| c4640c9eef850bb1c754bd6b477f1cc8350c6081 |
23-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Increase allowed number of VHQ resamplers to 3
Bug: 7378660
Change-Id: I69e33ca2eb4bb9bd38e2c63df62cd1130d68baf6
udioResampler.cpp
|
| 61ea117b03b53382b5ecbc33004c7d37ea70ea8b |
21-Oct-2012 |
Mathias Agopian <mathias@google.com> |
Fix a typo that caused the high quality resampler to produce garbage
the problem is that if libaudio_resampler is present, it is those
coefficients that will always be selected, but the correct
meta-data.
Bug: 7385994
Change-Id: Ieebeb37b4dfb62a1a051bc29fae2ce056dbc6621
udioResamplerSinc.cpp
|
| 291f824e02ff517a34cfe50220b4e2b402ee998d |
19-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Remove active track when thread goes to standby
Bug: 7369232
Change-Id: I7ff9f525dad4be0aef562a53015b06ee7d3d50f1
udioFlinger.cpp
|
| a045dcafd2b77036210f5b72e79d745ad4c1b848 |
16-Oct-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Fix track estimation for presentation complete
Audio tracks were not using the right latency estimation for
signalling the completion of their presetation. This caused
the synchronization mechanism between playback and record to be
off, and a synchronized recording would contain some of the audio
that was meant to be over once recording would start.
Use the playback thread's latency reporting which takes the audio
pipe into account.
Bug 7237669
Change-Id: I23a907a53ad0b0d68d246789ec595a77a79fced5
udioFlinger.cpp
|
| 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
udioPolicyService.cpp
udioPolicyService.h
|
| bac4a4a9073a440bc6df7ca2306604819aa1c342 |
09-Oct-2012 |
Eric Laurent <elaurent@google.com> |
audio policy: deprecate ro.camera.sound.forced
The volume and routing policy of AUDIO_STREAM_ENFORCED_AUDIBLE is
now controlled by AudioService.
Do not read ro.camera.sound.forced is not needed anymore.
Bug 7032634.
Change-Id: Ic0a6396fc4b6efb91cdb4dffe0c8eb035d0440bd
udioPolicyService.cpp
|
| c8823995d16b909fae30ff4f94217e875d3e8c8a |
01-Oct-2012 |
ty.lee <ty.lee@lge.com> |
audioflinger/resampler: add build source for libaudio-resampler
Bug: 7229644
Change-Id: I93bde36be1c3ec84174a4c98423e28f8b3d8782f
Signed-off-by: ty.lee <ty.lee@lge.com>
Signed-off-by: Iliyan Malchev <malchev@google.com>
ndroid.mk
|
| ac6020508acedd316391dee42329040bf45f8d90 |
01-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Integrate improved coefficient sinc resampler: VHQ
Summary:
Very high quality is enabled only for 44.1 -> 48 or 48 -> 44.1,
and uses low quality for all other use cases.
Track estimated CPU load and throttles the quality based on load;
as currently configured it should allow up to 2 instances of very high quality.
Medium quality and high quality are currently disabled unless explicitly requested.
Details:
Only load .so the first time it is needed.
Cleanup code style: formatting, indentation, whitespace.
Restore medium quality resampler, but it is not used (see next line).
Fix memory leak for sinc resampler.
Check sample rate in resampler constructor.
Add logs for debugging.
Rename DEFAULT to DEFAULT_QUALITY for consistency with other quality levels.
Renumber VERY_HIGH_QUALITY from 255 to 4.
Use enum src_quality consistently.
Improve parsing of property af.resampler.quality.
Fix reentrancy bug - allow an instance of high quality and an instance
of very high quality to both be active concurrently.
Bug: 7229644
Change-Id: I0ce6b913b05038889f50462a38830b61a602a9f7
ndroid.mk
udioMixer.cpp
udioResampler.cpp
udioResampler.h
udioResamplerCubic.h
udioResamplerSinc.cpp
udioResamplerSinc.h
|
| 087dd8e7232e4c009e9121ab7e8c37985522c9ad |
27-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Disable audio watchdog
It's not critical, and is wasting power
Bug: 7241714
Change-Id: I6ad4375f0000c92529688723dbe0ff0caa809c5d
ndroid.mk
udioFlinger.cpp
|
| 2bfc6b42b3733c12485dd51ed95191956abc3e4e |
28-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
bug 7253033 clean up before closing an output
An output can only be closed if there is no lock contention that
prevents ThreadBase::exit() from being blocked. If an output
device is waiting for an operation to complete (here a write
in the remote_submix module, because the pipe is full), signal
the module that it's entering the "exiting" state.
Change-Id: I8248add60da543e90c25a4c809866cdb26255651
udioFlinger.cpp
udioFlinger.h
|
| 76b111685010e1fea7c0a865c038aee35507fde4 |
17-Jan-2012 |
SathishKumar Mani <smani@codeaurora.org> |
audioflinger: use resample coefficients from audio-resampler library.
-Add a separate quality VERY_HIGH_QUALITY in resampler
-Use resample coefficients audio-resampler library for
quality VERY_HIGH_QUALITY.
-This improves the quality of resampled output.
Bug: 7024293
Change-Id: Ia44142413bed5f5963d7eab7846eec877a2415e4
Signed-off-by: Iliyan Malchev <malchev@google.com>
ndroid.mk
udioResampler.cpp
udioResampler.h
udioResamplerSinc.cpp
udioResamplerSinc.h
|
| 12bff5dd87e6b863482e6bfd9f4561078ebcf3f0 |
24-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
audioflinger/resampler: Add VERY_HIGH_QUALITY audio-resampler
Change-Id: I34f5d36ae60010ec64222d6660d10a84da3bf566
Bug: 7024293
Signed-off-by: Iliyan Malchev <malchev@google.com>
udio-resampler/Android.mk
udio-resampler/AudioResamplerCoefficients.cpp
udio-resampler/dnsampler_filter_coefficients_x128_10112011.h
udio-resampler/resampler_filter_coefficients_10042011.h
|
| 842c5d9553f3f8e97d04ed1bd0d37e4851240654 |
26-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Revert "Don't wait for presentation complete if terminated"
This reverts commit 44cda3a4e7ca3de0db9cb49145def3803b03ebb4
Change-Id: I7fd29b77690dab057ac966a42fb198b2772f092c
udioFlinger.cpp
|
| cc0f1cfb69ce8b8985fc2c0984847a06a13ad22d |
24-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Implement android.media.AudioManager.getProperty()
Bug: 6635041
Change-Id: I3386a4a6c226bc4eceaf65556119e4fb15f73224
udioFlinger.cpp
udioFlinger.h
|
| b6ba2fd0d61a4844c153e17843fbe2c841c4bf57 |
25-Sep-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: improve record start and thread exit
Use broadcast() instead of signal() on the
thread wake up condition when starting record or requesting thread
exit to make sure that if another thread is waiting for the same
condition (e.g binder thread calling setParameters()) the mixer
thread will be woken up.
Bug 7184317.
Change-Id: I3154a4509ca7af6ffae5236e522b0fab8e75ed06
udioFlinger.cpp
|
| 8d452f1c61aa70c4a86c4114532f2c0983120ef8 |
24-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Check clock_gettime(CLOCK_MONOTONIC) for failure" into jb-mr1-dev
|
| 80b3273cea8660fe8a5868d024d2788a1e083ffc |
24-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Check clock_gettime(CLOCK_MONOTONIC) for failure
Bug: 7100774
Change-Id: I15a84a19bb6d6ef1d9dac4beaa03587638196404
astMixer.cpp
|
| 1afc26db11b71c43f63a0f72a45a803f1a7910dd |
24-Sep-2012 |
Eric Laurent <elaurent@google.com> |
fix end of track presentation on suspended output
The code detecting the end of an audio track presentation before
removing it from the active track list is based on the
count of audio frames sent to audio HAL. When an output stream
is suspended (e.g. A2DP when SCO is active), this count does not
change and a track in stopped state will never be removed from
active track list causing the mixer thread to never release
the wake lock.
The fix consists in incrementing the audio HAL frame count even
if the output is suspended.
Also fix a problem in getRenderPosition() when the output is suspended.
Bug 7167534.
Change-Id: I3be836cbbea29b65dc087199cac6a1cd84c0a41d
udioFlinger.cpp
|
| 896adcd3ae6a1c7010e526327eff54e16179987b |
13-Sep-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: send priority request from a thread
When creating a fast AudioTrack, a request is sent to SchedulingPolicyService
to elevate the requesting thread priority. This generates a binder
call into system_server process and to a JAVA service via JNI.
If the thread from which the track was created is in the system_server
process and does not have the "can call java" attribute, a crash occurs because
the binder optimization reuses the same thread to process the returning binder
call and no JNI env is present.
The fix consists in sending the priority change request from the AudioFlinger
mixer thread, not from the binder thread.
This also reverts the workaround in commit 73431968
Bug 7126707.
Change-Id: I3347adf71ffbb56ed8436506d4357eab693078a3
udioFlinger.cpp
udioFlinger.h
|
| 4362f5300162ed55f56cf167eccf7f2e5b89d435 |
13-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Handle audio HAL returning 0 bytes in record thread
Enter standby when HAL returns an error, but also consider 0 bytes
returned as NOT_ENOUGH_DATA.
