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But still restricted to 16-bit PCM currently
Change-Id: I5df0e5033da9144ca73e44addf14a63d31406034
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Change-Id: I8721ecedfb429c4e233453d1e768ddf69ecabbe4
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Change-Id: Ife23b88474c1d62c0cf682c1a310d951f2c0f54a
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Change-Id: Id8aac1077c3de1bb1f58bfcfcca93d685abe9b79
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Change-Id: Iee41641252f65cea9ce0ea0ec1ea4229608f7c64
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Supports multiple clients both at native sample rate and with resampling.
Change-Id: Icea55b4fd30751761b7debaa3ce016c79e712d8d
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This removes a circular dependency between libmedia and libnbaio
Change-Id: I3692cbbe2b76028a953b36606b62cdda70c8c26f
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Change-Id: I58ff399ace74965ebf8e5690c20637385f9425ad
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so they can be properly freed.
Change-Id: I6f389035bc29e74e7c367c1c6d0252b180f666b3
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Change-Id: I7b1cc71057b2bd4f771e7bcf508a8c3abd6017ce
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Contains the necessary changes to make frameworks/av build and work
on a 64-bit machine.
Signed-off-by: Craig Barber <craig.barber@arm.com>
Signed-off-by: Kévin PETIT <kevin.petit@arm.com>
Signed-off-by: Ashok Bhat <ashok.bhat@arm.com>
Signed-off-by: Marcus Oakland <marcus.oakland@arm.com>
Change-Id: I725feaae50ed8eee25ca2c947cf15aee1f395c43
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Change-Id: I1cd3432c02db1600e49ade67e6a89e39e7a8d3ed
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if resampling is required"
* commit '4e12dadeec10993d2f77cce96600e73ca0dd86e2':
AudioTrack: Never try to use the fast path if resampling is required
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Unless AudioFlinger was built with FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
enabled, AudioFlinger would deny using the fast path (and internally
fall back to the normal codepath) when it realized that resampling
was required. Since the buffer size calculations within AudioFlinger
don't take resampling into account properly (see the calculation
below "AUDIO_OUTPUT_FLAG_FAST denied" in audioflinger/Threads.cpp,
just below the hunk that this patch changes), make sure AudioTrack
doesn't try to use the fast path if resampling is required.
This removes the possibility to enable
FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE in AudioFlinger since it
AudioTrack now won't even try to use the fast path for content
that requires resampling, regardless of the AudioFlinger configuration.
Change-Id: Icf0f8ad50bf0fdb84657f518c0120aa0535f23f9
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Change-Id: I1c611d1037685d52ccc84efe0fccd6413ec938e9
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Change-Id: I27dc108e2c1f7ffd414bb7ff3d4c349651da6c26
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Change-Id: I9add0872030a26933f6b6dad1be299154611cc56
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since we started to use java's HTTPConnection instead of the native
implementation. Also remove other remnants of the previous http implementation,
such as accounting for the http user's uid.
Change-Id: I60bfd31381ea40d2220db587ec5c433093b60034
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Change-Id: I62d6534a9581e84ae20c2422f7ad9aeda9b7c4df
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available to media code"
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to media code
Change-Id: I9f74a86e70422187c9cf0ca1318a29019700192d
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Change-Id: I665ba3358dd9502f0adec70d486e7bf8a2e1b0fe
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Eventually we may want to use uint64_t, but will need to confirm atomicity.
Bug: 12381724
Change-Id: Ia2c591d262d22b47b6f7dab4b9d9faa14b86d865
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AudioTrack::obtainBuffer() passes a pointer to a timeout
variable that has gone out of scope when calling
ClientProxy::obtainBuffer().
Same fix for AudioRecord.
Bug: 11968591.
Change-Id: I22af8f94fa2f8cc54ab5c25c89167c805e754317
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MemoryLeakTrackUtil"
* commit '9f875b2717b683cc61e3fa4ba24b3693fab5b46a':
Do partial reads in MemoryLeakTrackUtil
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Do partial read in MemoryLeakTrackUtil dumpMemoryAddresses
to avoid using more memory than what is allocated.
Change-Id: I94feb4e00647407f938571167b981c7371f39e3d
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When restoring an AudioTrack, the next position callback point
should not be modified and set ahead of current buffer head.
Otherwise, as frames are dropped, the new position is never reached
and an application relying on position callbacks to reload the buffer
would be stalled.
Bug: 11868603.
Change-Id: I93b2a311642a0c89944b78bcc0482d4ceed98ae4
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AudioPlayer must read the sampling rate from offloaded audio sinks
whenever a new time position is computed as the decoder can update
the sampling rate on the fly.
Change-Id: I997e5248cfd4017aeceb4e11689324ded2a5bc88
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AudioRecord::openRecord_l() code was refactored earlier to
remove the redundant parameters:
> Change-Id: I124dce344b1d11c2dd66ca5e2c9aec0c52c230e2
This changelist refactors AudioTrack similarly.
Change-Id: Iefd2bd662870ea81d04eff7b7c26f9c8b0dadd26
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The AudioRecord input handle code was refactored earlier
to fix a potential handle leak, and to simplify the code:
> Change-Id: I124dce344b1d11c2dd66ca5e2c9aec0c52c230e2
This changelist refactors AudioTrack similarly,
and adds further cleanup of both AudioTrack and AudioRecord.
We attempt to implement the rules for referencing counting I/O handles,
but there is still the possibility of a handle leak if the client process
dies after allocating the handle reference but before releasing it.
That issue is being tracked separately.
Details:
- AudioSystem::getOutput() is now called within createTrack_l
- restoreTrack_l was missing offload info
now it has the info available,
but is not yet being called for offloaded tracks
- AudioTrack::getOutput() is now const
- Remove getOutput_l()
Change-Id: I44a0a623d24fc5847bcac0939c276400568adbca
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in IAudioFlinger::createTrack and IAudioFlinger::openRecord
Change-Id: I09c644c80e92c8e744b1b99055988a2588b2a83d
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This makes it the same as for IAudioTrack.
Previously we had a mix of int, int32_t and uint32_t,
which could cause comparisons to fail.
Change-Id: Ic72ef650ee81f65db11e42ed16d818fdf3e310f4
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Change-Id: I515970aa0660418d5d1640fb1bf477e112c89bdd
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earlier"
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This will allow removal of most of the parameter list to createTrack_l.
Also check for valid stream type at client so we can log a better error
message.
Change-Id: Ia7176896d47fbb49106119fca26d9de8e7efe859
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