First approach to remove parts of the heavy load done for encoding, and
preparation for sending, from native audio thread to separate task queue.
With this change we will give the native input audio thread more time to
"relax" between successive audio captures.
Separate profiling done on Android has verified that the change works well;
the load is now redistributed and the load of the native AudioRecordThread
is reduced. Similar conclusions should be valid for all other OS:es as well.
BUG=NONE
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng
Review-Url: https://codereview.webrtc.org/2665693002
Cr-Commit-Position: refs/heads/master@{#17488}
This makes a few things a lot clearer when looking at perf trace data:
* What module instances (where they were created) are called
* On what thread
* How frequently
* For how long
ProcessThread will be replaced by TaskQueue moving forward and this is a step towards understanding the behavior of the affected code.
BUG=webrtc:7219
Review-Url: https://codereview.webrtc.org/2729053002
Cr-Commit-Position: refs/heads/master@{#16998}
The class basically implements a timer and can be replaced with a PostDelayedTask call down the line.
BUG=none
Review-Url: https://codereview.webrtc.org/2722613002
Cr-Commit-Position: refs/heads/master@{#16891}
This CL is in preparation to move the AudioFrame into webrtc/api. The
AudioFrame is a POD type used for representing 10ms of audio. It
appears as a parameter and return value of interfaces being migrated
to webrtc/api, in particular AudioMixer.
Here, methods operator+=, operator>>=, Mute are
moved into a new target webrtc/audio/utility/audio_frame_operations,
and dependencies are changed to use
the new versions. The old AudioFrame methods are marked deprecated.
The audio frame utilities in webrtc/modules/utility:audio_frame_operations
are also moved to the new location.
TBR=kjellander@webrtc.org
BUG=webrtc:6548
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2424173003
Cr-Commit-Position: refs/heads/master@{#15413}
Mostly, it's about replacing mutable reference arguments with pointer
arguments, and replacing C style casts with C++ style casts.
Review-Url: https://codereview.webrtc.org/2056653002
Cr-Commit-Position: refs/heads/master@{#13849}
Because passing ownership in raw pointers makes kittens cry.
This also means we can ditch the Destroy functions and the protected
destructors. (Well, almost. We need to keep the old CreateFilePlayer
and DestroyFilePlayer around for a little while longer because of an
external caller.)
Review-Url: https://codereview.webrtc.org/2049683003
Cr-Commit-Position: refs/heads/master@{#13797}
Muting/unmuting is triggered in the PeerConnection API by calling setEnable() on an audio track.
BUG=webrtc:5671
Review URL: https://codereview.webrtc.org/1810413002
Cr-Commit-Position: refs/heads/master@{#12121}
- Clean up unused methods in voe::Channel following removal of VoEDtmf APIs.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1785643006
Cr-Commit-Position: refs/heads/master@{#11976}
The restriction has been removed completely and AECM now supports any
number of higher bands. But this has been achieved by always zeroing out the
higher bands, instead of applying a constant gain which is the average over half
of the lower band (like it is done for the AEC), because that would be
non-trivial to implement and we don't want to spend too much time on AECM, since
we want to get rid of it in the long term anyway.
R=peah@webrtc.org, solenberg@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1774553002 .
Cr-Commit-Position: refs/heads/master@{#11931}
- Use better types in AudioSendStream::SendTelephoneEvent() and related methods.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1722253002
Cr-Commit-Position: refs/heads/master@{#11927}
Also introduce a pair of scoped_ptr <-> unique_ptr conversion
functions. By using them judiciously, we can keep these CL:s small and
avoid having to convert enormous amounts of code at once.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1702983002
Cr-Commit-Position: refs/heads/master@{#11658}
Also remove mischievous tab character!
This is a part of getting rid of CriticalSectionWrapper and makes the code slightly simpler.
BUG=
Review URL: https://codereview.webrtc.org/1607353002
Cr-Commit-Position: refs/heads/master@{#11346}
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
Removes ShouldIgnoreTrace from WebRtcVoiceEngine and removes the spammy
log instances instead. Also removes trace-style logging from getters
(::GetLocalSSRC() for instance would print what SSRC it got, spamming
the log).
BUG=
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1347353004 .
Cr-Commit-Position: refs/heads/master@{#10028}
* Added a way to notify a Module that it's been attached to a ProcessThread.
The benefit of this is to give the module a way to wake up the thread
when it needs work to happen on the worker thread, immediately.
Today, module instances are typically registered with a process thread
outside the control of the modules themselves. I.e. they typically
don't know about the process thread they're attached to.
* Improve ProcessThread's WakeUp algorithm to not call TimeUntilNextProcess
when a WakeUp call is requested. This is an optimization for the above
case which avoids the module having to acquire a lock or do an interlocked
operation before calling WakeUp(), which would ensure the module's
TimeUntilNextProcess() implementation would return 0.