Change-Id: Ica83142310e9c176f936e0440571a6034cbc575f
udioFlinger.cpp
|
| 52762410dbc9189cd92a4094a1dbd4cfe8e71cb6 |
13-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Don't deadlock on AudioRecord start when reads return 0
When calling start() on an AudioRecord with a HAL that
returns 0 on a read() operation, the start blocking
condition was never unblocked.
Add a boolean to track the first read operation so the returned
number of bytes (mBytesRead) is only evaluated after that
first read.
Change-Id: I8c735a00d48cd6a0da467ccdf75d3616b38f6afa
udioFlinger.cpp
|
| fe3156ec6fd9fa57dde913fd8567530d095a6550 |
11-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Communicate audio session ID to downmixer
The audio downmixer effect might need the audio session Id, pass it
from the track creation in AudioFlinger to the downmix effect
creation in AudioMixer.
Change-Id: I5e29540542ae89cf4a0cdb537b3e67f04442a20a
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
astMixer.cpp
|
| 6ce3d19cd35a49add6893298f70b87917fc60d52 |
09-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Add outlier statistics for fast mixer cycle times" into jb-mr1-dev
|
| 1ab212cf5cfa5b88c801840e11e3191cd10f48e4 |
07-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Add outlier statistics for fast mixer cycle times
Change-Id: I31c964caeb8b5d9ae0a426224f030cdcb01114a0
astMixer.cpp
|
| f1c04f952916cf70407051c9f824ab84fb2b6e09 |
28-Aug-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: changes for new audio devices enums
The ThreadBase class now has a separate member for input
and output devices (mInDevice, mOutDevice).
Only query get_supported_devices() from audio HAL if the function
is exposed and if the audio policy manager did not specify the
audio module to open.
Also fixed bug in AEC preprocessing that would reset
to default output device when an input device was given.
Change-Id: I19d4d06aeb920b068e3ef31e6e6be6345ce5d67a
udioFlinger.cpp
udioFlinger.h
|
| 57b2dd1e78af53115985f18d31ec5421c9da947e |
01-Sep-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: send audio source to audio effects
Added support for EFFECT_CMD_SET_AUDIO_SOURCE audio effect
command to inform preprocessings of current audio source
selection for capture.
Change-Id: Ib2418a9aa8114e8457fe828ecd43b230ed86cdd6
udioFlinger.cpp
udioFlinger.h
|
| c3ae93f21280859086ae371428ffd32f39e76d50 |
30-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Update audio comments
Change-Id: Ie7504d0ddb252f7e4d4f99ed0b44cfc7b1049816
udioFlinger.cpp
udioFlinger.h
|
| 7aa25591769685ae0e8349b3ca3534c724484375 |
03-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dead code
RecordThread::isValidSyncEvent() returns false, so most of
RecordThread::setSyncEvent() is never executed.
Change-Id: I0cf848beb46a367a45126d2df3073c5afa2ca59c
udioFlinger.cpp
|
| 2dd4bdd715f586d4d30cf90cc6fc2bbfbce60fe0 |
29-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Move libnbaio out of AudioFlinger
libnbaio is now a separate shared library from AudioFlinger, rather
than a static library used only by AudioFlinger.
AudioBufferProvider interface is now also independent of AudioFlinger,
moved to include/media/
Change-Id: I9bb62ffbc38d42a38b0af76e66da5e9ab1e0e21b
ndroid.mk
udioBufferProvider.h
udioBufferProviderSource.cpp
udioBufferProviderSource.h
udioFlinger.cpp
udioFlinger.h
udioMixer.h
udioResampler.h
udioStreamInSource.cpp
udioStreamInSource.h
udioStreamOutSink.cpp
udioStreamOutSink.h
xtendedAudioBufferProvider.h
astMixerState.h
ibsndfileSink.cpp
ibsndfileSink.h
ibsndfileSource.cpp
ibsndfileSource.h
onoPipe.cpp
onoPipe.h
onoPipeReader.cpp
onoPipeReader.h
BAIO.cpp
BAIO.h
ipe.cpp
ipe.h
ipeReader.cpp
ipeReader.h
ourceAudioBufferProvider.cpp
ourceAudioBufferProvider.h
oundup.c
oundup.h
|
| 0c676cd04ff15cc7eacf6144b4aead6deaccf60c |
22-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Make tee sink compile OK again
Change-Id: Ie865ceeabaeb698a62c11eac9f96e303c44c28ae
ipeReader.cpp
ipeReader.h
|
| 106e8a42038f9e90d5ff97f8ab6f1a42258bde9e |
02-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
const methods
Change-Id: I92e32ee16274c032c9d0ce910676be2a7fa52471
udioFlinger.cpp
udioFlinger.h
|
| 0dbb356050d0db9e0043dd43045c1864a933332b |
03-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify AudioFlinger::PlaybackThread::isValidSyncEvent()
Change-Id: I3e4af69b929d4ca04afaac26c7e41c89fce25b9c
udioFlinger.cpp
|
| d23eedca9b5a1812891c05d89850ab7ee707040d |
02-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Discard setSyncEvent() return value
setSyncEvent() returns a status_t which is sometimes ignored.
Emphasize this is intentional by casting to void.
Change-Id: Ic614988347cba36bd2504d7ad321594a355b0d9d
udioFlinger.cpp
|
| a959904968f1d300fe86334bf5099ae2dc39ee28 |
13-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Replace hard-coded "2" by a constant and comment" into jb-mr1-dev
|
| fcfc61da927e98d18cdcb53da8fa0a8a32ee1422 |
10-Aug-2012 |
Jean-Baptiste Queru <jbq@google.com> |
am ff4c64d6: am c64a1787: Merge "Fix initialization of audio mixer track resources for MIPS."
* commit 'ff4c64d6b1b58494a1b15f1a664e5c421d095d15':
Fix initialization of audio mixer track resources for MIPS.
|
| 3ed292031dc50c56110cdadb1e3778117e3be76a |
08-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Replace hard-coded "2" by a constant and comment
Bug: 6679403
Change-Id: I6c2701f9afedc26540dfad0b4e23348bbc4cb01a
udioFlinger.cpp
|
| 0d27c65ddb5c968baa6db0c26e80f5c451bc52bc |
07-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Tune the overrun correction
The new overrun correction is: if the previous write() cycle time is <
50% of expected, then sleep enough extra to make the total cycle time =
95% of expected. This should help compensate for HAL implementations
that have uneven cycle times due to sample rate conversion.
Bug: 6881638
Change-Id: I5ab58afdfceee7ac523177f021fbf62d743d571a
astMixer.cpp
|
| 2c3b2da3049627264b7c6b449a1622f002210f03 |
03-Aug-2012 |
John Grossman <johngro@google.com> |
AudioFlinger: fix timed audio
(cherry picked from commit e20ac92c564a2f4e8123885807abdf0a78de0dd7)
> AudioFlinger: fix timed audio
>
> Addresses Bug 6900517.
>
> Finish up support for timed audio in the new FastMixer world. Pay special
> attention to remaining lock-less and voluntary yield free on the FastMixer
> thread. This fixes audio playback for Q on JB.
>
> Change-Id: Iaf815e58a1b1d0a0190051794bec8dc5c9231785
> Signed-off-by: John Grossman <johngro@google.com>
Change-Id: I9bd687acc345a05867af48e71116690fdb0ce1b5
Signed-off-by: John Grossman <johngro@google.com>
udioBufferProviderSource.cpp
udioBufferProviderSource.h
udioFlinger.cpp
udioStreamOutSink.cpp
udioStreamOutSink.h
astMixer.cpp
onoPipe.cpp
onoPipe.h
onoPipeReader.cpp
onoPipeReader.h
BAIO.cpp
BAIO.h
ourceAudioBufferProvider.cpp
|
| ee578c0330319f04a48bccbdb26b53fea0388d04 |
24-Jul-2012 |
John Grossman <johngro@google.com> |
AudioFlinger: Better handling for master volume/mute
(cherry picked from commit 93d906837e0e89aa1d9c913ab2b531b809f9bb9e)
> AudioFlinger: Better handling for master volume/mute
>
> Changes to address bug 6842827.
>
> When a HAL is loaded, cache whether or not the HAL supports
> set_master_volume/mute in the AudioHwDevice structure. Store an
> AudioHwDevice in AudioStream(In|Out) structures instead of just an
> audio_he_device_t. This give threads (PlaybackThreads in
> particular) access to the cached capabilities.
>
> When setting master volume/mute, change the system to always set the
> setting on all HAL which support it and also to set the setting on all
> PlaybackThreads. Change PlaybackThreads to apply the setting at the
> in SW mix stage of the pipeline if its assigned HAL does not support
> the setting, or to ignore the setting of the assigned HAL does support
> it.
