| Commit message (Collapse) | Author | Age | Files | Lines |
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Change-Id: I7d8201590cda29c9fa99662a4fdba222091febfe
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Use it in AudioSystem::getOutput(), AudioSystem::getInput(),
IAudioPolicyService::getOutput(), IAudioPolicyService::getInput(),
and various other places in AudioFlinger.
Not done: AudioTrack and OutputDescriptor.
Change-Id: I70e83455820bd8f05dafd30c63d636c6a47cd172
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Change-Id: I48b193a742b32b6746aa938b84dc405124a6a5c3
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In IAudioFlinger::createTrack() and IAudioFlinger::openRecord(),
declare input parameter to use correct type audio_channel_mask_t.
In IAudioFlinger::getInputBufferSize(), input parameter is now channel mask
instead of channel count.
Remove unused IAudioFlinger::channelCount(audio_io_handle_t).
In AudioRecord::getMinFrameCount() and AudioSystem::getInputBufferSize(),
input parameter is channel mask instead of channel count.
Change-Id: Ib2f1c29bea70f016b3cfce83942ba292190ac965
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mRecord is not properly initialized if the call to AudioRecord::getMinFrameCount() fails.
media server crashes when the unintialized mRecord object is deleted in AudioSource's destructor.
Change-Id: Ia89222789d044c11c9957a99725bc89f9c709e17
related-to-bug: 6744014
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When creating a new file using open(..., O_CREAT), it is an error
to fail to specify a creation mode. If a mode is not specified, a
random stack provided value is used as the "mode".
This will become a compile error in a future Android change.
Change-Id: I36a3d67d294a915c1f79632a1b0ba45edd1214b1
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Change-Id: I29fb3ee5664c1f0ee0409c1bb2be087ecca637db
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Change-Id: I9e1b918b2635d961604a4a9d88eb1c7179a167a7
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Change-Id: I12ef9367d05dbe069c037b1b4acd6347a8cf3ece
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Change-Id: I021ddcc1bcb63132a4597d13e3d09db2a5f2c628
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Change-Id: Ib86f1af80a0d2833535a0ad170dce5ac061eec75
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buffers" into jb-dev
* commit 'e6cb412b677966c25b424766dc6ceb2787d49caa':
SurfaceMediaSource: keep refs to current buffers
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jb-dev
* commit '33e28dd3c033641f4db9d1b99c7c88d6d08c61f7':
SurfaceMediaSource: keep refs to current buffers
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This change fixes a bug in SurfaceMediaSource where it would not keep a
reference to all of the Gralloc buffers that the video encoder is currently
using.
Bug: 6655597
Change-Id: Ifd99976cc7ae57ed724bbf2bbcff357a39497391
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This change updates all the uses of ANativeWindow to use the new ANW functions
that accept and return Sync HAL fence file descriptors.
Change-Id: Id7db42d8d6380f8b440d88476ce9211c6225fb16
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Bug: 6682991
Change-Id: I8b0d91ab76b700ab981f2524124109234023a9da
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decoders back to 4.
* commit 'e77c76dc57f88ec9e6b370dcba8a8c3e0684461d':
Revert input buffer count for the SoftAAC* decoders back to 4.
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* commit '39a3113539f9bcc86a8efe562780c23ac8731098':
Revert input buffer count for the SoftAAC* decoders back to 4.
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This improves the netflix bandwidth change responsiveness.
With the old count of 32 the audio decoder running at a higher
priority would starve the video decoder for about 1 sec while
draining the input data.
I haven't been able to break any stream playback using MediaCodec
using the lower count, even though I'm sure I had a good reason
to increase it in the first place (see bug 6478823)
Change-Id: I7fd3e71794b0cc1570ea9d27d1d869ab1e1750c4
related-to-bug: 6685031
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dlopen libstagefright_chromium_http to allow it to be on the platform
side of the pdk without getting overwritten during a pdk fusion build.
Change-Id: I391e81d98beab165c3313bc8a71bb370ebb8584d
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* commit 'fc42e16d96b7e1b18b573f40df1319c56f8ab933':
Fix track selection code in AwesomePlayer
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Change-Id: I1c5a2f13f562a0ba253a93b4346cbe2c60099ddd
related-to-bug: 6671977
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* commit '7b4de10199f3a1803350e9a65baa671874d9c2f1':
stagefright: set scaling mode for blank frames
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This change sets the ANativeWindow scaling mode before pushing the blank frames
during decoder tear down.
Bug: 6603254
Change-Id: Ic64011645e2d3671b4a8d302ac7f39e6fd3affcd
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* commit '192cbbad773979a6fe3b5a0c223356de4fc3309c':
Fix audio track pause.
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AudioTrack::pause() should signal the control block condition
to release threads waiting for available buffers in obtainBuffer().
Otherwise the behavior relies on the timout on the condition
or the fact that audioflinger will mix a new audio buffer while executing
the pause.
Bug 6653769.
Change-Id: I5f8f73c471fe306070f30b814f32fd4b4dc1d575
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jb-dev
* commit '82233b4b5bee06f11618a558ae465abbfdedfd6e':
Better handle invalid duration in Xing/VBRI tags
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b/6654637
Change-Id: Ie92eddd84ef5914b77dbadaacd80b4de6cd66e1c
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* commit '49dd5cf3469cd755321f8ec10013ad4fcfaf723d':
Log track name on obtain/releaseBuffer warnings
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This should help diagnose problems by allowing us to correlate
the logs with the dumpsys media.audio_flinger output.
Change-Id: I8c7c592b4f87d13b0f29c66ce7a2f301a0f063c9
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/frameworks/av/" into jb-dev
* commit 'e20435de10f63af812da9d4f801b61a8817733a0':
Add NOTICE and MODULE_LICENSE_APACH2 to libs build under /frameworks/av/
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/frameworks/av/" into jb-dev
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Change-Id: I0a3af3e2abdedebd5934f3d941d01c32cfc75e26
related-to-bug: 6647465
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into jb-dev
* commit 'cac467b3af9235229522a134d2c57e4fa605ba2f':
Fix one of the potential ANR issues from AwesomePlayer
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Change-Id: I8ac7bbb5fe219f65bdbdd223538fbac0575e2957
related-to-bug: 6565826
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jb-dev
* commit 'b4ea1ab821d652cb080910d4081f5f1318597a90':
stagefright: fix AudioRecord callback buffer size
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Make sure that the maximum number of frames passed to
AudioSource by the AudioRecord callback always fits within
the maximum buffer size defined by kMaxBufferSize.
Also make sure that the total AudioRecord buffer size is more
than the minimum required.
Change-Id: I26a1f998e0cf75ac88b02e67ec9d8db3c0cca193
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libstagefright_soft_h264dec and libstagefright_soft_vpxdec
* commit '958f463c4cddba6437c86d5bfef68167f767ee66':
Add NOTICE and MODULE_LICENSE_APACHE2 for libstagefright_soft_h264dec and libstagefright_soft_vpxdec
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libstagefright_soft_vpxdec
Change-Id: Id4f1ef8dfc21731846a9862b8d12940a92fe99bb
related-to-bug: 6647465
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* commit '10cf121f7285e23c37264dab3bad7cffefd754b1':
Update make file
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