Modified default volume control logic in AudioService:
1 IN_CALL volume if in video/audio chat
2 NOTIFICATION if notification is playing or was playing less than 5s ago.
3 MUSIC
Modified silent mode:
- now also affect MUSIC stream type
- entering silent mode when VOL- hard key is pressed once while selected
stream volume is already at 0 (except for VOICE_CALL stream).
- exiting silent mode when pressing VOL+ hard key while in silent mode
Play sound FX (audible selections, keyboard clicks) at a fixed volume.
Modified audio framework:
- isStreamActive() method now implemented in AudioPolicyManagerBase (previously AudioFlinger)
- iStreamActive() now specifies a time window during which the stream is considered
active after it actually stopped.
Change-Id: I7e5a0724099450b9fc90825224180ac97322785f
Cherripick from master CL 79833, 79417, 78864, 80332, 87500
Add new audio mode and recording source for audio communications
other than telelphony.
The audio mode MODE_IN_CALL signals the system the device a phone
call is currently underway. There was no way for audio video
chat or VoIP applications to signal a call is underway, but not
using the telephony resources. This change introduces a new mode
to address this. Changes in other parts of the system (java
and native) are required to take this new mode into account.
The generic AudioPolicyManager is updated to not use its phone
state variable directly, but to use two new convenience methods,
isInCall() and isStateInCall(int) instead.
Add a recording source used to designate a recording stream for
voice communications such as VoIP.
Update the platform-independent audio policy manager to pass the
nature of the audio recording source to the audio policy client
interface through the AudioPolicyClientInterface::setParameters()
method.
SIP calls should set the audio mode to MODE_IN_COMMUNICATION,
Audio mode MODE_IN_CALL is reserved for telephony.
SIP: Enable built-in echo canceler if available.
1. Always initialize AudioRecord with VOICE_COMMUNICATION.
2. If echo canceler is available, disable our echo suppressor.
Note that this CL is intentionally not correcting the
getAudioSourceMax() return value in MediaRecorder.java as the
new source is hidden here.
Change-Id: Ie68cd03c50553101aa2ad838fe9459b2cf151bc8
EffectModule::process() was copying effect chain input buffer to
output buffer if no effect was active instead of accumulating it.
Change-Id: If4ca75601ea69a088d0f71d88aec53e90a1dec89
EffectModule::process() was copying effect chain input buffer to
output buffer if no effect was active instead of accumulating it.
Change-Id: I2838af2e7b6654d0a76547625929a5453da68d02
The problem is likely that one method is called on the AudioPolicyManagerBase
instance while it is still being constructed by AudioPolicyService.
To avoid this, the AudioPolicyService mutex is held by the constructor until the
platform specific AudioPolicyManager is constructed and the member
mpPolicyManager initialized.
Also added an initCheck() method to AudioPolicyInterface to verify successful
initialization of AudioPolicyManager.
A similar change is done in AudioFlinger constructor.
Also added some missing protections in AudioFlinger methods where the
playback thread list is parsed.
Change-Id: I006b244ec057e1bb0aa5ebe426ef006e3b171056
The fix consists in selecting the digital audio device (SPDIF/HDMI)
when available if the routing strategy is STRATEGY_PHONE.
Change-Id: Ie500ae92f5c01f2511988543852ba559c6e5994b
The problem is that when the A2DP headset is disconnected, there is a transition
period during which the A2DP sink pumps data at a very high pace.
This makes that:
1 the audio flinger mixer thread spins and starves binder threads thus delaying
the completion of the A2DP output stream shutdown
2 we read the audio http audio stream faster than normal and we reach the end of stream
for audio while video is still playing if the streamed file is small enough.
The fix consists in detecting abnormal short write intervals and sleep to restore
a normal write pace.
Change-Id: Iab127882494ab0e26266371dc0ce5c2ff6fa476e
The audio routing policy when speakerphone is on and a dock with built-in
speakers is connected should be to output audio to teh dock speakers
Also removed route to SCO car kit if forced usage is not SCO as the SCO
socket might not be established.
Change-Id: I1aa2954092e28de935304b90f7a7a64d661934c7
HDMI device should have a higher priority than analog dock audio but a lower priority
than wired headsets.
