffer for AudioVector in NetEq.
AudioVector used NetEq was based on a shift buffer, which has a high complexity, and the complexity is very much dependent on the capacity of the buffer.

This CL changes the shift buffer to a ring buffer.

Reduction in the CPU usages of NetEq is expected.

BUG=608644
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1948483002 .

Cr-Commit-Position: refs/heads/master@{#12676}
2016-05-10 17:56:10 +00:00
henrik.lundin
f3995f71ce NetEq: Implement Expand::Muted
Adding a new method to the Expand class, which will answer the question
whether an ongoing expansion has been faded down to zero
amplitude (i.e., been muted). Also adding a test.

This new functionality will be used in CLs to follow.

BUG=webrtc:5608
NOTRY=True

Review-Url: https://codereview.webrtc.org/1967473004
Cr-Commit-Position: refs/heads/master@{#12672}
2016-05-10 12:54:43 +00:00
henrik.lundin
60f6ce2a29 NetEq: Update stats earlier in the GetAudioInternal call
This is to prepare for implementation of NetEq muted state, which may
cause GetAudioInternal to make an early return just before the call to
GetDecision. With this change, the stats are updated in any case.

BUG=webrtc:5608
NOTRY=True

Review-Url: https://codereview.webrtc.org/1948663002
Cr-Commit-Position: refs/heads/master@{#12671}
2016-05-10 10:52:13 +00:00
Henrik Lundin
47b17dc59c NetEq: Replace timescale_holdoff_ with a Countdown timer
The timescale_holdoff_ is a counter in the DecisionLogic class. The
purpose is to enforce a minimum number of GetAudio calls
between (successfull) time-scaling operations (i.e., Accelerate and
Pre-emptive Expand operations). With this change, the counter is
replaced with a Countdown timer obtained from a TickTimer object.

BUG=webrtc:5608
R=tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1945863002 .

Cr-Commit-Position: refs/heads/master@{#12670}
2016-05-10 08:21:11 +00:00
pbos
c7a6569713 Revert of Disable failing modules_unittests for UBSan. (patchset #1 id:40001 of https://codereview.webrtc.org/1915813002/ )
Reason for revert:
Fix upstream should've landed in our repository.

Original issue's description:
> Disable failing modules_unittests for UBSan.
>
> BUG=webrtc:5820
> TBR=pbos@webrtc.org
>
> Committed: https://crrev.com/c23bf2e54d922486254cdd7657aafceaa958ce25
> Cr-Commit-Position: refs/heads/master@{#12482}

TBR=kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5820

Review-Url: https://codereview.webrtc.org/1937153002
Cr-Commit-Position: refs/heads/master@{#12647}
2016-05-06 19:50:08 +00:00
minyue
1746179c96 Reducing neteq sync buffer size.
BUG=608644

Review-Url: https://codereview.webrtc.org/1947453002
Cr-Commit-Position: refs/heads/master@{#12614}
2016-05-03 20:32:13 +00:00
henrik.lundin
b1fb72bebb NetEq: Move counting of generated CNG samples from DecisionLogic
The counting is moved to NetEqImpl, and the new counter is realized as a
Stopwatch object. The DecisionLogic class still has to maintain record
of when the CNG period is shortened, in order to reduce the delay. This
is recorded in a new noise_fast_forward_ member in DecisionLogic.

BUG=webrtc:5608

Review-Url: https://codereview.webrtc.org/1914303004
Cr-Commit-Position: refs/heads/master@{#12608}
2016-05-03 15:18:54 +00:00
kwiberg
5178ee86ba NetEq: Use a BuiltinAudioDecoderFactory to create decoders
Later steps in the refactoring will have the factory injected from the
outside rather than owned by NetEq.

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/1928293002
Cr-Commit-Position: refs/heads/master@{#12604}
2016-05-03 08:39:08 +00:00
minyue
5bd3397e53 Adding 120 ms frame length support in NetEq.
BUG=webrtc:1015

Review-Url: https://codereview.webrtc.org/1901633002
Cr-Commit-Position: refs/heads/master@{#12592}
2016-05-02 11:46:19 +00:00
minyue
53ff70f582 Reland "Avoiding overflow in cross correlation in NetEq."
The original CL is https://codereview.webrtc.org/1908623002/

An error was caused by that and this CL fix that problem and reland the CL.

BUG=

Review-Url: https://codereview.webrtc.org/1931933004
Cr-Commit-Position: refs/heads/master@{#12589}
2016-05-02 08:50:34 +00:00
kwiberg
bfefb03ec1 Replace scoped_ptr with unique_ptr everywhere
But keep #including scoped_ptr.h in .h files, so as not to break
WebRTC users who expect those .h files to give them rtc::scoped_ptr.

BUG=webrtc:5520

Review-Url: https://codereview.webrtc.org/1937693002
Cr-Commit-Position: refs/heads/master@{#12581}
2016-05-01 21:53:55 +00:00