Cleanup of the AudioDeviceBuffer class

Cleanup of the AudioDeviceBuffer class.

WebRTC works on 10ms buffer sizes in both directions but this class has contained
support for any size (with some limits) and for changes on the fly. It makes no sense to maintain such code and we have no tests to test it. This CL ensures that only 10ms audio buffers are supported and that nothing can be changed on the fly.

It also updates the style to follow the Google C++ style guide.

Finally, I remove very old (not tested and not maintained) support for file
handling since the code is never used. It was more or less dead code.

BUG=NONE
R=magjed@webrtc.org

Committed: https://crrev.com/cf327b45b9f5738950d4fca2b6a7b6030d508cdf
Cr-Commit-Position: refs/heads/master@{#13833}

[henrika_webrtc, 2016-08-18 09:36:03]

Description was changed from

==========
Cleanup of the AudioDeviceBuffer class

BUG=
==========

to

==========
Cleanup of the AudioDeviceBuffer class

BUG=
==========

[henrika_webrtc, 2016-08-18 12:17:44]

Description was changed from

==========
Cleanup of the AudioDeviceBuffer class

BUG=
==========

to

==========
Cleanup of the AudioDeviceBuffer class.

WebRTC works on 10ms buffer sizes in both directions but this class has
contained
support for any size (with some limits) and for changes on the fly. It makes no
sense to maintain such code and we have no tests to test it. This CL ensures
that only 10ms audio buffers are supported and that nothing can be changed on
the fly.

It also updates the style to follow the Google C++ style guide.

Finally, I remove very old (not tested and not maintained) support for file
handling since the code is never used. It was more or less dead code.


BUG=NONE
==========

[henrika_webrtc, 2016-08-18 12:17:44]

henrika@webrtc.org changed reviewers:
+ magjed@webrtc.org

[henrika_webrtc, 2016-08-18 12:18:02]

PTAL

[magjed_webrtc, 2016-08-18 14:07:01]

https://codereview.webrtc.org/2256833003/diff/20001/webrtc/modules/audio_devi...
File webrtc/modules/audio_device/audio_device_buffer.h (right):

https://codereview.webrtc.org/2256833003/diff/20001/webrtc/modules/audio_devi...
webrtc/modules/audio_device/audio_device_buffer.h:54: virtual int32_t
SetRecordedBuffer(const void* audio_buffer,
Why are many of these functions virtual? I don't find any subclasses of
AudioDeviceBuffer.

Are you planning to remove num_samples as an argument if it must always be 10ms
anyway?

[henrika_webrtc, 2016-08-18 14:22:00]

Thanks. PTAL ;-)

https://codereview.webrtc.org/2256833003/diff/20001/webrtc/modules/audio_devi...
File webrtc/modules/audio_device/audio_device_buffer.h (right):

https://codereview.webrtc.org/2256833003/diff/20001/webrtc/modules/audio_devi...
webrtc/modules/audio_device/audio_device_buffer.h:54: virtual int32_t
SetRecordedBuffer(const void* audio_buffer,
Virtual? Legacy ;-) Very old code where I assume the intention was to create
platform-specific versions and use this class as base. Removing all references
to virtual.

I would like to keep num_samples to be able to DCHECK that the audio layer is
actually using 10ms on its side. It will lead to a large amount of changes
otherwise.

[magjed_webrtc, 2016-08-19 07:51:15]

lgtm

https://codereview.webrtc.org/2256833003/diff/20001/webrtc/modules/audio_devi...
File webrtc/modules/audio_device/audio_device_buffer.h (right):

https://codereview.webrtc.org/2256833003/diff/20001/webrtc/modules/audio_devi...
webrtc/modules/audio_device/audio_device_buffer.h:54: virtual int32_t
SetRecordedBuffer(const void* audio_buffer,
On 2016/08/18 14:22:00, henrika_webrtc wrote:
> Virtual? Legacy ;-) Very old code where I assume the intention was to create
> platform-specific versions and use this class as base. Removing all references
> to virtual.
> 
> I would like to keep num_samples to be able to DCHECK that the audio layer is
> actually using 10ms on its side. It will lead to a large amount of changes
> otherwise.

Alright.

[henrika_webrtc, 2016-08-19 08:12:18]

Thanks. Looks like I do break some tests. Will check and try to fix.

[henrika_webrtc, 2016-08-19 08:12:19]

Thanks. Looks like I do break some tests. Will check and try to fix.