BUG=2822
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39239004
Cr-Commit-Position: refs/heads/master@{#8527}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8527 4adac7df-926f-26a2-2b94-8c16560cd09d
---
Fixes for re-enabling more MSVC level 4 warnings: webrtc/ edition
This contains fixes for the following sorts of issues:
* Possibly-uninitialized local variable
* Signedness mismatch
* Assignment inside conditional
This also contains a small number of other cleanups to nearby code. In
particular several warning-disables for MSVC are removed because they don't seem
to be necessary (either that warning is not enabled or the code does not trigger
it).
BUG=crbug.com/81439
TEST=none
R=henrika@webrtc.org, pkasting@chromium.org
Review URL: https://webrtc-codereview.appspot.com/18769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6667 4adac7df-926f-26a2-2b94-8c16560cd09d
Replace the two versions with a single DownConvertToCodecFormat. As
mentioned in comments, this could be further consolidated with
RemixAndResample but we should write a full audio converter class in
that case.
Along the way:
- Fix the bug present in Channel::Demultiplex with mono input and a
stereo codec.
- Remove the 32 kHz max from the OnDataAvailable path. This avoids a
48 -> 32 -> 48 conversion when VoE is passed 48 kHz audio; instead we
get a straight pass-through to ACM. The 32 kHz conversion is still
needed in the RecordedDataIsAvailable path until APM natively supports
48 kHz.
- Merge resampler improvements from ACM1 to ACM2. This allows ACM to
handle 44.1 kHz audio passed to VoE and was originally done here:
https://webrtc-codereview.appspot.com/1590004
- Reuse the RemixAndResample unit tests for DownConvertToCodecFormat.
- Remove unused functions from utility.cc.
BUG=3155,3000,b/12867572
TESTED=voe_cmd_test using both the OnDataAvailable and
RecordedDataIsAvailable paths, with a captured audio format of all
combinations of {44.1,48} kHz and {1,2} channels, running through all
codecs, and finally using both ACM1 and ACM2.
R=henrika@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11019005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5843 4adac7df-926f-26a2-2b94-8c16560cd09d
This will allow an embedder to use it directly.
Adding inertia/hangover time between updates of the reported detection status to the algorithm, controlled by a parameter. That is usually desired and this way a consumer of
the class don't have to implement that. (VoiceEngine will let it be 1, which results in the same behavior as before, and keep controlling the hangover itself.)
R=andrew@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5462 4adac7df-926f-26a2-2b94-8c16560cd09d
- Remove one unneeded lock in CaptureLevel(), as the call to this
method should always come on the same thread as PrepareDemux().
- Remove check on analog AGC before doing volume calculations. Saves a
bit of code. Instead check if the incoming volume is set to zero, which
is a potentially common occurrence as it indicates no volume is
available.
R=aluebs@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5366 4adac7df-926f-26a2-2b94-8c16560cd09d
Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)
Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.
TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.
R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
- Remove unneeded #defines and switch the remainder to consts.
- All AudioProcessing components are disabled by default, so remove
explicit disables.
- AudioProcessing uses a rational 16 kHz mono default, so no need to
explictly initialize.
- Add assert(false) to real-time errors which should not occur.
TESTED=trybots
R=bjornv@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2253005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4924 4adac7df-926f-26a2-2b94-8c16560cd09d
The complexity of the last ChannelManager and potentially usage of it as well caused race conditions and deadlocks in loopback voe_auto_test. This ref-counted solution takes no long-term locks, uses less locks overall and is significantly easier to understand.
ScopedChannel has been split up into a ChannelOwner with a reference to a channel and an Iterator over ChannelManager. Previous code was really used for both things. ChannelOwner is used as a shared pointer to a channel object, while an Iterator should work as expected.
BUG=2081
R=tommi@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1802004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4502 4adac7df-926f-26a2-2b94-8c16560cd09d
r4326 was mistakenly committed to stable, so this is to re-merge back to trunk.
Fixed the AGC and interface problems on the new path.
In order to make the AGC work properly, we need to cache the volume value passed
by the callback, compare it with the value returned by
shared->transmit_mixer()->CaptureLevel(). If they are the same, we need to
return 0 to indicate no volume needs changing, otherwise return the new volume.
By doing this, we avoid setting the volume all the same, which allows the users
to change the volume manually.
This patch also fixes some minor issues with the interfaces too: make the int
channel[] const, and correct the order of the input params in
channel::Demultiplex.
R=tommi@webrtc.org
BUG=[2134]
TEST=compile && manual AGC test
Review URL: https://webrtc-codereview.appspot.com/1921004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4450 4adac7df-926f-26a2-2b94-8c16560cd09d
r4326 was mistakenly committed to stable, so this is to re-merge back to trunk.
Add new interface to support multiple sources in webrtc.
CaptureData() will be called by chrome with a flag |need_audio_processing| to
indicate if the data needs to be processed by APM or not. Different from the old
interface that will send the data to all voe channels, the new interface will
specify a list of voe channels that the data is demultiplexing to.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4449 4adac7df-926f-26a2-2b94-8c16560cd09d
