The previous code was calling stat() on the parent directory rather than the actual file.
Change-Id: If64552cb37552c77618a81ae4333307a018efe13
Signed-off-by: Mike Lockwood <lockwood@android.com>
Added AudioEffect C++ class. AudioEffect is the base class for effect specific implementations,
OpenSL ES effect interfaces and audio effect JNI.
Added the AudioEffect JNI and AudioEffect JAVA class. AudioEffect is the base class
to implement more specific JAVA classes to control audio effects from JAVA applications.
Change-Id: If300a1b708f2e6605891261e67bfb4f8330a4624
First drop of audio framework modifications for audio effects support.
- AudioTrack/AudioRecord:
Added support for auxiliary effects in AudioTrack
Added support for audio sessions
Fixed left right channel inversion in setVolume()
- IAudioFlinger:
Added interface methods for effect enumeraiton and instantiation
Added support for audio sessions.
- IAudioTrack:
Added method to attach auxiliary effect.
- AudioFlinger
Created new classes to control effect engines in effect library and manage effect connections to tracks or
output mix:
EffectModule: wrapper object controlling the effect engine implementation in the effect library. There
is one EffectModule per instance of an effect in a given audio session
EffectChain: group of effects associated to one audio session. There is one EffectChain per audio session.
EffectChain for session 0 is for output mix effects, other chains are attached to audio tracks
with same session ID. Each chain contains a variable number of EffectModules
EffectHandle: implements the IEffect interface. There is one EffectHandle object for each application
controlling (or using) an effect module. THe EffectModule maintians a list of EffectHandles.
Added support for effect modules and effect chains creation in PlaybackThread.
modified mixer thread loop to allow track volume control by effect modules and call effect processing.
-AudioMixer
Each track now specifies its output buffer used by mixer for accumulation
Modified mixer process functions to process tracks by groups of tracks with same buffer
Modified track process functions to support accumulation to auxiliary channel
Change-Id: I26d5f7c9e070a89bdd383e1a659f8b7ca150379c
The problem is that when an input stream is opened for record over bluetooth SCO, the kernel
mono audio device should be opened in RW mode to allow further use of this same device by an output stream
also routed to bluetooth SCO.
This does not happen because of a bug in AudioSystem::isBluetoothScoDevice() that does not return true
when the device is DEVICE_IN_BLUETOOTH_SCO_HEADSET (input device for blurtooth SCO).
Change-Id: I9100e972931d8142295c7d64ec06e31304407586
Added IEffect and IEffectClient binder interfaces to exchange effect module control
and status information between application and media server processes.
Change-Id: I10e8e894898e52ed9956a765d0ef7075eb2593af
The problem is due to a too big difference between the buffer size used at the hardware interface and at the A2DP interface.
When no resampling occurs we don't notice problems but the timing is very tight. As soon as resampling is activated, the AudioTrack underruns.
This is because the AudioTrack buffers are not resized when moving the AudioTrack from hardware to A2DP output.
The AudioTrack buffers are calculated based on a hardware output buffer size of 3072 bytes. Which is much less than the A2DP output buffer size (10240).
The solution consists in creating new tracks with new buffers in AudioFlinger when the A2DP output is opened
instead of just transfering active tracks from hardware output mixer thread to the new A2DP output mixer thread.
To avoid synchronization issues between mixer threads and client processes, this is done by invalidating tracks
by setting a flag in their control block and having AudioTrack release the handle on this track (IAudioTrack)
and create a new IAudioTrack when this flag is detected next time obtainBuffer() or start() is executed.
AudioFlinger modifications:
- invalidate the tracks when setStreamOutput() is called
- make sure that notifications of output opening/closing and change of stream type to output mapping are sent synchronously to client process.
This is necessary so that AudioSystem has the new stream to output mapping when the AudioTrack detects the invalidate flag in the client process.
Previously their were sent when the corresponding thread loop was executed.
AudioTrack modifications:
- move frame count calculation and verification from set() to createTrack() so that is is updated every time a new IAudioTrack is created.
- detect track invalidate flag in obtainBuffer() and start() and create a new IAudioTrack.
AudioTrackShared modifications
- group all flags (out, flowControlFlag, forceReady...) into a single bit filed to save space.
Change-Id: I9ac26b6192230627d35084e1449640caaf7d56ee
this is used in a few places to get access to the android.view.Surface
native surface. use a macro instead. Also rename the field to mNativeSurface.
Change-Id: I1c6dea14abd6b8b1392c7f97b304115999355094
MediaMetadataRetriever uses a single static lock for all operations.
