IMediaPlayer:
new setMetadataFilter method so set a filter (2 lists of allowed and blocked metadata type)
serialized in a Parcel.
MediaPlayer.java/android_media_MediaPlayer.cpp/mediaplayer.cpp
new setMetadataFilter that passes the filter down to the MediaPlayerService's binder interface.
MediaPlayerService.cpp
The Client inner class holds the allowed and blocked metadata types.
These are in 2 vectors that get populated in the setMetadataFilter.
A new shourldDropMetadata method returns true if a type of metadata should be dropped according
to the filters.
The notify method in run the metadata update notifications thru the filter and possibly drop them.
The cause is very likely that the WaveGenerator *lpWaveGen returned by lpToneGen->mWaveGens.valueFor(lFrequency) just before calling lpWaveGen->getSamples(lpOut, lGenSmp, lWaveCmd) is invalid. The frequency lFrequency is not part of the frequencies in mWaveGens.
This can happen if a different tone is started while the callback function is active: The state is changed to TONE_RESTARTING and the call to prepareWave() at line 1226 will change the tone descriptor pointed to by mpToneDesc as well as the content of mWaveGens. However, mpToneDesc was cached in a local variable lpToneDesc when entering the callback and is not reloaded when exiting prepareWave(). This causes a mismatch between the tone frequencies listed in lpToneDesc and the frequencies present in mWaveGens.
This regression was introduced in change 973 when mpToneDesc was cached in a local variable.
Merge commit '88e209dcf8c2ebddda5c272f46d1bd5478bc639c'
* commit '88e209dcf8c2ebddda5c272f46d1bd5478bc639c':
Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR
Store sample rate on 32 bits instead of 16 bits in audio_track_cblk_t.
Removed sampleRate() methods from AudioTrack and AudioRecord: replaced by getSampleRate().
AudioTrack::setSampleRate() no returns a status.
MediaPlayer.java has 3 new methods:
* newRequest creates a Parcel that can be used to send data to the
native player using invoke.
* invoke issues synchronous calls to the native player using opaque
parcels for the request and reply.
IMediaPlayer.h has 1 new abstract method:
* invoke
The Midi and Vorbis players have a stub for these. So far only PV
makes use of that new feature.
To avoid any copy overhead, the JNI interface uses Parcel as a java
object (no serialization/copy happens at the JNI layer).
The remote interface token is inserted when the Parcel is constructed
in java. That way the parcel is already routable when it reaches
IMediaPlayer.cpp (proxy). No extra copy is needed there.
Merge commit '1bee98af6532eba3c6cda636eee890530c95a2a7'
* commit '1bee98af6532eba3c6cda636eee890530c95a2a7':
Fill in CDMA gaps and clean up ToneGenerator code
The android.hardware.ICamera string have not been changed (but is looks
like the camera service and client one should undergo the same procedure)
The implementation of the interface must provide a unique string for
it's interface name. Currently all these strings in the media framework
start with 'android.hardware' when it should really be 'android.media',
the interface token has nothing to do with hardware.
Merge commit 'b2df1699c996e62baa78877978cd2c5607ea4194'
* commit 'b2df1699c996e62baa78877978cd2c5607ea4194':
NJ-1409: (frameworks/base) Support new audio encoding types(AMR and AAC).
- make sure that all binder Bn classes define a ctor and dtor in their respective library.
This avoids duplication of the ctor/dtor in libraries where these objects are instantiated.
This is also cleaner, should we want these ctor/dtor to do something one day.
- same change as above for some Bp classes and various other non-binder classes
- moved the definition of CHECK_INTERFACE() in IInterface.h instead of having it everywhere.
- improved the CHECK_INTERFACE() macro so it calls a single method in Parcel, instead of inlining its code everywhere
- IBinder::getInterfaceDescriptor() now returns a "const String16&" instead of String16, which saves calls to String16 and ~String16
- implemented a cache for BpBinder::getInterfaceDescriptor(), since this does an IPC. HOWEVER, this method never seems to be called.
The cache makes BpBinder bigger, so we need to figure out if we need this method at all.
This change is the first part of a fix for issue 1846343, :
- Added new enum values for input sources in AudioRecord and MediaRecorder for voice uplink, downlink and uplink+downlink sources.
- renamed streamType to inputSource in all native functions handling audio record.
A second change is required in opencore author driver and android audio input to completely fix the issue.
Added new tone types for CDMA IS-95 specific tones.
Automatic selection between IS-95, CEPT and JAPAN version base on operator
country code for call supervisory tones.
Also improved tone generator capabilities:
- Each tone segment can now generate its own set of frequencies
- A tone does not have to be a succession of alternating ON/OFF segments
- The sequence repetition does not have to start from first segment
AudioTrack, AudioRecord:
- remove useless mAudioFlinger member of AudioTrack and AudioRecord.
- signal cblk.cv condition in stop() method to speed up stop completion.
- extend wait condition timeout in obtainBuffer() when waitCount is -1 to avoid waking up callback thread unnecessarily
AudioFlinger:
- remove some warnings in AudioFlinger.cpp.
- remove function AudioFlinger::MixerThread::removetrack_l() as its content is never executed.
- remove useless call to setMasterVolume in AudioFlinger::handleForcedSpeakerRoute().
- Offset VOICE_CALL stream volume to reflect actual volume that is never 0 in hardware (this fix has been made in the open source): 0.01 + v * 0.99.
AudioSystem.java:
- correct typo in comment
IAudioflinger, IAudioFlingerClient:
- make AudioFlinger binder interfaces used for callbacks ONEWAY.
AudioHardwareInterface:
- correct routeStrings[] table in AudioHardwareInteface.cpp
The problem comes from the fact that AudioSystem::getOutputFrameCount() calls getOutput() to retrieve the active output (A2DP or Hardware) before calling get_audio_flinger(). If it is the first time AudioSystem::getOutputFrameCount() is called in a given process, getOutput() will return a wrong value because gA2dpEnabled has not yet been updated by get_audio_flinger().
The fix consists in calling get_audio_flinger() in getOutput() to be sure that gA2dpEnabled is valid when getOutput() reads it.
Original author: elaurent
Merged from: //branches/cupcake/...
Automated import of CL 144054