>
> Change-Id: Ia14137a30b4c3ee6f2d7ddcc8cba87bf5eec87f4
> Signed-off-by: John Grossman <johngro@google.com>
Change-Id: Icb6bc13764e100a2003eb1dee2231132ab287d98
Signed-off-by: John Grossman <johngro@google.com>
udioFlinger.cpp
udioFlinger.h
|
| d8f178d613821c3f61a5c5e391eb275339e526a9 |
20-Jul-2012 |
John Grossman <johngro@google.com> |
Change audio flinger to user HAL master mute if available
(cherry picked from commit 91de9b56282d126ffb36344266af5fee3cefcfdd)
> Change audio flinger to user HAL master mute if available
>
> Hand merge from ics-aah
>
> > Change audio flinger to user HAL master mute if available: DO NOT MERGE
> >
> > Replicate the pattern used for HAL master volume support to make use
> > of master mute support if the HAL supports it. This is part of the
> > change needed to address bug 6828363. Because of the divergences
> > between ICS and master, this change will need to be merged by hand.
> >
> > Signed-off-by: John Grossman <johngro@google.com>
> > Change-Id: I6d83be524021d273d093bcb117b8f2fe57c23685
>
> Change-Id: I32280582905c969aaec2bb166ec5c61df82d737a
> Signed-off-by: John Grossman <johngro@google.com>
Change-Id: I5cd709187221d307fe25c5117ccaadca5f6b197b
Signed-off-by: John Grossman <johngro@google.com>
udioFlinger.cpp
udioFlinger.h
|
| 3c0a0e8541846427db0587c2fffb90f60ee680b0 |
02-Aug-2012 |
Paul Lind <plind@mips.com> |
Fix initialization of audio mixer track resources for MIPS.
The value 1 << 32 (maxNumTracks defaults to 32) is surprisingly not defined
in C, and differs on MIPS than the other arch.
Therefore the track resources were not initialized properly, resulting
in failure to play any audio.
The fix allows the mConfiguredNames bitmask to be correctly set to all
1's in the 32-track case, for all arch.
Change-Id: Ied3e1305952e9567602e2cd76c5ef3acb0809ee0
Signed-off-by: Paul Lind <plind@mips.com>
udioMixer.cpp
|
| 44cda3a4e7ca3de0db9cb49145def3803b03ebb4 |
01-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Don't wait for presentation complete if terminated
Change-Id: Ia04cf6c620693457dca87b4ffea5dd0fe71efdce
udioFlinger.cpp
|
| 510a3d6b8018a77683dac466127ffd0af34bef6e |
16-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Start adding support for multiple record tracks
Replace single mTrack by vector mTracks.
Destroy record tracks similarly to playback tracks.
Dump all record tracks, in addition to the active record track.
Change-Id: I503f10b51928b6b92698fe1c51a9ddd3215df1f4
udioFlinger.cpp
udioFlinger.h
|
| 0ec23ce0d1ff79566c402bc30df3074f6e25a22b |
10-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Clean up start() parameters
Document where int is used instead of AudioSystem::sync_event_t
(probably because of a header file dependency).
TrackBase::start() and RecordTrack::start() don't need default parameters.
Change-Id: I82f4a4d078be900f3aa4bd926697e32f5ed68ec8
udioFlinger.cpp
udioFlinger.h
|
| e4e2a37dbe2a4d923232305549101f779a2e3638 |
23-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Extract methods to enter standby and standby mode
Also move initial standby from to threadLoop to avoid a race condition.
Change-Id: I65afca83c36fb41b983b3b1d3dab35d4029560e3
udioFlinger.cpp
udioFlinger.h
|
| 0a7af18d0308295405491f86603e3d119450aba0 |
10-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use valueAt instead of editValueAt when possible
Change-Id: I885b169f4b176a6b5c2ca9a534214b4ffff1700e
udioFlinger.cpp
udioFlinger.h
|
| 58e5aa34f01d663654d8bafad65db1dda42161ff |
20-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
effect_descriptor_t const correctness
Change-Id: Iad008f20d35a18acf500f773900164552fd0c19e
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| 1d491ff06f4b9c90ff24fe953b90d0843eaf1c04 |
16-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Fix races in AudioRecord stop()
Change-Id: Id0ac1915f57fef4a938c7f90989c1162a8b6c51c
udioFlinger.cpp
udioFlinger.h
|
| 69d799679c8c0308e42057e7b5ad63a7ae806480 |
19-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use upmix/downmix utility routines
Change-Id: I9ae2ec938fb695ec576ea008a42205325af7bbf1
udioFlinger.cpp
|
| e65c89113232d070dd9153c54ca19301bca7a162 |
21-Jul-2012 |
Eric Laurent <elaurent@google.com> |
fix audio effect not destroyed when needed.
commit a5f44eba contained an error which made that audio effect
modules where not destroyed but left in the destroyed state
rendering them unuseable after being released.
Bug 6805168.
Change-Id: Ia4e683b3c970ffd01846c482fde73d799ff219de
udioFlinger.cpp
|
| 33e6e35b03a726e35203e97550f32154c91d5f13 |
17-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Miscellaneous audio record fixes
Call AudioSystem::stopInput() if exitPending() after wait() returns.
Acquire lock before clearing mActiveThread.
Change-Id: Ia55e4c4b3accc65ad5479cbdc094fd919152af9f
udioFlinger.cpp
|
| cd2d61016527bf48bd2e9a920bb3fdbb875eb3e4 |
19-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use constructor to initialize instead of memcpy
Also don't check for thread parameter as it's always non-NULL
Change-Id: Id23ded1370556ef3f76f81f5f0c6fa644bcba681
udioFlinger.cpp
|
| be5f05e0fdfc4e3799653702187861a2afa072ee |
19-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Internal dump methods return void not status_t
Only the IAudioFlinger::dump() needs to return a status_t.
Change-Id: Iffeb2a7db4846df850b6b2ed960276f1fd75dba0
udioFlinger.cpp
udioFlinger.h
|
| 5ad92f620fbbb6a8281f10169a23d38e3601e07a |
19-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Revert 94479fd5405642c67efd14cebe722feb9cbe6e77
Change-Id: I5ca78d5462badf541868785b2ba2e3f6d0cf492a
udioFlinger.cpp
|
| bb4350d3b9e9485ae59e084de270f86aecef8066 |
04-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_devices_t more places
Change-Id: Id0ace02ca6f480a6c46e11880acf6cdc66d83184
udioFlinger.cpp
udioFlinger.h
|
| a189a6883ee55cf62da1d7bf5bf5a8ab501938a4 |
20-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Use struct assignment instead of explicit memcpy
for POD structs effect_descriptor_t and effect_config_t
Change-Id: Ib2fc47f85fb65ed91b0abb1f87217c49b5eb571d
udioFlinger.cpp
udioPolicyService.cpp
|
| fd4e20c226eca185fc789de761beae64855bfbbb |
04-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Run audio at priorities 2 and 3 above kernel 1
Also run the watchdog at same priority as the fast mixer.
requestPriority() originally used only the caller's pid to decide which
cgroup to assign, but in the future it might look at the priority also.
So it's safer to use same priority as the fast mixer to be sure we
run in the same cgroup.
Bug: 6461925
Change-Id: Ia59c93e4b22dacbb6746bfa6ad491be7b72f2b8d
udioFlinger.cpp
|
| d96c5724818fb47917bb5e7abe37799735e1ec0e |
25-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
Don't call virtual methods in destructor
The result of calling virtual methods from a destructor is undefined.
Change-Id: I0fd4a19626e5ae564a60b753315b5f6c4b8d1f2c
udioFlinger.cpp
udioFlinger.h
|
| 1ea6d23396118a9cfe912b7b8a4e6f231e318ea2 |
09-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use atomic ops for thread suspend count
There was a theoretical but unlikely race if two binder threads
executed suspend() or restore() concurrently. Also added comments.
Change-Id: I0908acc810b83bdd66455b27ca3429de1662a2cd
udioFlinger.cpp
udioFlinger.h
|
| 1879fff068422852c1483dcf8365c2ff0e2fadfc |
12-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Add tid parameter to IAudioFlinger::openRecord
Not yet implemented
Change-Id: I35523fb15ad71727ecc9f4bb870f07e4b7397dc4
udioFlinger.cpp
udioFlinger.h
|
| bf04a5d7f287fc712e0ed91849dc85c90c1e182d |
12-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify AudioRecord::getInputFramesLost()
This also fixes a benign race in reading mActive without a lock.
