| d7086030fcf731e4bcef6c033cc6418cd04e6b91 |
|
10-Oct-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Support querying active record sources
Add support for querying whether there is currently a recording
underway from the specified audio source.
Bug 7314859
Change-Id: I986b231a10ffd368b08ec2f9c7f348d28eaeb892
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| cc0f1cfb69ce8b8985fc2c0984847a06a13ad22d |
|
24-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Implement android.media.AudioManager.getProperty()
Bug: 6635041
Change-Id: I3386a4a6c226bc4eceaf65556119e4fb15f73224
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 58e5aa34f01d663654d8bafad65db1dda42161ff |
|
20-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
effect_descriptor_t const correctness
Change-Id: Iad008f20d35a18acf500f773900164552fd0c19e
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 254af180475346b6186b49c297f340c9c4817511 |
|
03-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_channel_mask_t more places
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
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| dd8104cc5367262f0e5f13df4e79f131e8d560bb |
|
02-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_channel_mask_t more consistently
In IAudioFlinger::createTrack() and IAudioFlinger::openRecord(),
declare input parameter to use correct type audio_channel_mask_t.
In IAudioFlinger::getInputBufferSize(), input parameter is now channel mask
instead of channel count.
Remove unused IAudioFlinger::channelCount(audio_io_handle_t).
In AudioRecord::getMinFrameCount() and AudioSystem::getInputBufferSize(),
input parameter is channel mask instead of channel count.
Change-Id: Ib2f1c29bea70f016b3cfce83942ba292190ac965
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 20010053daabfa43fcfe781bbf004473b4c08538 |
|
22-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Remove acoustics from AudioSystem::getInput()
Change-Id: I29fb3ee5664c1f0ee0409c1bb2be087ecca637db
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 0ca3cf94c0dfc173ad7886ae162c4b67067539f6 |
|
18-Apr-2012 |
Eric Laurent <elaurent@google.com> |
rename audio policy output flags
Change-Id: I27c46bd1d1b2b5f96b87af7d05b951fef18a1312
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 1a9ed11a472493cac7f6dfcbfac2064526a493ed |
|
21-Mar-2012 |
Eric Laurent <elaurent@google.com> |
audio policy: add configuration file
removed outputs to stream mapping cache in audio system: the output for a
given stream type must always be queried from audio policy manager as the cache
is not always updated fast enough by audioflinger callback.
removed AudioFlinger::PlaybackThread::setStreamValid() not used anymore if
stream to output mapping is not cached.
Change-Id: Ieca720c0b292181f81247259c8a44359bc74c66b
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| e53b9ead781c36e96d6b6f012ddffc93a3d80f0d |
|
13-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Whitespace and indentation
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
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| a19ffb656616feec70613ba67ddfe15a504a4e76 |
|
09-Mar-2012 |
Eric Laurent <elaurent@google.com> |
Merge "audio policy: use audio_devices_t when appropriate"
|
| 6374252107fd6539397598195ea6defd5870fafb |
|
08-Mar-2012 |
Eric Laurent <elaurent@google.com> |
audio policy: use audio_devices_t when appropriate
Change-Id: I1b3a5879e81c789fb53d356af3d3a1ee2dca955f
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| b81cc8c6f3eec9edb255ea99b6a6f243585b1e38 |
|
01-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
IAudioFlingerClient::ioConfigChanged param2 const
The 3rd parameter (param2) to AudioFlingerClient::ioConfigChanged
is used as an input. So changed it from void * to const void *.
It is then cast to const OutputDescriptor *
or const audio_stream_type_t * depending on the event.
Change-Id: Ieec0d284f139b74b3389b5ef69c7935a8e5650ee
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 72ef00de10fa95bfcb948ed88ab9b7a177ed0b48 |
|
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_io_handle_t consistently instead of int
Other:
- add a comment to nextUniqueId
- made ThreadBase::mId const, since it is only assigned in constructor.
