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Change-Id: I9cc5d54883a3e5c75d553fabb619fc8e49f4f9e5
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Add binder methods to IAudioPolicyService to control activity
of external audio sources (e.g FM tuner).
Change-Id: I2008308a6a996baeae502b68a790d87281efe2ff
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Change-Id: I40c048c7644c46f4e4f7103875206c0785c4b1dc
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Change-Id: Ie8f9d42fc061f6d558f23b98414e04eb3d14b376
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Add limit on number of audio ports and patches requested by
listaudioPorts() and listAudioPatches().
Bug: 19261727.
Change-Id: I21dfdf11cf805734cc3b7b2a85762c5598f60580
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Add "device name" member to AudioPort.
Moved unique device ID from DevicePort to AudioPort.
Straighten out confusion between "name" and "address"
Created string constants for Intent "extra" data keys.
Combine card/address data into "address" at sender.Add additional info to AudioPort for enum/selection API.
Change-Id: I1be308e12b09cd0ead9a3e9fbc9c385462299fb5
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Change-Id: Ifd30546c11def40ad483f27c9d7baa4008fcbac3
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Add binder calls to register and unregister dynamic mixes
to AudioPolicyManager.
Bug: 16009464.
Change-Id: I4ef38166d0cfc88c984970969380d0cd6eb083ac
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of audio source." into lmp-mr1-dev
* commit '390ef84373b7edd1397381f780ca235326357094':
AudioRecord: use audio attributes instead of audio source.
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Added AudioRecord constructor with audio attributes.
Replaced AudioPolicymanager::getInput() by getInputForAttr().
No new functionality for now.
Also:
- Fixed warnings in AudioPolicyManager
- Allocate audio session ID before calling getOutputForAttr() in
AudioTrack.
Bug: 16006090.
Change-Id: I15df21e4411db688e3096dd801cf579d76d81711
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prototype." into lmp-mr1-dev
* commit 'bf14c62e7973ed793a3b00c2c67beca4484c1c7a':
audio policy: new getOutputForAttr() prototype.
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Update getOutputForAttr() prototype and group all logic
dealing with audio attributes to stream type conversion in
audio policy manager.
getOutputForAttr():
- specifies the audio session (for future use)
- returns a status code
- receives either stream type (for legacy) or audio attributes
- returns an updated streamtype
Remove logic dealing with legacy stream types to attributes conversion
from AudioTrack.
Use correct type for audio sessions in other APIs (startOutput() ...).
releaseOutput() specifies the audio session (for future use).
Bug: 18067208.
Change-Id: I1bfbe9626c04c7955d77f8a70aecfad2cb204817
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Change-Id: If10a9cc17245f95d5e10b1507445abbb4020670e
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to use the new static version.
Change-Id: Ia7b10eb38ca55b72278bfd33d3bf647f338b4e6a
Conflicts:
media/libmedia/IAudioFlinger.cpp
media/libmedia/IMediaPlayer.cpp
media/libstagefright/CameraSource.cpp
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Change-Id: I3f4110a88679b7e7841f40c6acdbf5dcde40fb19
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Bug: 18226810.
Change-Id: Ib8e2bfe835a8681aac50bf23161db14e50c9a124
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Make sure that accessibility prompts are heard when
a ringtone is active by forcing stream type
to AUDIO_STREAM_RING when phone state is AUDIO_MODE_RINGTONE.
Bug: 17558149.
Change-Id: Ia3bead8052fca5cbf282c267f7b9b06014fef628
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Audio policy:
- Added active capture indication to sound trigger service:
recognition stops if concurrent capture is not supported.
- Added generation of reserved I/O handle and session ID for
utterance capture.
Sound trigger service
- Added sound model update callback handling.
- Added service state callback
- Simplified callback shared memory allocation.
Bug: 12378680.
Change-Id: Ib0292c2733e6df90fdae480633dd9953d0016ef1
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Indicate the audio session ID when calling getInput(),
startInput(), stopInput(), releaseInput().
Bug: 12378680.
Change-Id: I763793752f93e2f4e1445a5ab217c895af011038
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Change-Id: I9f37be05f8dc7b85a8827a94e76ca0f45453e170
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In AudioPolicyManager, support querying an output
or playback strategy for audio attributes,
instead of a stream type,
In AudioTrack creation, use the output returned
for the track's attributes.
Change-Id: I0fef05845ba676404775e2e338c10e6a96237268
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Rewrite setAudioPortConfig() to take advantage of the new
AudioPortConfig class.
The configuration is now checked against AudioPort capabilities
and stored in the AudioPortConfig.
Fix a bug where the config mask was not set properly when calling
setAudioPortConfig at the PatchPanel interface.
Remove debug log in IAudioPolicyService.
Bug: 14815883.