Also modified AudioService so that HDMI is mapped to DEVICE_OUT_AUX_DIGITAL device and not
DEVICE_OUT_DGTL_DOCK_HEADSET as before to enable discrimination between SPDIF going to
digital dock and SPIDF going to HDMI.
Change-Id: I887d0c73479784dd2edaf41ce1a7d8d0bdcbb4bd
The problem is that the code in AudioPolicyManagerBase::checkAndSetVolume() that forces
voice volume to max when setting bluetooth SCO volume is not called if the bluetooth stream
volume did not actually change. So even if we re apply volumes when switching to bluetooth
device, the volume voice volume is not changed and remains what it was when routed to earpiece
What makes things worse on Passion is that stream volumes are limited when connected to bluetooth
and their actual value does not change as soon as they exceed the limit threshold.
Change-Id: Id7c317db45b392a1c20dca2859678e3c64a371ed
There is a bug in the way audio policy manager handles A2DP interface suspend/restore
when SCO is used. This bug is not new but has been triggered by a change in the timing
of the events received by audio policy manager when a call is setup and torn down
introduced by commit 164a8f86c7e48992691368c4895709c3bdb835a4.
The fix consists in grouping the control of A2DP suspended state in a single function
that is called systematically when conditions affecting this state are changed:
- call state change
- device connection/disconnection
- change in forced usage.
Change-Id: I46ee2399ee5547b60511fc6cfd32e2720091b0f8
Use a Mutex wherever atomic operations were used in AudioTrack,
AudioRecord, AudioFlinger and AudioEffect classes.
Change-Id: I6f55b2cabdcd93d64ef19446735b8f33720f8dbc
The audio mode MODE_IN_CALL signals the system the device a phone
call is currently underway. There was no way for audio video
chat or VoIP applications to signal a call is underway, but not
using the telephony resources. This change introduces a new mode
to address this. Changes in other parts of the system (java
and native) are required to take this new mode into account.
The generic AudioPolicyManager is updated to not use its phone
state variable directly, but to use two new convenience methods,
isInCall() and isStateInCall(int) instead.
Change-Id: Id744cd26520ea1d1a4795eabe6a1f0c58789af76
Update the platform-independent audio policy manager to pass the
nature of the audio recording source to the audio policy client
interface through the AudioPolicyClientInterface::setParameters()
method.
Change-Id: I6b4fd0f8a3acea0d7d30bbad98edd1977dc012bf
Add a recording source used to designate a recording stream for
voice communications such as VoIP.
Change-Id: I4091d67069b1a0170c1a5ca5e6acd51eb0aa08f9
If the pause request is received before the AudioTrack buffer was
completelly filled and the track ready for mixing, the pause is
not executed: the track just underruns and stays in pausing state.
The fix consists in considering the track ready for mixing immediately
if pausing.
Change-Id: Ia6cb4703fee2126e41011a6400ea8eeb3a3e5456
This issue showed that when an AudioTrack underruns during a too long period
of time and is therefore disabled by audioflinger mixer, it takes an additional
delay of up to 3 seconds to recover.
This fix adds a simple mechanism to recover immediately when the client application
is ready to write data again in the AudioTrack buffer
Also throttle warnings on record overflows
Change-Id: I8b2c71578dd134b9e60a15ee4d91b70f3799cb3d
Fixed the following issues in LVM effect bundle wrapper:
- memory leaks in EffectCreate() in case effect creation fails at various stages
- Added saturation when accumulating to output buffer
- Fixed problems with enabled effects count when an effect is released while enabled
- Do not allocate temporary buffer for accumulation each time process() is called
Fixed the following issues in effects framework (AudioFlinger)
- Release effect synchronously in the library when deleted from effect chain
- Do not call the effect process function if no tracks are present in the same
audio session
Change-Id: Ifbd80a163415cfb3c0a337c12082853ea45d9c91
Removed a cross deadlock condition between audioflinger and audio policy
service mutexes.
Audioflinger::createEffect() locks audioflinger mutex and then calls
AudioSystem::getOutputForEffect() which ends up in
AudioPolicyService::getOutputForEffect() which locks audio policy service
mutex. If at the same time, the command thread in audio policy service is
processing a command(set volume, set route...), the mutex is locked and the
command will call one audioflinger method which in turn will attempt to
lock audioflinger mutex.