Change-Id: I19e953d4f275e5c266ca1ca3fece7b6c02ad1707
udioFlinger.cpp
|
| 39c54f68804c1ce5c85ec588f3c2c63447a807b4 |
09-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dead code
Change-Id: If22a6c4e572b0734eba0c5a7ce29a2c61c581e5d
udioFlinger.cpp
udioFlinger.h
|
| 4fe1ec4f40b58abff6cec147aa786cb65698161a |
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Fix check for invalid channel count
Change-Id: Id9e3dce0e3d5971786212d3f70e17a17e32ce92b
udioFlinger.cpp
|
| 04270daf50f0c602d7c57a257a693e68246cbeb7 |
10-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Record overflow cleanup
Add comments and rename one method for clarity
Change-Id: I04a9147e46e88a072256c0211b112d52202419e2
udioFlinger.cpp
udioFlinger.h
|
| 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
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioPolicyService.cpp
udioPolicyService.h
|
| 4d2132c866228eaef563e8141ebf06aa25ee9ff7 |
10-Jul-2012 |
Eric Laurent <elaurent@google.com> |
Merge "audio policy: fix delayed command insertion"
|
| 2df8f55055fe431a508148cf525df1ba40f03113 |
10-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Add comments on use of volatile for track count
and add acquire load at the read
Change-Id: Ib41a58f5b1f6af87a8bd63d3f77d2ec0e48cb479
udioFlinger.h
|
| a34f8ec169986c5a28600c0decaa4e2db70df8e4 |
21-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Remove 'volatile' from mMute and add comments
Change-Id: I386ba27b2305a397aba70331c6bf0d35ea727cf6
udioFlinger.h
|
| 9f34a36d9cdb9595c288e50ffe00da038bc8abb9 |
21-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Whitespace and indentation
Change-Id: I424052b4ff9218147a5cfc8e6dcd67fe8105d229
udioFlinger.cpp
udioFlinger.h
|
| 7d6c35bf132a46c0a8a9826491882495fc98bd8c |
02-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Move constant initializations from onFirstRef() to constructor
Change-Id: I57f55b0bd1edee105c58c3a055f95f1e4a2c9646
udioFlinger.cpp
|
| 6648821933dc06c0b09ab2c8b32135edddcd4291 |
21-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger::getBuffer() always returns non-NULL
Change-Id: I543d3db507597cacbfdad5d9ea71732137fe54fb
udioFlinger.cpp
udioFlinger.h
|
| 94479fd5405642c67efd14cebe722feb9cbe6e77 |
10-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Fix build
Revert after system/core audio_devices_t is submitted
Change-Id: I5a8ee1a7b711e834501e927f41c62efa6a6600b6
udioFlinger.cpp
|
| cec4abb2a39eafa75cf6894170a200322aa7226e |
03-Jul-2012 |
Eric Laurent <elaurent@google.com> |
audio policy: fix delayed command insertion
When inserting a delayed command in the audio policy service
command thread queue, later similar commands are filtered out.
In case of volume commands and delayed unmuting, this can cause
a problem where a new volume command can cause the timing of a
delayed unmuting command to not be honored.
This change makes that a volume command gets the time stamp of
any delayed command it replaces.
Bug 6720482.
Change-Id: I216f85ac0ea46e4a046d3483e9e9b7ffe6c34a08
udioPolicyService.cpp
|
| 01542f2704f39956da09ae2840e192dab760091f |
02-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Only write to mDevice once
This fixes a bug where readers might see intermediate values.
Also add comments about how mStandby and mDevice are used.
Change-Id: Idc84e56c21381a45137a2ca5ff9c57d437201869
udioFlinger.cpp
udioFlinger.h
|
| c1dae24a08b67b98e18e4239d4f3a74d600d353c |
03-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Remove debug code HAVE_REQUEST_PRIORITY and SOAKER
Change-Id: I73a2afe72d8acb53e57e6b4e6fb5133e22b7875a
ndroid.mk
udioFlinger.cpp
udioFlinger.h
oaker.h
|
| a5f44ebaf58911805b4fb7fb479b19fd89d2e39b |
25-Jun-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix effect disconnect deadlock
Fix possible deadlock when several EffectHandles on the same
EffectModule are destroyed simultaneously:
A wp on an EffectHandle should not be promoted to a local sp
with ThreadBase mutex held as the EffectHandle destructor can be
called when the sp gets out of scope which will call
ThreadBase::disconnectEffect() and try to acquire the mutex.
Use raw pointers instead of weak pointers for the list of handles
on an EffectModule.
Bug 6679606.
Change-Id: Ice8b602fb03a7d363c44ce3dced8a53540d96270
udioFlinger.cpp
udioFlinger.h
|
| 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
udioFlinger.cpp
udioFlinger.h
|
| f1da96d8cf60842538e00a9c950cc451f7da2c10 |
03-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Remove longStandbyExit
It was never set (the assignment was within an "if" that was never true).
Change-Id: I01cc68e9df6b190eece621b2aa9858b4361880ce
udioFlinger.cpp
udioFlinger.h
|
| 415fa7599f48494f99206b8d6e1974abb52c5923 |
03-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Fix uninitialized field EffectModule::mPinned
Also mark EffectModule::mId and EffectModule::mSessionId const, and
document the initialization of other fields in EffectModule.
Change-Id: Ic1ca008e75e9b5924743ffc35bef80057f3a0669
udioFlinger.cpp
udioFlinger.h
|
| d5903ec1332630f2992a6f0d5ca69d13a185c665 |
18-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Compare sp<> to 0 and raw pointers to NULL
Change-Id: I50ff8a010d349d1d7e3dffa04a6331814c2128b0
udioFlinger.cpp
|
| a01992a0675a06df7d0bbe7b977207dd0c33fdc7 |
02-Jul-2012 |
Eric Laurent <elaurent@google.com> |
am dbbd5b86: am 109347d4: audioflinger: fix regression in attachAuxEffect().
* commit 'dbbd5b860a3a26bea3376410f75f27530d9cd10b':
audioflinger: fix regression in attachAuxEffect().
|
| 109347d421413303eb1678dd9e2aa9d40acf89d2 |
02-Jul-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix regression in attachAuxEffect().
Commit 717e1286 introduced a regression in PlaybackThread::Track::attachAuxEffect()
when called with an effect ID of 0 to detach the auxiliary effect.
It is normal in this case that AudioFlinger::getEffectThread_l() returns 0.
Bug 6768757.
Change-Id: I7430bd1aad2f68da38f7c3e4794e7ad657bfc6be
udioFlinger.cpp
|
| dbabf8a7dfe3aa8bf0ed169220d2009d5891fef2 |
01-Jul-2012 |
Eric Laurent <elaurent@google.com> |
am 651f9e7c: am 717e1286: audioflinger: fix auxiliary effect attachment
* commit '651f9e7c972b58a49066081187161268bcf9237a':
audioflinger: fix auxiliary effect attachment
|
| 717e128691f083a9469a1d0e363ac6ecd5c65d58 |
30-Jun-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix auxiliary effect attachment
Auxiliary effects (Reverb) are global effects and as such follow
the default rule which is to attach them to the output thread that
handles music streams by default. This causes a problem when several
threads are eligible to handle music streams as tracks can be attached
to either thread based on criteria unknown when teh effect is created.
The fix consists in moving the auxiliary effect if necessary when an
AudioTrack is attached to it and this track is not on the same
output thread.
Bug 6608561.
Change-Id: Ib32c3cabc731b2046aba728be1771982999c6069
udioFlinger.cpp
udioFlinger.h
|
| 9ee836fc8cf54d26e1e0819d1467d4d39bb9295c |
25-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove acoustics from AudioSystem::getInput()"
|
| 20010053daabfa43fcfe781bbf004473b4c08538 |
22-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Remove acoustics from AudioSystem::getInput()
Change-Id: I29fb3ee5664c1f0ee0409c1bb2be087ecca637db
udioPolicyService.cpp
udioPolicyService.h
|
| 22167855ff9af7b13fda669ca27c67a037a7d585 |
20-Jun-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix setStreamOutput()
AudioFlinger::setStreamOutput() should also work for direct outputs.
Also ignore the destination output specified to match the expected
behavior which is to invalidate all tracks using the specified stream
type so that they can be re created on the correct ouput thread.
Do not send STREAM_CONFIG_CHANGED event wich is ignored by AudioSystem
anyway since the stream to output cache has been removed.
Change-Id: I13d9d47922923b630dd755717875424c16be4637
udioFlinger.cpp
udioFlinger.h
|
| 5f12136299918ea30555f3481668ec0dbb775e5f |
15-Jun-2012 |
Eric Laurent <elaurent@google.com> |
audio policy: acquire lock for volume functions
Acquire the AudioPolicyService mutex when executing
setStreamVolumeIndex() and getStreamVolumeIndex().
These functions have become more complex and iterate on the
list of opened outputs and it is safer to acquire the
AudioPolicyService mutex to avoid concurrency with other
functions.
Bug 6626532.
Change-Id: Ifdcbbbf13338fd9a59f694d43faa7ade143f94bf
udioPolicyService.cpp
|
| eb15716b59020f342df62bce5b293f0603b94861 |
13-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Change definition of warmup period
Previously, warmup was considered done as soon as any write() took
more than 0.5 nominal cycle time. In practice, this was always the
first write() that turned on power to the output path, and it didn't
accurately account for filling the full kernel buffer queue, or for
buffering in the HAL sample rate conversion.
Now warmup is considered done when a write() _after_ the first write
takes more than 0.5 nominal cycle time.
This will throttle the initial pull rate after coming out of standby.
When combined with another change to throttle the pull rate for
devices with HAL sample rate conversion, it may help reduce some of the
notification glitches. The only downside is that it will increase the
warmup time a bit.
Bug: 6508317, 6650846, 6607056
Change-Id: I39f324c5195578170a55308e9601d3a1b41db3e0
astMixer.cpp
|
| 972af221ae7253e406e0e1ea853e56a3010ed6b1 |
14-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
FastMixer compensates for SRC jitter in audio HAL
Force a sleep if audio HAL write() returns too quickly.
Bug: 6661275
Change-Id: Ie810f5a087b89fb24dab229a0e500af1dee53571
astMixer.cpp
|
| 0a14c4ce1a41bc09eb7855fa531a3af629a69139 |
13-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Make CPU frequency statistics optional
Certain CPUs with dynamic cluster swapping and hotplug
don't report CPU frequency accurately. The file descriptors
used to read the frequency become stale and report bogus data.
