| Commit message (Collapse) | Author | Age | Files | Lines |
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o related-to-bug: 6383440
Change-Id: I0b1ae50a704b0979857ea447585eabe86602149a
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and fill in the OMX channel mask properly.
Change-Id: I915950a0b252142b9eb3277cf7c6e0d9f5875305
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Change-Id: I3c09da1dc0de5039d0c15ce7fb2bc373fa398712
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Change-Id: I97ce29ecfde4df171120a3570db9575b32c8d362
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The C++ APIs are going away.
Note: we use tid == 0 which is not supported yet by the C APIs,
do not submit this until that is added.
Change-Id: I0e90789e6c81c69f2544e899c52421ea5d1342be
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In DOWNMIX_TYPE_FOLD mode, add support for variants of quad and
5.1 channel masks where the "side" channels are used instead of
the "back" channels. These variants have the same channel order,
so they can share the same downmix function.
Change-Id: I5f9cb7cb120927189de680c2a968d6136dc1fc43
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Bug: 6379646
Change-Id: I12b53bc4118499ddc73a53a981f3f56328140868
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Change-Id: If261c4d87d5569a0d7cfd75afe311803a6ce0109
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Bug: 6378847
Change-Id: Ib3324c7a5951ac95f99993bfae27ea02bfda08ff
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Change-Id: I3131bb22d2d057e9197a2ebfa6aa1cfaab9e5321
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want to revisit it."
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revisit it.
Change-Id: Iab62562c4dea13520751c37c9448c0b19c82d739
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Change-Id: Ifd825590ba36996064a458f64453a94b84722cb0
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Made AudioRecord a subclass of RefBase to allow using strong
references and solve concurrency issues.
Issue 6254582.
Change-Id: Ic1f3845958f477e8b2d23d3d25bf0f666addcb3b
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Allow AudioSink to use deep audio buffering when the
source is audio only and its duration is more than
a certain threshold.
This helps improve battery life but implies higher
audio latency.
Change-Id: Ie79915b61c370292f05aabda9779356570e03cbb
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Change-Id: I0966270a2bd36698330beaa03f47d02fc6a40c44
related-to-bug: 6364341
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Hand merge from ics-aah
> TimedAudioTrack: Optimize the queue trim operation.
>
> Don't perform the end PTS calculation for each buffer during trimming.
> Instead, only calculate the ending PTS of a buffer if there is no next
> buffer in the queue. This optimization assumes that the buffers being
> queued are in monotonic media time order (a fair assumption for now)
> and that the timestamps in the audio are contiguous (not a requirement
> for this API, but a reality of how it is being used right now).
>
> In the case where the audio is discontinuous on purpose, it is
> that this optimization will cause the system hold one extra buffer
> which it could have safely trimmed. It should not be much of an issue
> since in real life the audio is almost always contiguous, and as long
> as the media clock is running and the mixer is mixing, the buffer will
> be used up and discard as part of the normal flow anyway.
>
> Change-Id: I00061e85ee7d5651fcf80751646c7d7415894a14
> Signed-off-by: John Grossman <johngro@google.com>
Change-Id: I0054b58e1389fa005aa990cb5710caf4af7b706a
Signed-off-by: John Grossman <johngro@google.com>
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Hand merge from ics-aah
> TimedAudioTrack: Fix continuity threshold handling.
>
> Fix issues with continuity threshold handling; notably
> + If the steady-state continuity threshold is exceeded, be sure to
> clear the on-time flag. Failure to do this will result in the
> system picking a new mix point which simply satisfies the
> steady-state continuity threshold instead of the startup threshold.
> Since we are putting a discontinuity in presentation anyway, we
> really want to pick a perfect point, not just an OK point.
> + Tighten the steady-state continuity threshold. It was currently set
> to 100mSec which is enormous. 4mSec (the new setting) is much more
> appropriate. On systems with a VCXO (like tungsten) this should
> never be wrong by more than a sample. If TimedAudioTracks are ever
> to be used on VCXO-less systems, this threshold should probably be a
> a parameter configurable by applications on a track by track basis
> so they can make the tradeoff between allowed error and frequency of
> disruptive corrections.
> + Reset the on-time flag if the mixer provides no PTS during a mix
> operation. This makes for a convenient way for the HAL to reset
> timed tracks when it makes changes for delay compensation across
> multiple outputs.
>
> Change-Id: I2cb23de5a3d1f75618abc1c8ab903db883837aa8
> Signed-off-by: John Grossman <johngro@google.com>
Change-Id: Ibd28c9d290494b0b19eb01caf2d9bfdef606a9b4
Signed-off-by: John Grossman <johngro@google.com>
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Updates:
- Add support for mono fast tracks
- Add support for optional sample rate conversion on fast tracks
- Log sample rate and frame count
- Enable statistics
Change-Id: Ife014edf4f452da361f3eaaae19209ef6ff6958b
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CryptoPlugin to"
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the higher layers.
Change-Id: I9f434ad55cdf575803c208bedf47b607baff2330
related-to-bug: 6365261
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This reverts commit f356cc9c9995ba292bf116f1cb28a6caa1b65fd2.
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o related-to-bug: 5542712
Change-Id: Ifdc730e616336e8b16783e8a5766f27df62a0592
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Change-Id: I4eda43166d23b3297cf769056a6db720401799aa
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Support VISUALIZER_PARAM__SCALING_MODE parameter.
Modify process() function to use new volume scaling parameter,
used to set whether captured values are maximized by current
headroom in the buffer (default existing behaviora) or left
as is and thus affected by volume.
Modify AudioEffect to allow subclasses to override the following
events: control status changed, enable status changed, command
executed.
In Visualizer class (a subclass of AudioEffect), reset the
scaling mode and capture size on the actual effect
as cached when control is regained.
This will allow the effect to be properly
configured whenever Visualizers with different scaling
modes or capture sizes are released (e.g. from java
release() method).
Change-Id: I05cb9f925a296dceca91bafa9fe294ef2b2c2bd5
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Change-Id: I5c7f0360883fc68e313d10b1aa5dac1463fae7fb
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Applications are not resumed under the lock screen now.
This API is not needed anymore.
bug:5584464
Change-Id: I115daf6b647348617ec0fc05b626878c945b9b29
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Fix bug with pausing audio
Change-Id: Icd6b095dac8d1a68b027de853d11ae02cc070b10
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Pausing a video player will cause the track audio mixer to be
disabled, which causes the downmixer to be deleted. When reenabled,
the track channel mask hasn't changed but the downmixer is there
anymore.
Fixed by:
- instanciating a downmixer when the AudioMixer track
gets initialized (in getTrackName(), now taking a channel mask
as input), and deleted when in deleteTrackName().
- when the channel changes on a track, check whether it
needs a downmixer or not. Preparing a track for downmix
automatically removes the old downmixer if there was one.
Also: initialize the track downmixerBufferProvider field
when AudioMixer is instanciated, so we can safely call
delete on it in AudioMixer's destructor, in case
deleteTrackName() wasn't called before the mixer was
destroyed.
Change-Id: I589b0781cda5b3c82f85b561c52b08546cac21f8
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Change-Id: I27c46bd1d1b2b5f96b87af7d05b951fef18a1312
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BufferQueue is no longer setting these for its endpoints; set the bits
to GRALLOC_USAGE_HW_VIDEO_ENCODER | GRALLOC_USAGE_HW_TEXTURE. In the
longer term, the set flags should be just
GRALLOC_USAGE_HW_VIDEO_ENCODER, but that requires other changes.
Change-Id: I4170658de49bf88d829d34605a15684e72a90706
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