Change-Id: I4e8b7bec4e45badcde6274d574b8a9aabd046837
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| a0d68338a88c2ddb4502f95017b546d603ef1ec7 |
|
28-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use NULL not 0 for raw pointers
Use if (p != NULL) instead of if (ptr)
Change-Id: Iaec3413a59ccbf233c98fcd918cc7d70ac5da9fa
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 211eeaf17e5565b68447d29799dbf158a33cf4cf |
|
20-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
More audio_stream_type_t
Change-Id: I1260259efe0aa3fc1ef13de69758aaa592e1f815
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| eba51fb3a361f67a6a64d5a16eba6084fe27d60e |
|
23-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_source_t consistently
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
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 58f30210ea540b6ce5aa6a46330cd3499483cb97 |
|
12-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_format_t consistently, continued
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
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| d967f0a099db2b71597a3127134afd4a46287a4a |
|
20-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove redundant get()"
|
| 7fc9a6fdf146ded90b51c52f4a05d797294dcb85 |
|
10-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Remove redundant get()
get() is almost always unnecessary, except in a LOG.
Also no need to check for != 0 before calling get().
Change-Id: Ib06e7a503f86cf102f09acc1ffb2ad085025516d
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| ea3cc3bca949139e401b77f2ac0cce7ac6e76f8f |
|
20-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove dead setRingerMode(mode, mask)"
|
| 241fc78866b2aefd75cd1890df5a75b7008728e8 |
|
19-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify range check for audio_mode_t (continued)"
|
| 347966c827883711d1ec631f204e4a6ab74e9d99 |
|
18-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify range check for audio_mode_t (continued)
Missed one place in earlier CL of same name
Change-Id: I0dd25364d0b8d5d731c02d352f139a0c8d4df1a8
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 0b07b8085d7b837b4dd5f09e0c8c39408f6bdbf7 |
|
18-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dead setRingerMode(mode, mask)
Change-Id: Ia4cc8be8424a40b3dcb7ebd0264fdff4e5247f7f
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| c813985abd8ba61e999b3505f6a332574f87a1be |
|
18-Jan-2012 |
Andreas Huber <andih@google.com> |
Temporarily restore AudioSystem/AudioTrack APIs with their former signatures
until we get updated prebuilts from vendor.
Change-Id: I8aae81d2513edca0ab268053a11c8c4206879e61
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 63ad6aacc6ce6b729bf25f41376cfea731a2c1eb |
|
18-Jan-2012 |
Eric Laurent <elaurent@google.com> |
Merge "audio framework: manage stream volume per device"
|
| 83844cc2f95dc279015b47fd1e18c7cb4eabe9a1 |
|
19-Nov-2011 |
Eric Laurent <elaurent@google.com> |
audio framework: manage stream volume per device
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
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 0696400a6bb9abbed62b3b9c6aa105495dc600a2 |
|
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use audio_mode_t consistently"
|
| fff6d715a8db0daf08a50634f242c40268de3d49 |
|
13-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_stream_type_t consistently
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
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| f78aee70d15daf4690de7e7b4983ee68b0d1381d |
|
04-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_mode_t consistently
It was int or uint32_t.
Also make getMode() const.
Change-Id: Ibe45aadbf413b9158e4dd17f2b3bcc6355288d37
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 3bba0e0a60b15895134bc2c731d21fd7ebd28784 |
|
11-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix race in AudioSystem::getInputBufferSize"
|
| f8c1a6f7ef515810356816b50bfe18af95f3ec32 |
|
10-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Fix race in AudioSystem::getInputBufferSize
It was caching the recording parameters without a mutex.
Change-Id: Ic4b9f621cbc080d224c2233cf3ca3454fc0f19bd
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 930f4caa1e311ef7ff538c421a324396157eb24f |
|
07-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify range check for audio_mode_t
AudioSystem::setMode previously allowed negative modes, but these were
then rejected by AudioFlinger.
