Previously, the optimized asm option is only enabled when
__ARM_ARCH_5E__ is defined, which is assigned in armv5te.mk
rather than armv7-a series targets. This patch checks the ARM CPU
feature about half-word multiply instructions to enable ARMv5TE
resampler optimization routines properly.
Change-Id: I4c5a5d8c932416f23bedb0b389db958349f21ea4
Changes:
- Move declaration of kClassPathName to top of file so it can be used
in more than one place, instead of "android/media/AudioSystem".
- Make private methods static.
- Add comment to stream_type, audio_mode, force_use types that they must match
values in AudioSystem.java.
- Add comment about unused types mp3_sub_format and vorbis_sub_format.
- Fix typos.
- Use @ in javadoc comments.
- Delete dead APIs setMode, getMode, setRouting, getRouting in AudioSystem.java
(they are all hidden, deprecated, and unused by rest of framework)
- Delete unused private log method.
- Fix pathname for android_media_AudioSystem.cpp.
- Improve code formatting for space after == and !=.
- Add logging of delta for changing audio policy manager ref count.
Change-Id: I18037c7beb8ab76d1fda08c11e589f6e591d36e1
The first fix (commit 913af0b4) is problematic because it makes threads
in mediaserver process block on the cblk mutex. This is not permitted
as it can cause audio to skip or worse have a malicious application
prevent all audio playback by keeping the mutex locked.
The fix consists in using atomic operations when modifying the control
block flags.
Also fixed audio_track_cblk_t::framesReady() so that it doesn't block
when called from AudioFlinger (only applies when a loop is active).
Change-Id: Ibf0abb562ced3e9f64118afdd5036854bb959428
Make sure that all read/modify/write operations on the AudioTrack
and AudioRecord control block flags field are protected by the
control block's mutex.
Also fix potential infinite loop in AudioTrack::write() if the
written size is not a multiple of frame size.
Change-Id: Ib3d557eb45dcc3abeb32c9aa56058e2873afee27
This change makes sure that the VOICE_CALL stream volume tracks
the BLUETOOTH_SCO stream volume when SCO audio is enabled.
The down link audio volume now reflects what is being displayed
when pressing volume hard keys on the device while in a video chat
with a BT SCO headset.
Volume settings on the headset and the device are still independent as
we do not support handsfree profile yet.
Change-Id: Ie0d2714730ea359b9318b9cbe6f0b2557ef0f976
- To track the currently used audio device
- The devices are separated as speaker and other audio devices
- Provide the collected data to battery application through pullBatteryData()
Change-Id: I374c755266b5ac6b1c6c630400f4daf901ea8acc
Do not select A2DP output for media strategy when it is suspended because
BT SCO is active. Media audio will be routed to speakers or SCO HS
(depending on phone state and activity on stream VOICE_CALL) which is less
confusing than not hearing anything while music progress bar is moving.
Change-Id: Iff8cc1ea9bf9bde0b33035c4d91398db0934b836
The problem is that when an AudioRecord using the resampler is restarted,
the resampler state is not reset (as there is no reset function in the resampler).
The consequence is that the first time the record thread loop runs, it calls the resampler
which consumes the remaining data in the input buffer and when this buffer is released
the input index is incremented over the limit.
The fix consists in implementing a reset function in the resampler.
A similar problem was also present for playback but unoticed because the track buffer is always
drained by the mixer when a track stops. The only problem for playback was that the initial
phase fraction was wrong when restarting a track after stop (it was correct after a pause).
Change-Id: Ifc2585d685f4402d29f4afc63f6efd1d69265de3
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
When resampling too short sound, AudioMixer uses previous
tracks buffer. So we re-initialize the temporary buffer per
loop to avoid it.
Change-Id: I55a59a3b14faa8445e09c450478fe79cef704760
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
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