Change-Id: I7eb7683a96e049581c13f2a212486507883d008d
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Added IAudioPolicyServiceClient client binder interface
for client process to receive notifications from AudioPolicyService
when audio ports are added/removed or audio patches created/released.
The audio patches owned by a given client are automatically released when
this client binder dies.
Bug: 14815883.
Change-Id: I6013f6aec03b50565cffb1ad2cd1f0f8852032c5
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Change-Id: I0a62e5416edc41c3a0e816275085ab18a23066f1
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Change-Id: I5776313b9b49072cd666d28880f0d07cc73f827b
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Change-Id: If16d6495c16e0d61a221f81bfd49e7d14bbfdc12
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- start() returns a status so that upper layers can
recreate a non offloaded track in case of error.
- Added states to handle offloaded tracks specific:
- waiting for stream end (drain) notification by
audio flinger
- allow pause while waiting for stream end notification
- getPosition() queries the render position directly from
audio HAL.
- disable APIs not applicable to offloaded tracks
- Modified track restoring behavior for invalidated
offloaded tracks: just send the callback and wait for
upper layers to create a new track.
- Added wait for stream end management in audio track client
proxy. Similar to obtainBuffer and should be factored in.
Change-Id: I0fc48117946364cb255afd653195498891f622bd
Signed-off-by: Eric Laurent <elaurent@google.com>
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NOTE: this does _not_ include all private member variables added
to classes as part of offload support. Only public/protected functions
and stubs functions/variables needed to make the changes buildable.
- isOffloadSupported() added to audio policy service
A stub implementation is required to build, this always returns false
- setParameters() added to IAudioTrack
A stub implementation is required to build, this always returns
INVALID_OPERATION
- CBlk flag for stream end
- Change AudioSystem::getRenderPosition() to take an audio_output_t
so caller can specify which output to query
- Add AudioSystem::isOffloadSupported()
This is fully implemented down to the AudioFlinger function
AudioPolicyServer::isOffloadSupported() which is just a stub
that always returns false.
- Add EVENT_STREAM_END to AudioTrack interface.
STREAM_END is used to signal when the hardware has actually finished
playing all the data it was sent.
- Add event type enumeration to media player interface AudioSink callbacks
so that the same callback can be used to handle multiple types of
event. For offloaded tracks we also have to handle STREAM_END and
TEAR_DOWN events
- Pass audio_offload_info_t to various functions used for opening outputs,
tracks and audio players. This passes additional information about the
compressed stream down to the HAL when using offload.
For publicly-available APIs this is an optional parameter (for some of
the internal and low-level APIs around the HAL interface it is mandatory)
- Add getParameters() and setParameters() API to AudioTrack
Currently dummy implementations.
- Change AudioPlayer contructor so that it takes a set of bitflags defining what
options are required. This replaces the original bool which only specified
whether to use deep buffering.
- Changes to StageFright class definition related to handling tearing-down of
an offloaded track when we need to switch back to software decode
- Define new StageFright utility functions used for offloaded tracks
Currently dummy implementations.
- AudioFlinger changes to use extended audio_config_t.
Fills in audio_offload_info_t member if this info is passed in when
opening an output.
- libvideoeditor changes required to add the new event type parameter
to AudioSink callback functions
- libmediaplayerservice changes required to add the new event type parameter
to AudioSink callback functions
Change-Id: I3ab41138aa1083d81fe83b886a9b1021ec7320f1
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Eric Laurent <elaurent@google.com>
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Bug 7485803
Change-Id: I0744374f130fd2dd0714102354cffed2fa915361
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Change-Id: Ib28fd7b9ce951a6933f006e7f8812ba617625530
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Add support for querying whether there is currently a recording
underway from the specified audio source.
Bug 7314859
Change-Id: I986b231a10ffd368b08ec2f9c7f348d28eaeb892
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Change-Id: Iad008f20d35a18acf500f773900164552fd0c19e
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Use it in AudioSystem::getOutput(), AudioSystem::getInput(),
IAudioPolicyService::getOutput(), IAudioPolicyService::getInput(),
and various other places in AudioFlinger.
Not done: AudioTrack and OutputDescriptor.
Change-Id: I70e83455820bd8f05dafd30c63d636c6a47cd172
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Change-Id: I29fb3ee5664c1f0ee0409c1bb2be087ecca637db
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Change-Id: I27c46bd1d1b2b5f96b87af7d05b951fef18a1312
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Fix indentation to be multiple of 4.
Make it easier to search:
sp< not sp < to
"switch (...)" instead of "switch(...)" (also "if" and "while")
Remove redundant blank line at start or EOF.
Remove whitespace at end of line.
Remove extra blank lines where they don't add value.
Use git diff -b or -w to verify.
