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}
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}
uint32_t parameters don't need to be passed by reference. The
VCMJitterBuffer destructor doesn't need to be virtual because the
class has no virtual methods.
R=stefan@webrtc.org
BUG=none
Review URL: https://webrtc-codereview.appspot.com/55499004
Cr-Commit-Position: refs/heads/master@{#9288}
In my previous cl, https://webrtc-codereview.appspot.com/52479004/, there is 'UnLock()' left when we switched to scoped lock, which will cause TSan warning sometimes.
===========================================================
WARNING: ThreadSanitizer: unlock of an unlocked mutex (or by a wrong thread) (pid=9981)
#0 pthread_mutex_unlock <null> (libjingle_peerconnection_unittest+0x00000046836f)
#1 webrtc::CriticalSectionPosix::Leave() webrtc/system_wrappers/source/critical_section_posix.cc:39:10 (libjingle_peerconnection_unittest+0x000000bc368d)
#2 ~CriticalSectionScoped webrtc/system_wrappers/interface/critical_section_wrapper.h:46:48 (libjingle_peerconnection_unittest+0x000000a61fcb)
#3 webrtc::AudioDeviceLinuxPulse::RecThreadProcess() webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc:3003 (libjingle_peerconnection_unittest+0x000000a61fcb)
===========================================================
BUG=3056
TEST=bots
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/56439004
Cr-Commit-Position: refs/heads/master@{#9282}
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}
tl;dr - non-continuous frames (due to padding) would get stuck as incomplete if the previous complete frame arrived and was decoded before the padding arrived.This fix re-checks the incomplete frame list for continuous frames after old packets arrive.
When padding is enabled and RTX is not, padding is sent as empty RTP packets tacked onto the end of completed frames (meaning: same timestamp, but after a packet with the marker bit set). Given the following set of circumstances, codified in the new unit test method, a frame can get permanently stuck in the incomplete frames list:
- Frame A decoded (packets 94-95). Next expected sequence number is 96.
- Frame C arrives (packets 100-101) and is marked complete. It isn't continuous, since it starts at 100, so it's placed in the incomplete frame list.
- Frame B arrives (packets 96-97) and is complete, since 97 has a marker bit. Turns out that packets 98-99 are padding, but the receiver doesn't know that.
- Frame B is decoded, removed from the decodable frames list, and last decoded state is updated.
- Packets 98-99 arrive. They hit the IsOldPacket check and update the last decoded state, but they don't trigger FindAndInsertContinuousFrames.
- Further packets/frames arrive and complete, but FindAndInsertContinuousFrames only runs on frames that are newer than the newly completed frame.
In this state, Frame C is permanently stuck as incomplete, so the jitter buffer overall is stuck until max NACK age (default: 450 packets), the max NACK list size (default: 200 packets), or a keyframe arrives and IsContinuous returns true for the keyframe.
(Before the November refactoring, Frame B wouldn't have to have been decoded for the bug to trigger; just having a complete continuous frame at any time before the padding arrived would cause this state, as FindAndInsertContinuousFrames was only called when the frame originally became continuous and was inserted into the decodable frames list. Post refactoring, the frame is removed/re-added to the decodable list on every padding packet that arrives)
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50959004
Cr-Commit-Position: refs/heads/master@{#9264}
Because of the Finch experiment, this will not affect Chrome's behaviour at all.
The SNRs in AudioProcessingTest.Formats were only increased to the next multiple of 5.
BUG=webrtc:3146
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43359004
Cr-Commit-Position: refs/heads/master@{#9263}
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}
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}
This CL change the function declaration from uint16_t to size_t, and CHECKs that the size fits in uint16_t before proceeding.
BUG=484432
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47229004
Cr-Commit-Position: refs/heads/master@{#9246}
BUG=3056, 1320
TEST=AutoTest
Mainly add threadchecker and remove unnecessary lock.
And some more styling working.
- audio_device_pulse_linux.cc: wrap lines longer than 80 chars. And add '.' to some comments around. Not do it to all places.
- audio_mixer_manager_pulse_linux.cc: Here I adopt some chromium practice. We use to do many things to the failure of pulse operation, which causes most of the data race issue. In chromium, if we failed to call any pulse function, we just fail it w/o use the previous results. Here I did same. Please check if it's good.
R=bjornv@webrtc.org, henrika@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52479004
Cr-Commit-Position: refs/heads/master@{#9243}
It is necessary for adding 48kHz support to the AudioProcessing::AnalyzeReverseStream int interface (It was not necessary for 32kHz since in that case the splitting filter is more efficient).
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/56389004
Cr-Commit-Position: refs/heads/master@{#9241}
Avoids existing crash and ensures that error message is passed up to Libjingle. Will lead to the following logcat output:
E/libjingle(31404): Error(channel.cc:1514): Failed to SetSend 2 on voice channel
BUG=b/21273153
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54459004
Cr-Commit-Position: refs/heads/master@{#9236}
compile time.
The condition of static_assert() is evaluated at compile time which is safer and
more efficient.
Note that static_assert() requires C++11.
The changes were generated by the misc-static-assert ClangTidy check by alexfh@google.comR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51019004
Cr-Commit-Position: refs/heads/master@{#9231}
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}
The class doesn't do anything in almost all cases except for grabbing and releasing locks + allocate memory. There are a couple of methods there such as WaitForKey and GetTimeInMs that are used, but those methods aren't specific to audio and we have implementations of these elsewhere. The third method, StringCompare isn't used anywhere (and also isn't specific to audio).
BUG=
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50009004
Cr-Commit-Position: refs/heads/master@{#9220}
From 6kHz-6.5kHz to 3kHz-5kHz. Previous range had unreliable mask values, letting high frequencies from all directions through. The new range is wider and lower, which results in better estimates.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47089004
Cr-Commit-Position: refs/heads/master@{#9213}
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
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}
