Record and playback objects (resp AudioRecord and AudioTrack)
are created using a channel mask, but this information is lost
in the mixer because only the channel count is known to
AudioFlinger. A channel count can always be derived from a
channel mask.
The change consists in:
- disambiguiting variable names for channel masks and counts
- passing the mask information from the client to AudioFlinger
and the mixer.
- when using the DIRECT ouput, only verifying the format of
the track is compatible with the output's for PCM.
Change-Id: I50d87bfb7d7afcabdf5f12d4ab75ef3a54132c0e
Moved and renamed media/EffectApi.h to hardware/audio_effect.h
Modified the effect library API to expose a library info structure
containing an interface functions table.
Also removed enums for audio channels, audio format and devices
from effect API and use values from system/audio.h instead.
Modified effects factory to support new library interface format and
load libraries and efffects listed in audio_effects.conf file.
The file audio_effects.conf is first loaded from /vendor/etc and
then from /system/etc/audio_effects.conf if not found.
Modified existing effect libraries to implement the new library interface.
Change-Id: Ie52351e071b6d352fa2fbc06c3846686f8c45df9
Fix two issues in audio effect framework reported by partners.
1 - Fixed duplicated audio buffer sent to effect process function when
pausing a track.
Modified Effectchain::process_l() function to clear the effect chain
input buffer before calling the effect process functions when no track
is active on the session. Previous code was clearing the buffer after
calling the process functions and when transitioning from active
to inactive, the last processed buffer was passed again once to effect
process function before being cleared.
2 - Fixed potential mutex cross deadlock when disconnecting an effect
while playback is active. This is because EffectChain::process_l()
was calling PlaybackThread::hasAudioSession() thus creating an inversion
in the mutex lock order (EffectChain mutex locked before ThreadBase mutex).
The fix consists in removing the call to hasAudioSession() from process_l()
and requires each effect chain to keep count of the number of audio tracks
attached to it (previously only the active tracks were accounted for).
Change-Id: Iee4246694ea8c7a66c012120c629d72dd38f9c35
Keep track of the primary interface that handles the master volume,
etc.
Change-Id: Ib0701fccff8d8783a99035a241ab7c8ec75c00ac
Signed-off-by: Dima Zavin <dima@android.com>
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
