- 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
Part 1 of the fix: when the user doesn't elect to use the car dock
for music and media, the APM was not aware of the device being
docked.
This is fixed by dissociating the notification for the APM of
the docking to the dock from the sink state change of the A2DP
device.
Also missing was forcing the volumes to be reevaluated whenever
the device is docked or undocked, as volumes for docks may
differ, even when the same output device is being used.
Change-Id: If5314e27821a71adbd6df6fdf887c45208241d96
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.
This is not a real fix for the issue but a change to make sure that the behavior is consistent regardless of
external condidions (WIFI ON or OFF, music started before call or not, A2DP device same as SCO device...).
As there is now way to guaranty good quality audio over both SCO and A2DP simultaneously, especially when WIFI is on, We will stick to this behavior:
When music is playing and we are docked to the desk dock and a call is answered with a BT SCO headset, A2DP output will be suspended.
If music is restarted during the call, it will appear muted to the user until the call is terminated.
The noise is the residual ring tone that is still playing while the call is answered and the
audio route changed to headset or earpiece.
The fix consists in muting the ringing tone when changing mode from ringtone to in call
and delaying the route change until the audio buffers are emptied.
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.
First implementations of audio policy manager in Eclair branch have shown that most code is common to all platforms.
Creating AudioPolicyManagerBase base class will improve code maintainability and readability.
Audio policy manager code for platforms using generic audio previously in AudioPolicyManagerGeneric is replaced by AudioPolicyManagerBase.
Audio policy manager test code previously in AudioPolicyManagerGeneric is moved to AudioPolicyManagerBase.
Also added a wake lock for delayed commands in AudioPolicyService.
Modified AudioFlinger duplicating output thread so that audio tracks are not mixed until both outputs (A2DP and hardware) have exited standby mode. This avoids to have one output far ahead of the other and audio frames dropped because the compensation mechanism cannot keep up.
Also calculate the maximum wait time in OutputTrack::write() based the on smallest frame count of all output threads instead of the frame count of the thread the OutputTrack is connected to. This avoids starving the thread with the smallest frame count by waiting too long on the other thread.
Since the frame count was reduced on hardware output to reduce latency the difference between A2DP and hardware outputs frame counts had become problematic.
Also increased the number of overflow buffers to cope with bigger timing differences among outputs.
The ToneGenerator failed to initialize because no more tracks were available in AudioFlinger mixer.
All tracks were used because the duplicating output was failing to free the tracks on audio hardware output mixer when exiting due to a misplaced test on output activity: output tracks where only freed if the duplicating output was active when exiting.
The fix consists in freeing the output tracks when the duplicating thread is destroyed without condition.
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.
Merge commit '1ac56b602aa6a1ac54c608e5a8b76f44638db23b' into eclair-mr2
* commit '1ac56b602aa6a1ac54c608e5a8b76f44638db23b':
Fix issue 2292062: Audio freezes for three seconds when choosing ringtones with a headset connected and music playing.
The problem comes from a deadlock with AudioPolicyService mutex: When the second ringtone starts,
this mutex is locked by AudioPolicyService::startOutput() which in turn calls setParameters() to change the output device.
Audioflinger::ThreadBase::setParameters() signals the parameter change to the AudioFlinger mixer thread and waits for a condition
indicating that the parameter change has been processed.
At the same time, the mixer thread detects that the audio track corresponding to the first ring tone has been killed and calls its destructor.
This calls AudioPolicyService::releaseOutput() which tries to lock the AudioPolicyService mutex.
If this happens before the mixer thread can process the setParameters() command we are deadlocked.
The deadlock ends because setParameters() uses a timeout when waiting for the condition.
This regression was introduced by change 33736 fixing issue 2265163.
The fix consists in calling AudioPolicyService::releaseOutput() from Track::destroy() instead of from Track destructor: as detroy() is never called from the mixer thread loop (as opposed to the destructor) the deadlock described above cannot occur.
This is a second attempt to fix the audio routed to earpiece syndrom.
The root cause identified this time is the crash of an application having an active AudioTrack playing on the VOICE_CALL stream type.
When this happens, the AudioTrack destructor is not called and the audio policy manager is not notified of the track stop.
Results a situation where the VOICE_CALL stream is considered as always in use by audio policy manager which makes that audio is routed to earpiece.
The fix consists in moving the track start/stop/close notification to audio policiy manager from AudioTrack to AudioFlinger Track objet.
The net result is that in the case of a client application crash, the AudioFlinger TrackHandle object (which implements the remote side of the IAudioTrack binder interface) destructor is called which in turn destroys the Track object and we can notify the audio policy manager of the track stop and removal.
The same modification is made for AudioRecord although no bug related to record has been reported yet.
Also fixed a potential problem if record stop is called while the record thread is exiting.
This change goes with a kernel driver change that reduces the audio buffer size from 4800 bytes (~27ms) to 3072 bytes (~17ms).
- The AudioFlinger modifcations in change 0bca68cfff161abbc992fec82dc7c88079dd1a36 have been removed: the short sleep period was counter productive when the AudioTrack is using the call back thread as it causes to many preemptions.
- AudioFlinger mixer thread now detects long standby exit time and in this case anticipates start by writing 0s as soon as a track is enabled even if not ready for mixing.