So make this feature a build time option for debugging only.
This will also improve performance of the fast mixer loop.
Change-Id: I602f81ec3281a37992769208be08084ed1469e8c
ndroid.mk
astMixer.cpp
astMixer.h
|
| 362ebcbf100f7fccd37551c77e67c4faa7241b63 |
24-May-2012 |
Glenn Kasten <gkasten@google.com> |
DO NOT MERGE Remove log spam for fast track denied
Bug: 6531054
Change-Id: Iedf58e810a157aae88b5900da27c81054c437058
udioFlinger.cpp
|
| e6adde403480333b613452ae0df442df9da4a5b2 |
14-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Shorten dumpsys media.audio_flinger output" into jb-dev
|
| 49dd5cf3469cd755321f8ec10013ad4fcfaf723d |
14-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Log track name on obtain/releaseBuffer warnings" into jb-dev
|
| 3d51d5cb53cc630709a0ba78d0e60501a675f2d5 |
13-Jun-2012 |
James Dong <jdong@google.com> |
Add NOTICE and MODULE_LICENSE_APACH2 to libs build under /frameworks/av/
Change-Id: I0a3af3e2abdedebd5934f3d941d01c32cfc75e26
related-to-bug: 6647465
ODULE_LICENSE_APACHE2
OTICE
|
| 868c0abe9f26852c217bc201b337e72c68f829b8 |
13-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Shorten dumpsys media.audio_flinger output
Don't include in FastMixer if it's not present.
This removes confusing clutter especially for devices
with a separate deep buffer output thread, or for the
duplicating thread and A2DP output thread.
Change-Id: I84933f5555593256f11ba0895ec915c09cf16963
astMixer.cpp
|
| 0c9d26d187017f7fb028ab52a0fbc6395142faa4 |
31-May-2012 |
Glenn Kasten <gkasten@google.com> |
Log track name on obtain/releaseBuffer warnings
This should help diagnose problems by allowing us to correlate
the logs with the dumpsys media.audio_flinger output.
Change-Id: I8c7c592b4f87d13b0f29c66ce7a2f301a0f063c9
udioFlinger.cpp
|
| c15d6657a17d7cef91f800f40d11760e2e7340af |
30-May-2012 |
Glenn Kasten <gkasten@google.com> |
Add audio watchdog thread
Change-Id: I4ed62087bd6554179abb8258d2da606050e762c0
ndroid.mk
udioFlinger.cpp
udioFlinger.h
udioWatchdog.cpp
udioWatchdog.h
|
| 28ed2f93324988767b5658eba7c1fa781a275183 |
07-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Reduce underruns in screen off, esp. with EQ
Add MonoPipe APIs to specify setpoint.
Use screen state to configure pipe setpoint.
Fix a long-standing bug where pipe sleep time was excessive,
which interacted poorly with governor and low clock frequencies.
Now it deducts the elapsed time since last write(),
which was significant when there was EQ and low clock frequency.
Bug: 6618373
Change-Id: I6f3b0072c2244aeb033ef0795ad164491a164ff5
ndroid.mk
udioFlinger.cpp
udioFlinger.h
onoPipe.cpp
onoPipe.h
|
| a4f7e0e9a0e92a063f1b3a08988cf46e2cf1fa94 |
08-Jun-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix duplicating thread standby
The code that waits for a track presentation to be complete
before disabling it caused a regression for duplicating threads.
Because of the way output tracks activity is managed, the number
of frames output by the duplicating thread would never
reach the target set for a track to be considered presented.
The track would not be removed from active list and the thread would
not go to standby and keep its wakelock held.
Bug 6606922.
Change-Id: I4b46b420ac4cbf79a86b6791ae6589d407b01c92
udioFlinger.cpp
|
| e35a55fbdff5a695ee35ff43f1466a62fb75662b |
07-Jun-2012 |
Marco Nelissen <marcone@google.com> |
Merge "Take latency and current time into account for visualization" into jb-dev
|
| f06c2ed50e1db871ae9eb2bd15a196064f8c278c |
06-Jun-2012 |
Marco Nelissen <marcone@google.com> |
Take latency and current time into account for visualization
Buffer more data, and return the data that is currently being
output from the audio output, to ensure that visualizations are
smooth and responsive even when the audio output has a large
latency and/or large buffers.
b/6522728
Change-Id: I401637f01be7600b3c594a55c869036c13b206c0
udioFlinger.cpp
udioFlinger.h
|
| 9a0d82df8fd681257bc5b29fb5e6c65fcc7cef5a |
06-Jun-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: modify fast mixer overrun management
Do not force sleep when an overrun occurs. This was for non
blocking audio HAL and is unused.
Keep statistics on overruns.
Bug 6605215.
Change-Id: I295455d4562201aa3814c88bdd6e52c9226712af
astMixer.cpp
|
| 67c0a58e05f4c19d4a6f01fe6f06267d57b49305 |
02-May-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: various fixes on direct output
Various fixes in direct output playback thread implementation:
- threadLoop_write() was broken for playback threads that do not
use a pipe sink.
- output buffer size calculation was hard coded for stereo.
- removed software volume that was implemented for PCM stereo
format only: the audio HAL has to implement volume if needed
for direct outputs.
Change-Id: If211b4489be9af395435707b8cf0388cce1347b2
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
|
| ab9071b8d1b375418eb797c9a790da71de644344 |
04-Jun-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix active sleep time calculation.
When an audioTrack underruns, the audioflinger mixer thread
sleeps for a certain time to give the app a chance to recover.
This time is based on the reported audio HAL latency.
Some audio HALs implementing deep buffering have a variable
latency and this creates a problem if the sleep time is cached
when the output stream is opened and not updated afterwards.
This change derives the active sleep time from the mix buffer
framecount instead of the latency. This is more conservative
but works for variable latency.
Bug 6588525.
Change-Id: Ia892fc290fe06f836565c3ae15f7a2ce026c88c6
udioFlinger.cpp
|
| 399930859a75d806ce0ef124ac22025ae4ef0549 |
31-May-2012 |
Glenn Kasten <gkasten@google.com> |
State queue dump
Bug: 6591648
Change-Id: Iac75e5ea64e86640b3d890c46a636641b9733c6d
ndroid.mk
udioFlinger.cpp
udioFlinger.h
tateQueue.cpp
tateQueue.h
|
| 1295bb4dcff7b29c75cd23746816df12a871d72c |
31-May-2012 |
Glenn Kasten <gkasten@google.com> |
Fast track dumpsys
Bug: 6591648
Change-Id: I696f51c682e7233ba690d97da26012084989b412
udioFlinger.cpp
astMixer.cpp
astMixer.h
|
| bf0d21fb1310e8677caa53b90e8c3aecebc7fc13 |
31-May-2012 |
Glenn Kasten <gkasten@google.com> |
Count underruns for normal tracks also
Bug: 6591648
Change-Id: Iff9cabe392bb2ce97062603adb9c9dc7aa4170d5
udioFlinger.cpp
|
| 81787e56f8f21407eade5accc50c6da39b8af9b2 |
03-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Was calculating measured warmup time wrong" into jb-dev
|
| 7dc5b165d4f98a2059eaf1c7fec20b3b5773cd4c |
01-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix fast track leak if out of normal track names" into jb-dev
|
| 0c22272f1c97fd4b7e106030a1fd5e6b50680b61 |
01-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Pipe throttle based on requested frame count" into jb-dev
|
| 04a4ca4217e9b30342d5865024c340013d7184e8 |
01-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Was calculating measured warmup time wrong
Bug: 6592923
Change-Id: I980c6fc69d78c569147ea44ce33a7f8a41739c91
astMixer.cpp
|
| 893a05479c96f911d02beb0443da3ed6508143a7 |
30-May-2012 |
Glenn Kasten <gkasten@google.com> |
Fix fast track leak if out of normal track names
Bug: 6580402
Change-Id: I3ac7f012062c35833147f47ba822eb4bf532a824
udioFlinger.cpp
udioFlinger.h
|
| 820ba70df8ba595ae9055dfd34fdbfa32f70f14d |
31-May-2012 |
Glenn Kasten <gkasten@google.com> |
Pipe throttle based on requested frame count
Now does throttling based on the requested frame count, but
still permits overfilling pipe up to the rounded-up frame count.
Also still uses the larger frame count for non-blocking arithmetic.
Bug: 6585323
Change-Id: Ic2420a071419905524a14c22ebe1ce0e32d70a9d
onoPipe.cpp
onoPipe.h
|
| 91b14c4c144d0cc957a427cffc02ba10d0615677 |
30-May-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix effect problem during underrun
When an audio track underruns, the input buffer of the
corresponding effect chain (if any) must be cleared, otherwise
audio from previous mixer run will be fed again to the effect process
function.
Bug 6551652.
Change-Id: I5cd02196745f756c85af82d6937e9dc54369b37f
udioFlinger.cpp
udioFlinger.h
|
| f436fdcf93bd417fd3c9d2a8b19fd221d894b5e3 |
24-May-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: change session check in createTrack.
Do not refuse to create a track on an output thread if the same session
is present on another thread. It is now possible that two tracks
with the same session ID are on different threads if one can use deep
buffering and the other can't.
In this case, move effects attached to this session to the output
thread ion which the new track is created.
Bug 6530324.
