The visualizer enables application to retrieve part of the currently playing audio for visualization purpose.
It is not an audio recording interface and only returns partial and low quality audio content as a waveform or
a frequency representation (FFT).
Removed temporary hack made in MediaPlayer for animated wall papers based on audio visualization (snoop() method.
This commit also includes a change in AudioEffect class:
- the enable()/disable() methods have been replaced bya more standard setEnabled() method.
- some fixes in javadoc
Change-Id: Id092a1340e9e38dae68646ade7be054e3a36980e
- An alternative would be to define a common base class that
both MediaRecorderClient and MediaPlayerClient can derive.
But since the common code, onTransact() and notify() uses
some Binder code, having a common base class may not gain
us too much in terms of code reuse.
Change-Id: Ibc06720278ad173fceacff3d267b7060856c6316
Effect API:
- Use different definitions for audio device, channels, formats... in AudioSystem and EffectApi:
Removed media/AudioCommon.h file created for initial version of EffectApi
- Indicate audio session and output ID to effect library when calling EffectCreate(). Session ID can be useful to optimize
the implementation of effect chains in the same audio session. Output ID can be used for effects implemented in audio hardware.
- Renamed EffectQueryNext() function to EffectQueryEffect() and changed operating mode:
now an index is passed for the queried effect instead of implicitly querying the next one.
- Added CPU load and memory usage indication in effects descriptor
- Added flags and commands to indicate changes in audio mode (ring tone, in call...) to effect engine
- Added flag to indicate hardware accelerated effect implementation.
- Renamed EffectFactoryApi.h to EffectsFactoryApi.h for consistency with EffectsFactory.c/h
Effect libraries:
- Reflected changes in Effect API
- Several fixes in reverb implementation
- Added build option TEST_EFFECT_LIBRARIES in makefile to prepare integration of actual effect library.
- Replaced pointer by integer identifier for library handle returned by effects factory
Audio effect framework:
- Added support for audio session -1 in preparation of output stage effects configuration.
- Reflected changes in Effect API
- Removed volume ramp up/down when effect is inserted/removed: this has to be taken care of by effect engines.
- Added some overflow verification on indexes used for deferred parameter updates via shared memory
- Added hardcoded CPU and memory limit check when creating a new effect instance
Change-Id: I43fee5182ee201384ea3479af6d0acb95092901d
Audio sessions are used to associate audio effects to particular instances (or groups) of MediaPlayers or AudioTracks.
Change-Id: Ib94eec43241cfcb416590f435ddce7ab39a07640
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
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.