- AudioTrack::start() is modified to start call back thread before starting the IAudioTrack so that thread startup time is masked by IAudioTrack start and mixer thread wakeup time.
Reduce sleep time in AudioFlinger mixer thread when no data has been written to output to speed up startup time when exiting standby.
The rest of the modifications for this issues is in kernel driver:
commit 0dbb0ee136ed8de757df1ae26d84556c1751deae for buffer size modification from 8192 to 4800 bytes.
Another kernel improvement that is not submitted yes will reduce delay when audio output is exiting standby.
Added setVoiceVolume() method to AudioSystem, AudioFlinger, IAudioFlinger, AudioPolicyService.
Removed call to AudioHardwareInterface::setVoiceVolume() from AudioFlinger::setStreamVolume().
Added a workarouond to request the A2DP output standby directly to audio hardware when the sink is suspended as it seems that the suspend request often fails.
Also take into account resume requests received while a suspend request is pending.
This change is a complement to the main fix in kernel driver for the same issue (partner change #1250).
It removes clicks sometimes heard after the end of the tones while audio flinger is sending 0s to the audio output stream.
The problem was that the sleep time between two writes was more than the duration of one audio output stream buffer which could cause some underrun.
Also fixed a recent regression in ToneGenerator that made that the end of previous tone was repeated at the beginning of current one under certain timing circumstances when the maximum tone duration was specified.
Wait for the parameter set completed condition with a time out in ThreadBase::setParameters().
Also lock AudioFlinger mutex before accessing thread list in AudioFlinger::setParameters() and keep a strong reference
on the thread being used in case it is exited while processing the request.
There was a regression introduced in AudioFlinger by change 24114 for suspended output:
The suspended output was not reading and mixing audio tracks.
When the phone is ringing, the A2DP output is suspended if the SCO headset and A2DP headset are the same. As the ringtone is played over the duplicated output, the fact that the A2DP output was not reading data was causing the hardware output to be stalled from time to time.
The fix consists in locking AudioFlinger::mLock mutex in the TrackBase destructor before clearing the strong pointer to the shared memory client. The mutex is not locked in removeclient() any more which implies that we must make sure that the Client destructor is always called from the TrackBase destructor or that we hold the mLock mutex before calling deleting the Client.
The problem comes from the fact that when the duplicated output is closed after BT headset disconnection, the OUTPUT_CLOSED notification is not sent to AudioSystem. Then the mapping between notification stream and duplicated output cached in AudioSystem is not cleared and next time a notification is played, the duplicated output is selected and the createTrack() request is refused by AudioFlinger as the selected output doesn't exist.
The notification is ignored by AudioFlinger because when it is sent by the terminating playback thread, the thread has already been removed from the playback thread list.
The fix consists in sending the notification in closeOutput() and not when exiting the playback thread.
The same fix is applied to record threads.
This is due to a regression introduced by change 24114: when no audio tracks are ready for mixing, 0s are written to audio hardware. However this should only happen if tracks have already been mixed since the audio flinger thread woke up.
Also do not write 0s to audio hardware in direct output threads when audio format is not linear PCM.
AudioFlinger: verify that mCblk is not null before using it in Track and RecordTrack contructors.
IAudioFlinger: check result of remote transaction before reading IAudioTrack and IAudioRecord.
IAudioTrack and IAudioRecord: check result of remote transaction before reading IMemory.
Add a parameter to ToneGenerator.startTone() allowing the caller to specify the tone duration. This is used by the phone application to have a precise control on the DTMF tone duration which was not possible with the use of delayed messaged.
Also modified AudioFlinger output threads so that 0s are written to the audio output stream when no more tracks are ready to mix instead of just sleeping. This avoids an issue where the end of a previous DTMF tone could stay in audio hardware buffers and be played just before the beginning of the next DTMF tone.
Do not ramp volume if the first frame of a track is processed after the track was stopped.
In the case of very short sounds, the track stop request can be received by AudioFlinger just after the start request before the first frame is mixed by AudioMixer. In this case, the track is already in stopped state and initial volume is applied with a ramp for the first frame processed which should not be the case: initial volume change is always applied immediatelly.
In AudioFlinger::MixerThread::putTracks(), change the mFillingUpStatus flag to FS_FILLING for active tracks so that mute request is executed without ramping volume down when the track is moved from A2DP to hardware output.
Also modified AudioFlinger::setStreamOutput() so that the notification of the change is sent only once to AudioSystem.
Apparently the problem is caused by the fact that A2dpAudioStreamOut::standby() calls a2dp_stop() after the headset has been powered down.
The workaround consists in indicating to A2DP audio hardware that a close request is pending and that stanby() must be bypassed.
This is because the AudioFlinger duplicating thread is closed while the output tracks are still active. This cause the output tracks objects to be destroyed at a time where they can be in use by the destination output mixer.
The fix consists in adding the OutputTrack to the track list (mTracks) of its destination thread so that a strong reference is help during the mixer processed and the track is detroyed only when safe by destination thread.
Also added detection of problems when creating the output track (e.g. no more tracks in mixer). In this case the output track is not added to output track list of duplicating thread.
When changing the audio output stream sampling rate with setParameters() make sure that all tracks have a sampling rate less or equal to 2 times the new output sampling rate.