Now negative modes (including AUDIO_MODE_INVALID and AUDIO_MODE_CURRENT)
are explicitly disallowed.
Change-Id: I0bac8fea737c8eb1f5b6afbb893e48739f88d745
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 29357bc2c0dd7c43ad3bd0c8e3efa4e6fd9bfd47 |
|
06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/#/c/157220
Bug: 5449033
Change-Id: Ic9c19d30693bd56755f55906127cd6bd7126096c
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 5ff1dd576bb93c45b44088a51544a18fc43ebf58 |
|
06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/157065
Bug: 5449033
Change-Id: I00a4b904f9449e6f93b7fd35eac28640d7929e69
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| b8a805261bf0282e992d3608035e47d05a898710 |
|
20-Dec-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/156016
Bug: 5449033
Change-Id: I4c4e33bb9df3e39e11cd985e193e6fbab4635298
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 3856b090cd04ba5dd4a59a12430ed724d5995909 |
|
20-Oct-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/#/c/143865
Bug: 5449033
Change-Id: I0122812ed6ff6f5b59fe4a43ab8bff0577adde0a
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 71b63e3ef687c379368be6b02e70bd2feb0b6b8d |
|
02-Sep-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 5252593: any app can restart the runtime
Replace null device address string by empty sting.
Change-Id: I285c35f3345334e6d2190493b1a8a5aca1a361a4
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 9f6530f53ae9eda43f4e7c1cb30d2379db00aa00 |
|
30-Aug-2011 |
Eric Laurent <elaurent@google.com> |
226483: A2DP connected, but music out to speaker
When the A2DP headset is connected, there is a possible
race condition when the audio tracks are moved from
the mixer thread attached to the speaker output to the thread
attached to A2DP output.
As the request to clear the stream type to output mapping cache in
the client process is asynchronous, it is possible that the flag
indicating to the client audio track to re-create the IAudioTrack
on the new thread is processed before the cache is invalidated.
In this case, the track will be attached to the old thread and
music will continue playing over the device speaker instead of being
redirected to A2DP headset.
Change-Id: Ib2ce1eb5320eaff83287b93779061bf4e7a330df
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| db7c079f284f6e91266f6653ae0ec198b1c5006e |
|
10-Aug-2011 |
Eric Laurent <elaurent@google.com> |
Audio effects: track CPU and memory use separately
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
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 3a34befc6fb04a4945a849e8bda8b84e4bf973fe |
|
02-Aug-2011 |
Marco Nelissen <marcone@google.com> |
Keep effects sessions active when the caller dies.
Don't remove effects until the session they are in goes away or all
AudioEffects have been explicitly released. This allows the control
panel process to die without stopping the effects.
Change-Id: I4496e5df080230ca1af149dec95c1309ab8ea888
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 7c7f10bd4fda9a084e5e7f0eb3a040dfcbf01745 |
|
18-Jun-2011 |
Eric Laurent <elaurent@google.com> |
Audio framework: support for audio pre processing
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
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 0512ab559d4670c2204078470d7ef5d376811c57 |
|
05-May-2011 |
Glenn Kasten <gkasten@google.com> |
Remove dead code related to gettid
The gettid system call is always available now.
Change-Id: Ib78b41781eda182dc8605daf456bbea7ff7c2dc0
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 64760240f931714858a59c1579f07264d7182ba2 |
|
11-May-2011 |
Dima Zavin <dima@android.com> |
update for new audio.h header location
Change-Id: Ic4c62c4037800802427eb7d3c7f5eb8b25d18876
Signed-off-by: Dima Zavin <dima@android.com>
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| fce7a473248381cc83a01855f92581077d3c9ee2 |
|
20-Apr-2011 |
Dima Zavin <dima@android.com> |
audio/media: convert to using the audio HAL and new audio defs
Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| db5cb14318bb24cd6ea14ff7ceea0d5e1f83d903 |
|
20-Apr-2011 |
Dima Zavin <dima@android.com> |
libmedia: move AudioParameter out of AudioSystem
Change-Id: I9eb7e002d141936258050d4fa4f0ccd8202bfc54
Signed-off-by: Dima Zavin <dima@android.com>
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 6b2718c67aa7b1a8e3b0f25a73a0d5f72c59ffc3 |
|
04-Feb-2011 |
Glenn Kasten <gkasten@google.com> |
Bug 3352047 Wrong message when adjusting volume
Add hidden AudioManager.getDevicesForStream and output device codes.