Change-Id: I9019b3ee382e374c89d2319033afcfa7f886e4c4
udioFlinger.cpp
|
| e737cda649acbfa43fc1b74612a83f2fac9aa449 |
23-May-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: refine latency latency calculation.
There is an audio pipe between the normal mixer output and the fast
mixer to cope for scheduling delays and buffer size difference.
This pipe depth was not taken into account in latency calculation.
Adding the pipe contribution to the latency significantly improves A/V sync.
Bug 6520569.
Change-Id: I5584908e8aa8a02170eb38b22b4370eea800a235
udioFlinger.cpp
udioFlinger.h
onoPipe.h
|
| fbae5dae5187aca9d974cbe15ec818e9c6f56705 |
21-May-2012 |
Glenn Kasten <gkasten@google.com> |
Keep a copy of most recent audio played
Change-Id: I6b2f97881c39998a2fae9ab79d669af6c0a37e94
ndroid.mk
udioFlinger.cpp
udioFlinger.h
astMixer.cpp
astMixerState.cpp
astMixerState.h
ipeReader.cpp
|
| 6d8aabe8a3be1ac0789d00b82c3ca8b81381f5ab |
18-May-2012 |
Glenn Kasten <gkasten@google.com> |
Add throttle to have pipe fill at more stable rate
Change-Id: Ibff9ab62764e2acd04518a4a00e25f19e28aff4a
onoPipe.cpp
onoPipe.h
|
| 99c99d00beb43b939dedc9ffb07adb89f6a85ba5 |
15-May-2012 |
Glenn Kasten <gkasten@google.com> |
systrace for audio
Trace fast track buffer fill status for underruns etc.
Move the definition of macro to Android.mk.
No overhead if disabled.
Change-Id: If0e83e21b61b059ca38f543f8a6ffb58e08c79ee
ndroid.mk
udioFlinger.cpp
astMixer.cpp
|
| 552477608b137fb66af126655c6865207a82d888 |
18-May-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Display pipe underrun counters in dumpsys" into jb-dev
|
| 88cbea8a918bbaf5e06e48aadd5af5e81d58d232 |
15-May-2012 |
Glenn Kasten <gkasten@google.com> |
Display pipe underrun counters in dumpsys
The normal mixer writes it's submix to a pipe, which is read by the fast
mixer. Now dumpsys media.audio_flinger display the raw underrun counters
when fast mixer tries to pull from the pipe but doesn't get enough frames.
Change-Id: I72505f149f9e12802784da654a651d43734e1c79
udioFlinger.cpp
udioFlinger.h
|
| c059bd4246c1b3944965be921e5b334d51cd236c |
15-May-2012 |
Glenn Kasten <gkasten@google.com> |
Skip bad kHz values, and display them accurately
One device reports zero CPU kHz occasionally, so skip those values.
Preserve accuracy of the low-order 4 bits of the the CPU frequency.
Change-Id: I7d934f697412277462464fca82d112d0306a214e
astMixer.cpp
|
| 9017e5e0ebad9664bb7b6f2057e5bb29c852c64f |
15-May-2012 |
Glenn Kasten <gkasten@google.com> |
Increase normal mixer's pipe to fast mixer
Change-Id: I330925c7d07b6adb30b773bda3657e4efef9ae9b
udioFlinger.cpp
|
| 44a957f06400a338e7af20b3d16c4c4ae22a673c |
16-May-2012 |
Eric Laurent <elaurent@google.com> |
Fix static track activity ref counting
When a static AudioTrack underruns, it means that playback is over.
As apps do not necessarily stop playback explicitly, AudioFlinger
should call stopOutput() to decrease activity ref count in
audio policy manager.
Bug 6486311.
Change-Id: I1ea722c443780329ded6310c958b24726e918d16
udioFlinger.cpp
udioFlinger.h
|
| 2986460984580833161bdaabc7f17da1005a8961 |
09-May-2012 |
Eric Laurent <elaurent@google.com> |
Fix issues with synchronous record start.
- Added a timeout in case the trigger event is never fired.
- Extend AudioRecord obtainBuffer() timeout in case of
synchronous start to avoid spurious warning.
- Make sure that the event is triggered if the track is
destroyed.
- Reject event if the triggering track is in an incompatible state.
Also fix a problem when restoring a static AudioTrack after
a mediaserver crash.
Bug 6449468.
Change-Id: Ib36e11111fb88f73caa31dcb0622792737d57a4b
udioFlinger.cpp
udioFlinger.h
|
| 4adcede0dc54a85c31abaf139921aebd7a072d8e |
14-May-2012 |
Glenn Kasten <gkasten@google.com> |
Reduce video frame drop rate
The video playback engine depends on having relatively precise audio
progress updates for its A/V sync and frame drop calculations. For small
audio HAL buffer sizes, this was not a problem, but when the HAL buffer
size was > 12 ms, the normal mix buffer size became > 24 ms and this
then caused video problems. The new formula tries to keep the normal
mix buffer size within a closer tolerance of 20 ms to 24 ms.
Also use consistent term: multiplier instead of multiple.
Bug: 6479613
Change-Id: I903bad74461908e8c8f0a61e99ab5e24d5c44433
udioFlinger.cpp
|
| 09474df67278c0cd621b57c4aef1deaec4d8447f |
10-May-2012 |
Glenn Kasten <gkasten@google.com> |
Improve underrun handling for fast tracks
Maintain more accurate accounting of type of underrun.
Automatically remove track from active list after a series of "empty" underruns.
Change-Id: If042bf80e1790dcaaf195c99dc9c0ed9b55382c1
udioFlinger.cpp
udioFlinger.h
astMixer.cpp
astMixer.h
|
| d08f48c2ad2941d62b313007955c7145075d562c |
02-May-2012 |
Glenn Kasten <gkasten@google.com> |
Fix stopping process for fast tracks
Previously, the state of a fast track "wiggled" back and forth at the end.
Now it goes through these transitions:
active -> stopping_1 -> stopping_2 -> stopped
This CL is only for fast tracks, and does not change how
normal tracks work.
Change-Id: Icc414f2b48c46dda63cfa6373ca22d033dd21cd4
udioFlinger.cpp
udioFlinger.h
|
| 808e7d16504cbe5b28bb88c31afb2542ab488965 |
12-May-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix global effects suspend logic
Audio effects on the output mix should not be suspended when effects
on the output stage (post processing) are enabled.
Change-Id: I2e1c08fa9358ea3cbaec68856738d504b1be54e4
udioFlinger.cpp
|
| e58ccce45598bcf4b4874b0e87cd1eb8d05ba9a9 |
12-May-2012 |
Glenn Kasten <gkasten@google.com> |
Fix 2 bugs in fast mixer statistics
Was losing the upper half of the bounds, and MHz was off.
Change-Id: I61c98027b35ce7c3578ea6f3acf05aa5c48a5831
astMixer.cpp
|
| d8e6fd35ec2b59ee7d873daf1f1d9d348221c7bc |
07-May-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio tag for system tracing
Disabled by default; uncomment ATRACE_TAG to enable
Change-Id: I99af894022a859ee5644bd853cfd8a48e4735ff9
udioFlinger.cpp
astMixer.cpp
|
| 42d45cfd0c3d62357a6549c62f535e4d4fe08d91 |
02-May-2012 |
Glenn Kasten <gkasten@google.com> |
Update fast mixer statistics
Compute statistics on fast mixer elapsed time and CPU load per cycle using a
simple moving average rather than by fixed blocks. This has a couple advantages:
- remove burstiness of CPU usage due to former floating-point calculations in fast mixer
- gives us flexibility in how to report (e.g. could report over just the last 1 second)
Disadvantage is increased RAM, and since the samples are not updated
atomically, it is possible to have an error in the statistics. This
should not be much of an issue due to the relatively large number of samples.
Also add CPU raw ns and adjusted MHz statistics.
Change-Id: Iaa2cd13f18250c3162aff40409b3694b769d9505
astMixer.cpp
astMixer.h
|
| e213c86d36414a8fc75e37c52999522fe09c7328 |
25-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
dumpsys fCnt and flags
Previously displayed TrackBase::mFrameCount but not control block frameCount.
Now displays both.
Also display the track flags in control block.
Change-Id: Ie53781d4784633d78b6f928d69ebd494d8f110ec
udioFlinger.cpp
|
| 31dfd1db7a4d2228d9642008af6f3dd744368eb6 |
01-May-2012 |
Glenn Kasten <gkasten@google.com> |
Disable fast track log spam
except for "denied by client" and "denied by server"
Change-Id: I133ab747933729cc1f386813ee06ece055bdb294
udioFlinger.cpp
|
| 810280460da5000785662f6c5b0c7ff3ee0a4cb3 |
01-May-2012 |
Glenn Kasten <gkasten@google.com> |
Temporary fix for both normal tracks & fast tracks
If there is at least one active fast track, it forces a mixer
status of ready, which messes up the logic for normal track underruns.
Change-Id: I9de2fcaef090e2c2f99682333af3d3dd618b0d6b
udioFlinger.cpp
udioFlinger.h
|
| 288ed2103d96f3aabd7e6bea3c080ab6db164049 |
26-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
Fix race condition for non-started fast tracks
This required re-implementing how fast tracks are considered active.
Now, they use the same logic as normal tracks, except underrun is ignored.