Change-Id: I966b7ba852faa5474be6907fb212f5e267c2874e
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Change-Id: I1b3a5879e81c789fb53d356af3d3a1ee2dca955f
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Was a mix of audio_source_t, uint8_t, and int.
Related fixes:
- fix comments in MediaRecorder.java
- AudioPolicyService server side was not checking source parameter at
all, so if the client wrapper was bypassed, invalid values could be
passed into audio HAL
- JNI android_media_AudioRecord_setup was checking source for positive
values, but not negative values. This test is redundant, since already
checked at Java and now checked by AudioPolicyService also, but might
as well make it correct.
Change-Id: Ie5e25d646dcd59a86d7985aa46cfcb4a1ba64a4a
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Was int or uint32_t.
When AudioFlinger::format can't determine the correct format,
return INVALID rather than DEFAULT.
Init mFormat to INVALID rather than DEFAULT in the constructor.
Subclass constructors will set mFormat to the correct value.
Change-Id: I9b62640aa107d24d2d27925f5563d0d7407d1b73
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Change-Id: Ia4cc8be8424a40b3dcb7ebd0264fdff4e5247f7f
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Improve volume management by keeping track of volume for each type
of device independently.
Volume for each stream (MUSIC, RINGTONE, VOICE_CALL...) is now maintained
per device.
The main changes are:
- AudioService now keeps tracks of stream volumes per device:
volume indexes are kept in a HashMap < device , index>.
active device is queried from policy manager when a volume change request
is received
initalization, mute and unmute happen on all device simultaneously
- Settings: suffixes is added to volume keys to store each device
volume independently.
- AudioSystem/AudioPolicyService/AudioPolicyInterface: added a device argument
to setStreamVolumeIndex() and getStreamVolumeIndex() to address each
device independently.
- AudioPolicyManagerBase: keep track of stream volumes for each device
and apply volume according to current device selection.
Change-Id: I61ef1c45caadca04d16363bca4140e0f81901b3f
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It was int or uint32_t.
Also make getMode() const.
Change-Id: Ibe45aadbf413b9158e4dd17f2b3bcc6355288d37
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At native level it was a mixture of audio_stream_type_t, int, uint32_t,
and uint8_t. Java is still int. Also fixed a couple of hard-coded -1
instead of AUDIO_STREAM_DEFAULT, and in startToneCommand a hard-coded 0
instead of AUDIO_STREAM_VOICE_CALL.
Change-Id: Ia33bfd70edca8c2daec9052984b369cd8eee2a83
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Before this change, CPU and memory usage for an audio effect were
registered and checked against the limit by audio policy manager
upon effect instantiation. Even if an effect was not enabled
it would prevent another effect to be created if the CPU load budget
was exceeded, which was too restrictive.
This change adds a method to register/unregister CPU load only when
an effect is enabled or disabled.
It also adds a mechanism to place all effects on the global output mix
in suspend state (disabled) when an effect is enabled on a specific session.
This will allow applications using session effects to have the priority
over others using global effects.
Also fixes some issues with suspend/restore mechanism:
- avoid taking actions when an effect is disconnected and was not enabled.
- do not remove a session from the suspended sessions list when corresponding
effect chain is destroyed.
Change-Id: I5225278aba1ae13d0d0997bfe26a0c9fb46b17d3
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Added APIs to control pre processes applied on captured audio.
Those APIs are still hidden until reviewed by API council.
Three types of standard pre processes are supported:
- Automatic Gain Control (AGC) by AutomaticGainControl class
- Acoustic Echo Cancellation (AEC) by AcousticEchoCanceler class
- Noise Suppression (NS) by NoiseSuppressor class
A method is added to AudioEffect class to query audio pre processings
applied by default by the platform on a given AudioRecord session ID.
Change-Id: I0b9fceeb8c704dd06319c3b52b85c96fe871d51d
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Audio effect framework is extended to suport effects on
output and input audio path.
AudioFlinger: Support for audio effects and effect chains is
moved from PlaybackThread class to ThreadBase class so that
RecordThread can manage effects.
Effects of type pre processing are allowed on record thread
only. When a pre processing is enabled, the effect interface handle is
passed down to the input stream so that the audio HAL can call the
process function. The record thread loop calls the effect chain process
function that will only manage the effect state and commands and skip the
process function.
AudioRecord: The audio session is allocated before calling getInput() into
audio policy serice so that the session is known before the input theead is
created and pre processings can be created on the correct session.
AudioPolicyService: default pre processing for a given input source are
loaded from audio_effects.conf file.
When an input is created, corresponding effects are created and enabled.
Change-Id: Id17119e0979b4dcf189b5c7957fec30dc3478790
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Change-Id: Ic4c62c4037800802427eb7d3c7f5eb8b25d18876
Signed-off-by: Dima Zavin <dima@android.com>
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Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
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