Change-Id: I4d1c1d3b6a077cd117720817d1f733dda557b947
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| eda6c364c253ba97ee45a3adeb8c2b45db1f81db |
|
02-Feb-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3371080
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
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 56ecd20263d7f63476f756fc5d8b043b325c7bfb |
|
09-Nov-2010 |
Jean-Michel Trivi <jmtrivi@google.com> |
Add support for audio recording source in generic audio policy mgr.
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
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 8184a5734690b30f4de0f6b6c16daf1e089f67df |
|
20-Jul-2010 |
Eric Laurent <elaurent@google.com> |
resolved conflicts for merge of dd206093 to master
Change-Id: I21dd2321a4839d034d49092baccbf40986f17dae
|
| de070137f11d346fba77605bd76a44c040a618fc |
|
13-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Audio policy manager changes for audio effects
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
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| fc499ca2babff1315d0188ddfbe0268fe5d5e8ae |
|
04-Jun-2010 |
Eric Laurent <elaurent@google.com> |
am 030a1553: am 2ea200c5: Merge "Issue 2667801: [Audio Effect Framework] AudioFlinger, AudioMixer AudioTrack modifications." into kraken
|
| be916aa1267e2e6b1c148f51d11bcbbc79cb864c |
|
02-Jun-2010 |
Eric Laurent <elaurent@google.com> |
Issue 2667801: [Audio Effect Framework] AudioFlinger, AudioMixer AudioTrack modifications.
First drop of audio framework modifications for audio effects support.
- AudioTrack/AudioRecord:
Added support for auxiliary effects in AudioTrack
Added support for audio sessions
Fixed left right channel inversion in setVolume()
- IAudioFlinger:
Added interface methods for effect enumeraiton and instantiation
Added support for audio sessions.
- IAudioTrack:
Added method to attach auxiliary effect.
- AudioFlinger
Created new classes to control effect engines in effect library and manage effect connections to tracks or
output mix:
EffectModule: wrapper object controlling the effect engine implementation in the effect library. There
is one EffectModule per instance of an effect in a given audio session
EffectChain: group of effects associated to one audio session. There is one EffectChain per audio session.
EffectChain for session 0 is for output mix effects, other chains are attached to audio tracks
with same session ID. Each chain contains a variable number of EffectModules
EffectHandle: implements the IEffect interface. There is one EffectHandle object for each application
controlling (or using) an effect module. THe EffectModule maintians a list of EffectHandles.
Added support for effect modules and effect chains creation in PlaybackThread.
modified mixer thread loop to allow track volume control by effect modules and call effect processing.
-AudioMixer
Each track now specifies its output buffer used by mixer for accumulation
Modified mixer process functions to process tracks by groups of tracks with same buffer
Modified track process functions to support accumulation to auxiliary channel
Change-Id: I26d5f7c9e070a89bdd383e1a659f8b7ca150379c
/frameworks/av/media/libmedia/AudioSystem.cpp
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| 2dadcda205d995f7094b7569b076921872505143 |
|
26-May-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2712130: Sholes: problem when playing audio while recording over bluetooth SCO.
The problem is that when an input stream is opened for record over bluetooth SCO, the kernel
mono audio device should be opened in RW mode to allow further use of this same device by an output stream
also routed to bluetooth SCO.
This does not happen because of a bug in AudioSystem::isBluetoothScoDevice() that does not return true
when the device is DEVICE_IN_BLUETOOTH_SCO_HEADSET (input device for blurtooth SCO).