Other changes:
- add framesReady() to AudioBufferProvider interface
- rebased
- add track underrun counter state to fast mixer dump state
- move dumpsys header to Track::appendDumpHeader()
so it closer to where tracks are dumped
- display track state in dumpsys as a character code
- measure and display warmup time and cycles in dumpsys
- copy in the presentation complete code
- add ExtendedAudioBufferProvider for framesReady() which returns size_t
- simplify underrun tracking
- deferred reset track after stop()
- add comments
Change-Id: I7db8821bc565230ec76da1f9380fe3fb09735e5b
udioFlinger.cpp
udioFlinger.h
xtendedAudioBufferProvider.h
astMixer.cpp
astMixer.h
astMixerState.h
ourceAudioBufferProvider.cpp
ourceAudioBufferProvider.h
|
| 83faee053cfd4251dbb591b62039f563ffdac399 |
28-Apr-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix stop detection for static tracks
The end of playback and end of presentation detection was broken for
static AudioTracks (tracks using shared memory buffers passed by client).
The mixer should not wait for a minimal amount of frames to be available to mix
a static track otherwise the last frames might never be consumed.
A static track should be removed from active list in case of underrun even if not
stopped().
Issue 6411521.
Change-Id: I66a2c1a77e98149e5049a223a6f04c3b8c5ad11a
udioFlinger.cpp
udioFlinger.h
|
| 300a2ee9327c05fbf9d3a5fd595b558097c7c5e8 |
25-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
Fast mixer configuration
Add compile-time option for when to use fast mixer.
Double HAL frame count for fast tracks due to SRC, and make the normal
frame count multiplier an even number for compatibility. Sample rate
conversion can result in underruns if the HAL frame count is used as is,
due to jitter.
Change-Id: Ia1f8da1b8ac247d9807acfce3c318161db000905
udioFlinger.cpp
|
| da747447c1d4b5205469b4e94485b8769df57a97 |
26-Apr-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix tracks ready for mixing logic.
Commit fec279f5 broke the logic allowing to wait for an application
to provide frames for mixing in the case of several active tracks.
This was causing audio gaps when playing music and superposing a
sound Fx (keyboard clicks...).
Issue 6185007.
Change-Id: Id0fad150d0b615646d6b1387c0de8ca944d228f6
udioFlinger.cpp
udioFlinger.h
|
| e0fa467e1150c65a7b1b1ed904c579b40f97c9df |
24-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
Move frame count calculations for fast tracks
For fast tracks: move the default and minimum frame count calculations
from client to server. If accepted, the default and minimum frame count
is the fast mixer (HAL) frame count. If denied, the default and minimum
frame count is the same as it currently is for normal tracks.
For normal tracks: there is no change yet, preserve legacy behavior for
now but add a FIXME to change this later.
Bug fix: the test for buffer alignment matches channelCount was wrong.
Bug fix: check for 8-bit data in shared memory, which isn't supported.
Optimizations:
- in set(), only call AudioSystem::getOutputSamplingRate() when needed
- in createTrack_l(), only call AudioSystem::getSamplingRate() and
AudioSystem::getFrameCount() when needed
Change-Id: I79d2fe507db1a8f7bb094c71da8a129951dbb82f
udioFlinger.cpp
|
| 1dc28b794587be22c90a97070d928f94586db638 |
24-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
Use scheduling policy service
Change-Id: I3c09da1dc0de5039d0c15ce7fb2bc373fa398712
ndroid.mk
udioFlinger.cpp
SchedulingPolicyService.cpp
SchedulingPolicyService.h
chedulingPolicyService.cpp
chedulingPolicyService.h
|
| 58912562617941964939a4182cda71eaeb153d4b |
03-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger normal mixer uses FastMixer
Change-Id: I3131bb22d2d057e9197a2ebfa6aa1cfaab9e5321
ndroid.mk
udioFlinger.cpp
udioFlinger.h
oaker.h
|
| 3acbd053c842e76e1a40fc8a0bf62de87eebf00f |
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Configure policy of mediaserver threads
Change-Id: Ifd825590ba36996064a458f64453a94b84722cb0
ndroid.mk
udioFlinger.cpp
udioFlinger.h
|
| c95cfbb87d0ac5e773037019a96bfc29972d4b4e |
12-Apr-2012 |
John Grossman <johngro@google.com> |
TimedAudioTrack: Optimize the queue trim operation.
Hand merge from ics-aah
> TimedAudioTrack: Optimize the queue trim operation.
>
> Don't perform the end PTS calculation for each buffer during trimming.
> Instead, only calculate the ending PTS of a buffer if there is no next
> buffer in the queue. This optimization assumes that the buffers being
> queued are in monotonic media time order (a fair assumption for now)
> and that the timestamps in the audio are contiguous (not a requirement
> for this API, but a reality of how it is being used right now).
>
> In the case where the audio is discontinuous on purpose, it is
> that this optimization will cause the system hold one extra buffer
> which it could have safely trimmed. It should not be much of an issue
> since in real life the audio is almost always contiguous, and as long
> as the media clock is running and the mixer is mixing, the buffer will
> be used up and discard as part of the normal flow anyway.
>
> Change-Id: I00061e85ee7d5651fcf80751646c7d7415894a14
> Signed-off-by: John Grossman <johngro@google.com>
Change-Id: I0054b58e1389fa005aa990cb5710caf4af7b706a
Signed-off-by: John Grossman <johngro@google.com>
udioFlinger.cpp
|
| 8d314b709fdd81bb64bdaa8d72a0b19c355cefb9 |
19-Apr-2012 |
John Grossman <johngro@google.com> |
TimedAudioTrack: Fix continuity threshold handling.
Hand merge from ics-aah
> TimedAudioTrack: Fix continuity threshold handling.
>
> Fix issues with continuity threshold handling; notably
> + If the steady-state continuity threshold is exceeded, be sure to
> clear the on-time flag. Failure to do this will result in the
> system picking a new mix point which simply satisfies the
> steady-state continuity threshold instead of the startup threshold.
> Since we are putting a discontinuity in presentation anyway, we
> really want to pick a perfect point, not just an OK point.
> + Tighten the steady-state continuity threshold. It was currently set
> to 100mSec which is enormous. 4mSec (the new setting) is much more
> appropriate. On systems with a VCXO (like tungsten) this should
> never be wrong by more than a sample. If TimedAudioTracks are ever
> to be used on VCXO-less systems, this threshold should probably be a
> a parameter configurable by applications on a track by track basis
> so they can make the tradeoff between allowed error and frequency of
> disruptive corrections.
> + Reset the on-time flag if the mixer provides no PTS during a mix
> operation. This makes for a convenient way for the HAL to reset
> timed tracks when it makes changes for delay compensation across
> multiple outputs.
>
> Change-Id: I2cb23de5a3d1f75618abc1c8ab903db883837aa8
> Signed-off-by: John Grossman <johngro@google.com>
Change-Id: Ibd28c9d290494b0b19eb01caf2d9bfdef606a9b4
Signed-off-by: John Grossman <johngro@google.com>
udioFlinger.cpp
|
| 21e8c50bd13ebe44f3088e26c9c6df0e163c469c |
12-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
FastMixer update
Updates:
- Add support for mono fast tracks
- Add support for optional sample rate conversion on fast tracks
- Log sample rate and frame count
- Enable statistics
Change-Id: Ife014edf4f452da361f3eaaae19209ef6ff6958b
ndroid.mk
astMixer.cpp
astMixer.h
astMixerState.cpp
astMixerState.h
|
| 9bd23229fdec1657398abc682ccccfce1c95f8aa |
16-Apr-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Fix multichannel downmix pause bug on video player
Pausing a video player will cause the track audio mixer to be
disabled, which causes the downmixer to be deleted. When reenabled,
the track channel mask hasn't changed but the downmixer is there
anymore.
Fixed by:
- instanciating a downmixer when the AudioMixer track
gets initialized (in getTrackName(), now taking a channel mask
as input), and deleted when in deleteTrackName().
- when the channel changes on a track, check whether it
needs a downmixer or not. Preparing a track for downmix
automatically removes the old downmixer if there was one.
Also: initialize the track downmixerBufferProvider field
when AudioMixer is instanciated, so we can safely call
delete on it in AudioMixer's destructor, in case
deleteTrackName() wasn't called before the mixer was
destroyed.
Change-Id: I589b0781cda5b3c82f85b561c52b08546cac21f8
udioFlinger.cpp
udioMixer.cpp
udioMixer.h
astMixer.cpp
|
| 0ca3cf94c0dfc173ad7886ae162c4b67067539f6 |
18-Apr-2012 |
Eric Laurent <elaurent@google.com> |
rename audio policy output flags
Change-Id: I27c46bd1d1b2b5f96b87af7d05b951fef18a1312
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| acb86cccbd9d245439a04cef0bcefa589addaa4c |
16-Apr-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Configure the resampler with the correct channel count when downmixing
When a track needs to be downmixed and resampled, it gets downmixed
first before being resampled. Therefore the resampler needs to
be configured with the channel count of the output of the downmixer
instead of that of the track.
Removed frame size checks that don't apply anymore now that we support
frame sizes that are not powers of 2 (e.g. 12 for 5.1 16bits), and
changed test performed for every buffer during playback into an
assert.