This effectively serializes all metadata retrieval operations in a
single process. This patch uses the object level lock for all normal
operations and only uses the static lock to serialize calls to
release.
Change-Id: I81c9f234c2f0007a26d18e1398c709b41a4dbbd7
Condition must be initialized with SHARED for the old behavior, where
they can be used accross processes.
Updated the two places android that require SHARED conditions.
PRIVATE conditions (and mutexes) use more efficient syscalls.
Change-Id: I9a281a4b88206e92ac559c66554e886b9c62db3a
- AudioPolicyManager: allow platform specific choice for opening a direct output.
Also fixed problems in direct output management.
- AudioFliinger: use shorter standby delay and track inactivity grace period for direct output
thread to free hardware resources as soon as possible.
- AudioSystem: do not use cached output selection in getOutput() when a direct output
can be selected.
Change-Id: If44b50d29237b8402ffd7a5ba1dc43c56f903e9b
Previous range-checking fix removed an inequality check. This change
restores it.
Offending change was I5eb310ced58c3c64a7af2d11b80326efe5adbcab
Change-Id: Ic952c3ba5a4f7e5ab2148ec623b6f083cb7495fb
Previously invoke would work only after prepare but not
when the player is in play or pause state (for instance).
This new change just check that the player has been initialized
and is not in the error state.
Bug:2488931
Change-Id: I9b9f3679593a3b7697c1a84d993fdcd7e1693a90
Previously invoke would work only after prepare but not
when the player is in play or pause state (for instance).
This new change just check that the player has been initialized
and is not in the error state.
Bug:2488931
Change-Id: I7a69d1b6e3eec1e5dbdf7378ff2085329062595a
The problem is that AudioRecord never exits read() when a timeout occurs while trying
to get new PCM data from audio hardware input buffer: it just keeps waiting and retrying until stop() is called.
In the same time, opencore AndroidAudioInput::audin_thread_func() loop cannot be exited when stuck
in AudioRecord::read() because the iExitAudioThread flag can only be sampled when AudioRecord::read()
returns. We remain stuck with the audio input thread running.
The fix consists in modifying AudioRecord behavior in case of timeout when getting new PCM samples.
We now wait only one timeout period and try to restart audio record, in case the problem is due to a media_server
process crash. If this fails, we exit read() with a number of bytes read equals to 0 so that
AndroidAudioInput::audin_thread_func() loop can exit.
Also modified Audioflinger::RecordThread() loop so that we attempt to recover from HAL read errors.
In case of read error, the input stream is forced to standby so that next read attempt does a
reconfiguration and restart of the audio input device.
an error. This makes 'playback complete' essentially equivalent to
being paused at the end, and treats it the same as being paused at
any other position.
- I decided to completely remove jpeg decoding related stuff from this change
I think that setting is better off if it is specified by the system properties.
We don't have to include MediaProfiles.h header in skia files
This change fixes a problem where an unwanted tone is generated by audio policy manager when a MT call is answered.
This is because of a policy that replaces high visibility system sounds (ringtones, alarms...) by a beep when in call.
There is a transitory phase while the call is being answered where the phone state is changed to IN_CALL but the
ringtone is still playing. The audio policy manager then mutes the end of the ringtone and starts playing a beep
in replacement because the ringtone is categorized as high visibility.
The fix consists in changing the ringtone stream type from high visibility to low visibility. This is not a problem as
the only actual use case where a ringtone would be generated while in call is if another call is received.
But in this case, the phone system does not generate a ringtone but a call waiting tone instead.
It is therefore not required to handle a ringtone as a high visibiltiy tone that must be somehow signaled to the user
while in call.
Make sure we don't have an empty string before checking if it's a
directory since this string is tainted.
Change-Id: I5eb310ced58c3c64a7af2d11b80326efe5adbcab
At some point the implementation became complicated because of
SurfaceFlinger's special needs, since we are now relying on gralloc
we can go back to much simpler MemoryDealer.
Removed HeapInterface and AllocatorInterface, since those don't need
to be paramterized anymore. Merged SimpleMemory and Allocation.
Made SimplisticAllocator non virtual.
Removed MemoryDealer flags (READ_ONLY, PAGE_ALIGNED)
Removed a lot of unneeded code.
Added getRenderPosition() API to IAudioFlinger to retreive number of audio frames
written by AudioFlinger to audio HAL and by DSP to DAC.
Added getRenderPosition() API to AudioHardwareInterface to retreive number of audio frames
written by DSP to DAC.
Exposed AudioTrack::getPosition() to AudioSink() to make it available to media player.
Removed excessive log in AudioHardwareGeneric.
