Change volume attenuation curve to provide more attenuation at
low volume settings, and finer steps at high volume.
See bug entry for link to doc with curve values.
Change-Id: I750548b2161a4c550ef982ba793156e4518119e8
Add a delay before restoring output path when a notification ends so that
short sounds can be heard on proper device before the path is actualy switched.
Change-Id: I1d2dd8e7e28e15fbcab344256f88499b26297372
Change the device selection order as follows to enable easier use of
A2DP while the device is docked:
1 - wired Headset
2 - A2DP Headset
3 - SPDIF/HDMI
4 - Dock
Also do not limit notifications volume when on dock.
Change-Id: I55ea6bea9f2d9ff284b54023e541b2788d0f1eb8
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
The stream volume was handled the same way for all different stream,
the only potential difference between each of them being the number
of steps available to the user to change the volume. This was
mapped to 99 steps of 0.5dB amplitude, offering a maximum attenuation
of -49.5dB.
This change consists in defining for each stream a curve with two
knees (3 segments) for conversion from volume index to attenuation.
This curve is defined in the AudioPolicyManager in
initializeVolumeCurves(), and can therefore be overridden by the
platform.
Note that this change doesn't modify the volume curves: this CL
enables the curves to be changed by overriding this default
behavior.
Change-Id: I575b66799c52df2906db248943b15120b8a79ea2
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 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
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
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
Added methods for audio effects management by audio policy manager.
- control of total CPU load and memory used by effect engines
- selection of output stream for global effects
- added audio session id in parameter list for startOutput() and stopOutput().
this is not used in default audio policy manager implementation.
Modifications of audio effect framework in AudioFlinger to allow moving and reconfiguring
effect engines from one output mixer thread to another when audio tracks in the same session
are moved or when requested by audio policy manager.
Also fixed mutex deadlock problem with effect chains locks.
Change-Id: Ida43484b06e9b890d6b9e53c13958d042720ebdb
moved surfaceflinger, audioflinger, cameraservice
all native services should now reside in this location.
Change-Id: Iee42b83dd2a94c3bf5107ab0895fe2dfcd5337a8