The fix consists in releasing audioflinger mutex before calling
getOutputForEffect().
Change-Id: Id44e7feb36e0a295731f6aa97cf32d022edd34d0
Applications creating an audio effect on the output mix must
have the MODIFY_AUDIO_SETTINGS permission.
Change-Id: I57d88533f91ad0d33680107d79abcec28f7263b5
- Use a constant input level to the reverb engine and implement volume control in the
insert reverb. This avoids the volume spikes when an effect that was inserted after
the reverb is disabled or removed.
- Fix clicks (one silent buffer) at the end of the reverb disable period.
- Modified volume management in audioflinger so that the volume ramp is also done by
the insert effect if present when the track is paused (avoids clicks).
- Increased room level for all presets.
Also fixed problems with output stage session (-1):
- effect bundle wrapper was not designed to support session -1
- the permission check in audioflinger for using session -1 failed due to a wrong usage of
getCallingPid()
Change-Id: Id1ff51327263364bf71d3f2668fa5cde4311d84f
The *pReplyData argument of the command() function was left unitialized by EffectHandle::command()
when command was EFFECT_CMD_ENABLE, EFFECT_CMD_DISABLE and EFFECT_CMD_SET_PARAM_COMMIT.
Change-Id: I91a19817ead2a8cfbdd8e2d77ca270c7ce9d5bd4
- Fixed click when re-enabling effect during the turn off phase:
make sure the effect states where effect is processed are the same
where volume control is delegated to effect.
- Fixed click when effect is deleted while still active: do not apply
volume ramp if an effect having volume control was just removed from the
effect chain.
Also fixed a crash when PCM dump is enabled in effect bundle wrapper.
Change-Id: Ib562f5cf75c69af75df0e862536262e2514493e4
The problem is that the audio policy manager does not handle the input devices
when forced use for telephony is changed.
The problem does not appear in a call over PSTN becasue only teh output devices drives the
routing of in call audio to/from the base band.
The fix consists in modifying AudioPolicyManagerBase::setForceUse() to check for active inputs
and update the input device if needed.
Change-Id: I0d36d1f5eef1cce527929180c29b025439902f10
Fixed regression introduced by change a54d7d3d7dd691334189aab20d23c65710092869 in audioflinger mixer thread:
When the output stream is suspended, the sleep time between two writes must match the actual duration
of one output stream buffer otherwise the playback rate is not respected.
Change-Id: Ic5bebe890290d1f44aeff9dd3c142d18e26fff2a
Fixed regression introduced by commit 2a6b80bc65c4782b5a7168b300e1dc5ec9f617ee:
master mute was not working if no effect chains were present on session 0.
Change-Id: I66d107e045d159cb94d29c7476fa1e12d92f2ae7
- Fixed constant inversions in AudioEffect.java
- Do not return error when enabling an already enabled effect
- Update cached effect state in native AudioEffect class when effect is enabled/disabled by command() method
- Remove click when restarting effect during disable sequence
- Fixed problem in master mute management when volume control is delegated to effect.
Change-Id: I6df4ce9fcc54fdc7345df858f639d20d802d6712
When all audio tracks have been disabled and the mixer is running idle before the output stream is placed in standby,
the mixer sometimes fails to write to the output stream on time to avoid underrun.
This is because the sleep period used to wait before the next write to output stream is too close to the actual buffer duration.
In fact this sleep time is not critical as if we write too early to the output stream, the kernel driver will wait for free buffers
from the audio DSP DMA and we will sleep anyways.
The fix consists in dividing the calculated wait period by 2 to increase the margin.
Change-Id: I5730887dc2ccce2a511bc858494a6f7da6b392a0
Merge commit '45dc4f82a00e52b12389b22a7cfbbee5609e8e28'
* commit '45dc4f82a00e52b12389b22a7cfbbee5609e8e28':
Allow creation of an audio effect on a session with no audio tracks.
This is necessary to allow creating and enabling an effect attached to a particular player
session before the playback is started. As a matter of fact, the implementation of the mediaplayer
does not create the AudioTrack before playback starts.
Change-Id: I1266e8885f9d756acc949303321aaac0fbf83e34