Change-Id: I9100e972931d8142295c7d64ec06e31304407586
/frameworks/av/media/libmedia/AudioSystem.cpp
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| be55a2d66f03e80524a346500ffa9fd046410b28 |
|
11-Mar-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2416481: Support Voice Dialer over BT SCO.
- AudioPolicyManager: allow platform specific choice for opening a direct output.
Also fixed problems in direct output management.
- AudioFliinger: use shorter standby delay and track inactivity grace period for direct output
thread to free hardware resources as soon as possible.
- AudioSystem: do not use cached output selection in getOutput() when a direct output
can be selected.
Change-Id: If44b50d29237b8402ffd7a5ba1dc43c56f903e9b
/frameworks/av/media/libmedia/AudioSystem.cpp
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| 05bca2fde53bfe3063d2a0a877f2b6bfdd6052cf |
|
26-Feb-2010 |
Eric Laurent <elaurent@google.com> |
Issue 2071329: audio track is shorter than video track for video capture on sholes
Add API to retrieve number of frames dropped by audio input kernel driver.
Submitted on behalf of Masaki Sato <masaki.sato@motorola.com>
/frameworks/av/media/libmedia/AudioSystem.cpp
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| df49e8926e85088bc7d1dc7905362437c9806b69 |
|
22-Feb-2010 |
Eric Laurent <elaurent@google.com> |
am 8978547f: am f5fe3949: Fix issue 2459650.
Merge commit '8978547f254b6b6ba2e322794aa044803f3edc2a'
* commit '8978547f254b6b6ba2e322794aa044803f3edc2a':
Fix issue 2459650.
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| 7c7fa1b51bec497cd7f46c1bdb5bb0adfaa181b2 |
|
22-Feb-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2459650.
This change fixes a problem where an unwanted tone is generated by audio policy manager when a MT call is answered.
This is because of a policy that replaces high visibility system sounds (ringtones, alarms...) by a beep when in call.
There is a transitory phase while the call is being answered where the phone state is changed to IN_CALL but the
ringtone is still playing. The audio policy manager then mutes the end of the ringtone and starts playing a beep
in replacement because the ringtone is categorized as high visibility.
The fix consists in changing the ringtone stream type from high visibility to low visibility. This is not a problem as
the only actual use case where a ringtone would be generated while in call is if another call is received.
But in this case, the phone system does not generate a ringtone but a call waiting tone instead.
It is therefore not required to handle a ringtone as a high visibiltiy tone that must be somehow signaled to the user
while in call.
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 342e9cf388cceb807def720e40e8b0a217f4bcaa |
|
20-Jan-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2285561: New AudioFlinger and audio driver API needed for A/V sync
Added getRenderPosition() API to IAudioFlinger to retreive number of audio frames
written by AudioFlinger to audio HAL and by DSP to DAC.
Added getRenderPosition() API to AudioHardwareInterface to retreive number of audio frames
written by DSP to DAC.
Exposed AudioTrack::getPosition() to AudioSink() to make it available to media player.
Removed excessive log in AudioHardwareGeneric.
/frameworks/av/media/libmedia/AudioSystem.cpp
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| b72a396826da8bd934b9531bbd40f86d7509e71c |
|
25-Jan-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2378022: AudioService should direct volume control to STREAM_VOICE_CALL stream when STREAM_VOICE_CALL stream is active.
Modified AudioService.getActiveStreamType() so that STREAM_VOICE_CALL is selected when a track using this stream
type is playing.
Chanded isMusicActive() for a more generic isStreamActive(stream) method in AudioSystem, IAudioFlinger and AudioFlinger.
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 0ef583f785528ef2785e6149d5964004cd1016b0 |
|
25-Jan-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2363154: Speech synthesis fails to start over A2DP after media server process crash.
The problem is that after a media_server crash, the value of the A2DP output handle can change.