Modified AudioService.getActiveStreamType() so that STREAM_VOICE_CALL is selected when a track using this stream
type is playing.
Chanded isMusicActive() for a more generic isStreamActive(stream) method in AudioSystem, IAudioFlinger and AudioFlinger.
The problem is that after a media_server crash, the value of the A2DP output handle can change.
As this value is cached in AudioSystem for all client processes there can be a mismatch between the cached
and actual value after a media_server restart.
The fix consists in clearing the cached output handles and output to stream map values cached
in AudioSystem in AudioFlingerClient::binderDied() which is called when the media_server crashes.
commit 144b1c40e9cf08a584c50e1bef7ba3f287e81a4f
Author: Andreas Huber <andih@google.com>
Date: Wed Dec 16 09:28:23 2009 -0800
This H264 file shows a certain problem even better.
commit 3245f1f3b7471975aeeb824a756c987abd610f55
Author: Andreas Huber <andih@google.com>
Date: Wed Dec 16 09:20:08 2009 -0800
Using only the QA testfiles now.
commit 074817eb3816c5dd70858a3594e3b92d799d873b
Author: Andreas Huber <andih@google.com>
Date: Tue Dec 15 16:17:39 2009 -0800
Yay, roles are back again now that the API is in place.
commit 6d847e4932cc38301ae27cb7283b7f1553a95457
Author: Andreas Huber <andih@google.com>
Date: Tue Dec 15 13:01:20 2009 -0800
Added commandline option for specifying the random seed for reproducable tests.
commit 62ab37b26336eaa67e49791c41c996acb6acee3f
Author: Andreas Huber <andih@google.com>
Date: Mon Dec 14 10:53:27 2009 -0800
When issuing a seek it is important that only the first MediaSource::read call has the seek option.
commit e77c46644b2fb6862bafa3569f7d304252074f1e
Author: Andreas Huber <andih@google.com>
Date: Mon Dec 7 16:39:07 2009 -0800
Make sure the tests are actually built, sp<OMXCodec> becomes sp<MediaSource>
commit 6df56915bd55a9445b3c6f953d3cc251d81579b8
Author: Andreas Huber <andih@google.com>
Date: Thu Dec 3 14:25:36 2009 -0800
Temporarily disable support for querying the roles of OMX components.
commit 31bb26930df9e3658dea684cedb4b0f1a06a4a88
Author: Andreas Huber <andih@google.com>
Date: Tue Dec 1 13:36:52 2009 -0800
Disregard EOS events, slightly change the way the EOS flag on output buffers is handled.
commit 4c382fbc9aebee8197d5988d04378062809e7c48
Author: Andreas Huber <andih@google.com>
Date: Tue Dec 1 09:37:24 2009 -0800
New random seek test for the codec tests. Fixed "sticky" end-of-output-buffers flag behaviour in OMXCodec.
commit c762eac3e44309592b61a168d66e091cf609fa03
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 3 14:13:43 2009 -0800
Fix a typo.
commit 50540a59b65c7d476b0193c7494cd75895e6ca6d
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 3 09:48:35 2009 -0800
Some more fine tuning of the unit tests, make MPEG4Extractor less verbose.
commit 1157a7e52a0636706caa235abe16d2ff8a0b8140
Author: Andreas Huber <andih@google.com>
Date: Wed Oct 28 12:01:01 2009 -0700
Changes to the IOMX::listNodes API, this now returns the component's roles as well, unit tests now test all components in all supported roles by default.
commit 30fbf2d8c6cb927689f7ba75eb550a81e9df488a
Author: Andreas Huber <andih@google.com>
Date: Mon Oct 26 09:45:26 2009 -0700
Initial check-in of unit tests for OMX components.
Merge commit '6d42d80653f2c41f3e72a878a1d9a6f9693b89f7' into eclair-mr2
* commit '6d42d80653f2c41f3e72a878a1d9a6f9693b89f7':
Fix issue 2304669: VoiceIME: starting and canceling voice IME yields persistent "error 8" state on future attempts and breaks voice search.
Fixed AudioFlinger::openInput() broken in change ddb78e7753be03937ad57ce7c3c842c52bdad65e
so that an invalid IO handle (0) is returned in case of failure.
Applied the same correction to openOutput().
Modified RecordThread start procedure so that a failure occuring during the first read from audio input stream is detected and causes
the record start to fail.
Modified RecordThread stop procedure to make sure that audio input stream fd is closed before we exit the stop function.
Fixed AudioRecord JAVA and JNI implementation to take status of native AudioRecord::start() into account
and not change mRecordingState to RECORDSTATE_RECORDING if start fails.