[henrika_webrtc, 2016-08-19 13:24:37]

The CQ bit was checked by henrika@webrtc.org

[henrika_webrtc, 2016-08-19 13:24:37]

The patchset sent to the CQ was uploaded after l-g-t-m from magjed@webrtc.org
Link to the patchset: https://codereview.webrtc.org/2256833003/#ps100001 (title:
"Ensures that all tests passes")

[commit-bot: I haz the power, 2016-08-19 13:24:50]

CQ is trying da patch. Follow status at
 
https://chromium-cq-status.appspot.com/v2/patch-status/codereview.webrtc.org/...

[commit-bot: I haz the power, 2016-08-19 14:12:40]

The CQ bit was unchecked by commit-bot@chromium.org

[commit-bot: I haz the power, 2016-08-19 14:12:41]

Try jobs failed on following builders:
  android_arm64_rel on master.tryserver.webrtc (JOB_TIMED_OUT, no build URL)

[commit-bot: I haz the power, 2016-08-19 14:38:11]

Description was changed from

==========
Cleanup of the AudioDeviceBuffer class.

WebRTC works on 10ms buffer sizes in both directions but this class has
contained
support for any size (with some limits) and for changes on the fly. It makes no
sense to maintain such code and we have no tests to test it. This CL ensures
that only 10ms audio buffers are supported and that nothing can be changed on
the fly.

It also updates the style to follow the Google C++ style guide.

Finally, I remove very old (not tested and not maintained) support for file
handling since the code is never used. It was more or less dead code.


BUG=NONE
==========

to

==========
Cleanup of the AudioDeviceBuffer class.

WebRTC works on 10ms buffer sizes in both directions but this class has
contained
support for any size (with some limits) and for changes on the fly. It makes no
sense to maintain such code and we have no tests to test it. This CL ensures
that only 10ms audio buffers are supported and that nothing can be changed on
the fly.

It also updates the style to follow the Google C++ style guide.

Finally, I remove very old (not tested and not maintained) support for file
handling since the code is never used. It was more or less dead code.

BUG=NONE
R=magjed@webrtc.org

Committed: https://crrev.com/cf327b45b9f5738950d4fca2b6a7b6030d508cdf
Cr-Commit-Position: refs/heads/master@{#13833}
==========

[henrika_webrtc, 2016-08-19 14:38:11]

Description was changed from

==========
Cleanup of the AudioDeviceBuffer class.

WebRTC works on 10ms buffer sizes in both directions but this class has
contained
support for any size (with some limits) and for changes on the fly. It makes no
sense to maintain such code and we have no tests to test it. This CL ensures
that only 10ms audio buffers are supported and that nothing can be changed on
the fly.

It also updates the style to follow the Google C++ style guide.

Finally, I remove very old (not tested and not maintained) support for file
handling since the code is never used. It was more or less dead code.


BUG=NONE
==========

to

==========
Cleanup of the AudioDeviceBuffer class.

WebRTC works on 10ms buffer sizes in both directions but this class has
contained
support for any size (with some limits) and for changes on the fly. It makes no
sense to maintain such code and we have no tests to test it. This CL ensures
that only 10ms audio buffers are supported and that nothing can be changed on
the fly.

It also updates the style to follow the Google C++ style guide.

Finally, I remove very old (not tested and not maintained) support for file
handling since the code is never used. It was more or less dead code.


BUG=NONE
R=magjed@webrtc.org

Committed:
https://chromium.googlesource.com/external/webrtc/+/cf327b45b9f5738950d4fca2b...
==========

[commit-bot: I haz the power, 2016-08-19 14:38:12]

Patchset 6 (id:??) landed as
https://crrev.com/cf327b45b9f5738950d4fca2b6a7b6030d508cdf
Cr-Commit-Position: refs/heads/master@{#13833}

[henrika_webrtc, 2016-08-19 14:38:12]

Committed patchset #6 (id:100001) manually as
cf327b45b9f5738950d4fca2b6a7b6030d508cdf (presubmit successful).

[henrika_webrtc, 2016-08-19 15:09:07]

A revert of this CL (patchset #6 id:100001) has been created in
https://codereview.webrtc.org/2260183002/ by henrika@webrtc.org.

The reason for reverting is: Seems to break an external client..

[henrika_webrtc, 2016-08-22 11:14:01]

Patchset #7 (id:120001) has been deleted