As this value is cached in AudioSystem for all client processes there can be a mismatch between the cached
and actual value after a media_server restart.
The fix consists in clearing the cached output handles and output to stream map values cached
in AudioSystem in AudioFlingerClient::binderDied() which is called when the media_server crashes.
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| f0ee6f4055e26fb35d9c526a596668a4dc9da5ba |
|
21-Oct-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 2192181: AudioFlinger must provide separated methods to set VOICE_CALL stream volume and down link audio volume.
Added setVoiceVolume() method to AudioSystem, AudioFlinger, IAudioFlinger, AudioPolicyService.
Removed call to AudioHardwareInterface::setVoiceVolume() from AudioFlinger::setStreamVolume().
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| a9c322e398a1f5fdcace3b8b73967f010b1c31ca |
|
27-Aug-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 2045911: Camera Shutter tone does not play correctly while listening to music.
Add the possibility to delay routing and volume commands in AudioPolicyClientInterface. The delay is not blocking for the caller.
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| fa2877b9ea48baed934b866d2ab3658b69c4c869 |
|
28-Jul-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 2001214: AudioFlinger and AudioPolicyService interfaces should not use pointers as handles to inputs and outputs.
Use integers instead of void* as input/output handles at IAudioFlinger and IAudioPolicyService interfaces.
AudioFlinger maintains an always increasing count of opened inputs or outputs as unique ID.
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| c2f1f07084818942352c6bbfb36af9b6b330eb4e |
|
17-Jul-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 1795088 Improve audio routing code
Initial commit for review.
Integrated comments after patch set 1 review.
Fixed lockup in AudioFlinger::ThreadBase::exit()
Fixed lockup when playing tone with AudioPlocyService startTone()
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 7562408b2261d38415453378b6188f74fda99d88 |
|
20-May-2009 |
Mathias Agopian <mathias@google.com> |
move libbinder's header files under includes/binder
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 48f7f5e8359909ddfc6492a79a8b9c44759ca6c3 |
|
02-Apr-2009 |
Eric Laurent <> |
AI 144097: am: CL 144054 am: CL 144053 Fix issue #1751242 A2DP playback fails first time: Invalid buffer size: minFrameCount 10240, frameCount 4800
The problem comes from the fact that AudioSystem::getOutputFrameCount() calls getOutput() to retrieve the active output (A2DP or Hardware) before calling get_audio_flinger(). If it is the first time AudioSystem::getOutputFrameCount() is called in a given process, getOutput() will return a wrong value because gA2dpEnabled has not yet been updated by get_audio_flinger().
The fix consists in calling get_audio_flinger() in getOutput() to be sure that gA2dpEnabled is valid when getOutput() reads it.
Original author: elaurent
Merged from: //branches/cupcake/...
Original author: android-build
Merged from: //branches/donutburger/...
Automated import of CL 144097
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 89fa4ad53f2f4d57adbc97ae1149fc00c9b6f3c5 |
|
04-Mar-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //depot/cupcake/@135843
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 15f767b960b38059a74a42a33e16d8df2aec8bc1 |
|
04-Mar-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //depot/cupcake/@135843
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 7a2146d5807030b2629f347736be5301b61e8811 |
|
13-Feb-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //branches/cupcake/...@131421
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 5e07b5774c8b376776caa4f5b0a193767697e97e |
|
11-Feb-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //branches/cupcake/...@130745
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 7b5eb023f8d87cca6d830ae6c11c6aadbe02aca8 |
|
18-Dec-2008 |
The Android Open Source Project <initial-contribution@android.com> |
Code drop from //branches/cupcake/...@124589
/frameworks/av/media/libmedia/AudioSystem.cpp
|
| 2729ea9262ca60d93047e984739887cfc89e82eb |
|
21-Oct-2008 |
The Android Open Source Project <initial-contribution@android.com> |
Initial Contribution
/frameworks/av/media/libmedia/AudioSystem.cpp
|