Reason for revert:
voice_engine_unittests: FilePlayerTest.PlayWavPcm16File and FilePlayerTest.PlayWavPcmuFile fail on 32-bit android (android_rel and android-dbg try bots, Android32 Tests (L Nexus5) and Android32 Tests (L Nexus7.2) build bots).
Not sure why this would happen, since I just moved the test without modifying it. Some test filtering that no longer manages to disable them? Anyway, reverting until I know how to fix.
This was actually caught by the try bots, but I missed it because I was manually ignoring them because of an error with the bots. :-(
Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> R=perkj@webrtc.org, solenberg@webrtc.org
>
> Committed: 65874b163eTBR=perkj@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2092633002
Cr-Commit-Position: refs/heads/master@{#13267}
- 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}
- 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}
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
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
* VoE can now exchange 44.1 kHz audio with AudioDevice.
* Changes still required in AudioDevice to remove the 44 kHz workarounds and
enable native 44.1 kHz.
BUG=webrtc:1395
TESTED=voe_cmd_test loopback running through codecs using all combinations of {8, 16, 32} kHz and {1, 2} channels, and Opus (48 kHz, stereo)
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1373004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3930 4adac7df-926f-26a2-2b94-8c16560cd09d
* Remove the unneeded _mixingFrequency.
* Rename CheckForSendCodecChanges to better elucidate its function.
* Remove an unnecessary memcpy.
Upsampling should be done late in the chain. This is practically relevant
on mobile, where the capture rate is fixed at 16 kHz. When using Opus, the
signal was upsampled to 32 kHz and was no longer compatible with AECM, which only supports up to 16 kHz.
NEEDS_QA=true
TEST=run calls with a variety of capture device rates and codecs
BUG=chromium:178040,webrtc:1446
Review URL: https://webrtc-codereview.appspot.com/1146004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3594 4adac7df-926f-26a2-2b94-8c16560cd09d