Change-Id: Ia220f00ee382f4f7848b661c58555bdca664e194
udioFlinger.cpp
udioMixer.cpp
|
| 97b5d0d5b5ef766eb5dd680d05a5d199662d4ae0 |
24-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Fast mixer
Change-Id: I61552f83507e08e4c706076b9fb15362869e6265
ndroid.mk
astMixer.cpp
astMixer.h
astMixerState.cpp
astMixerState.h
tateQueueInstantiations.cpp
|
| 0845682ff93c9765b74b53dc404d0954efc28c2e |
18-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Add template class StateQueue"
|
| dc998c809e084b617990b281e2ed5271830cc2e0 |
24-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Add template class StateQueue
Change-Id: Iccc5eb42bc295a22b2e429a4551f083cd7b6831a
ndroid.mk
tateQueue.cpp
tateQueue.h
|
| f7ffb8bf0a58037f0bc9662c5275005a4e539948 |
14-Apr-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: update to new audio HAL.
Updated audio flinger to new function prototypes
for open_output_stream() and open_input_stream().
Check audio hw device version when loading a module and
reject devices with a version different from current one.
Change-Id: I9d4c81a1f59a15db78b3989417c2a411c638fe88
udioFlinger.cpp
|
| 4e2293f29f2e719af1245d365747ea06d074b345 |
12-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
AudioMixer new cmd: remove sample rate converter
Add a new command REMOVE to remove any sample rate converter
on a track, without having to delete the track name.
Add comments.
Remove some dead code.
Fix whitespace and comparison to NULL.
Change-Id: Id55a23ab5ee673189e99675b4e6fa5df7b617987
udioMixer.cpp
udioMixer.h
|
| a5e821439996de6005b2fa36b3bdd31f003ce23f |
16-Apr-2012 |
Eric Laurent <elaurent@google.com> |
AudioMixer: fix crash in destructor
Commit deeb1282 made that track->resampler is only set to NULL when
getTrackName() is called. It remains uninitialized for all unused tracks
causing a crash when deleted by Audiomixer destructor.
Issue 6309441.
Change-Id: I49451387a6afe81299b8368ef206524df32f74c4
udioMixer.cpp
|
| b388e531613730572067e193b6b66afb5d042233 |
14-Apr-2012 |
Eric Laurent <elaurent@google.com> |
Fix ALOG_ASSERT in AudioFlinger
Fix broken ALOG_ASSERT in updateFramesPendingAfterTrim_l() introduced by
commit 1c345196.
Change-Id: Ie1b2653069283f23ff0367f2628828e37fb0749c
udioFlinger.cpp
|
| d3030da2ac3c0ebb8b7bdf38418263caf405b863 |
12-Apr-2012 |
John Grossman <johngro@google.com> |
Fix the build
forgot to upload final fixup during merge. sry about that
Change-Id: I2ddd2c08d8efa83c0a8d1e378ae4c28686145154
udioFlinger.cpp
|
| 1c345196edc61694f29307a1826a64a0d26028dc |
27-Mar-2012 |
John Grossman <johngro@google.com> |
TimedAudio: Track of the number of pending frames.
This is a manual merge from ics-aah
> TimedAudio: Track of the number of pending frames.
>
> Keep track of the number of frames pending in the timed audio queue so
> we can implement framesReady in O(1) time instead of O(N). This
> change partially addresses bug 6020970; the bug will be completely
> addressed once this change has been up-integrated into master.
>
> Change-Id: I599eb15ea1f6d715b97b30e65214fb6fadd169df
> Signed-off-by: John Grossman <johngro@google.com>
Change-Id: I6cbbbc3afc8efd066fe94865326ede0c6b3db2bd
Signed-off-by: John Grossman <johngro@google.com>
udioFlinger.cpp
udioFlinger.h
|
| 9fbdee13d09447550dd22ae72c2dbabdce7f0a80 |
27-Mar-2012 |
John Grossman <johngro@google.com> |
TimedAudio: Fix a cause of audio popping.
This is a manual merge from ics-aah
> TimedAudio: Fix a cause of audio popping.
>
> Fix an issue with buffer lifecycle management which could cause audio
> pops on timed outputs. There were two issues at work here.
>
> 1) During trim operations for the queued timed audio data, buffers
> were being trimmed based on their starting PTS instead of when the
> chunk of audio data actually ended. This means that if you have a
> very large chunk of audio data (larger than the mixer lead time),
> then a buffer at the head of the queue could be eligible to be
> trimmed before its data had been completely mixed into the output
> stream, even though the output stream was fully buffered and in no
> danger of underflow.
> 2) The implementation of getNextBuffer and releaseBuffer for timed
> audio tracks was not keeping anything like a reference to the data
> that it handed out to the mixer. The original architecture here
> seemed to be expecting a ring buffer design, but timed audio tracks
> use a packet based design. Pieces of packets are handed out to the
> mixer which then frequently will hold onto that chunk of data
> across two mix operations, using the first part of the chunk to
> finish a mix buffer and then using the end of the chunk for the
> start of the next mix buffer. If the buffer that the mixer is
> holding a piece of got trimmed before the start of the next mix
> operation, it would return to its heap and could be filled with who
> knows what by the time it actually got mixed. On debug builds,
> they seem to get zero'ed out as they go back to the heap causing
> obvious pops in presentation.
>
> This change addresses both issues. Trim operations are now based on
> ending presentation time for a chunk of audio, not the start. Also,
> when the head of the queue is in flight to the mixer, it can no longer
> be trimmed immediately, merely flagged for trim by the mixer when the
> mixer finally does call releaseBuffer.
>
> Signed-off-by: John Grossman <johngro@google.com>
> Change-Id: Ia1ba08cb9dea35a698723ab2d9bcbf804f1682fe
Change-Id: I2c5e2f0375c410f0de075886aac56ff6317b144c
Signed-off-by: John Grossman <johngro@google.com>
udioFlinger.cpp
udioFlinger.h
|
| 3f9c84c0a5af83fceb8669390e2d71b75ec7b550 |
04-Apr-2012 |
Eric Laurent <elaurent@google.com> |
audio pre processing: test code for dual mic
Added functional test code to validate effect API for
multi mic simplementations.
Also fixed warning in AudioFlinger.
Change-Id: I07be4d2e4d17791d3626c804ba3e9f87ff26d05a
udioFlinger.cpp
|
| 7d5b26230a179cd7bcc01f6578cd80d8c15a92a5 |
05-Apr-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
AudioMixer uses downmix effect for multichannel content
In the AudioMixer structure associated with each track, add an object
that acts as the buffer provider when the track has more than two
channels of input in the mixer. This object, DownmixerBufferProvider,
gets audio from the actual buffer provider of the track, and applies
a downmix effect on it.
The downmix effect is created and configured when the track gets
created in AudioFlinger, which causes AudioMixer::getTrackName()
to be called with the new track's channel mask. It is released
when the track is disabled in the mixer.
Change-Id: I05281ed5f61bef663a8af7ca7d5ceac3517c82db
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
|
| 3f502bfc6d04d08369cf147284c5c4c2ac7c9a0e |
06-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Non-blocking audio I/O interface, WIP"
|
| a4c5a550e2a3bc237179b8684e51718e05894492 |
29-Mar-2012 |
Eric Laurent <elaurent@google.com> |
audio policy: load audio hw modules.
Audio HW modules are now loaded upon request from audio policy manager
according to the configuration in audio_policy.conf.
Removed hard coded HW module loading by AudioFlinger at init time.
Added methods to IAudioFlinger and AudioPolicyInterface
to control the loading of audio HW modules.
Added methods to open an output or input stream on a specific hw module.
Change-Id: I361b294ece1a9b56b2fb39cc64259dbb73b804f4
udioFlinger.cpp
udioFlinger.h
udioPolicyService.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
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
|
| 73d227557ba5192735356bacab9f77b44980793b |
19-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger track flags and server's fast policy
Change-Id: I72358c8e6829d173b3e60ced8a8babc089869fac
udioFlinger.cpp
udioFlinger.h
|
| 0bf65bdde04b8e66c998ff37e2b2afafddddfa33 |
29-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
const methods and comments
Change-Id: Ifd16750174fdb15b72507787502b587562ffc99e
udioFlinger.cpp
udioFlinger.h
|
| 010662326b9c43c703725f933e95e0897f8a6bdd |
27-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Non-blocking audio I/O interface, WIP
Yet another abstraction similar to AudioTrack::Buffer and AudioBufferProvider,
but with support for streaming, non-blocking, and eventually PTS.
This is intended to be used as follows:
- primary HAL output stream implements a Sink
- primary HAL input stream implements a Source
- Pipe implements a Sink
- PipeReader implements a Source or TimedSource (not shown yet),
which supports "read at PTS"
- fast AudioTrack on server side will implement a Source using cblk
- normal AudioTrack on server side will not be changed initially
- fast AudioRecord on server side will implement a Sink using cblk
- normal AudioRecord on server side will not be changed initially
- fast mixer thread will read from Sources and write to a Sink,
or (unlikely) implement a Source and multiple Sinks
- Visualization and PCM logger will read from Source or TimedSource
- A2DP normal mixer will be connected directly to its output stream
and there will be a kind of OutputTrack for duplication that will
read from a Sink with non-blocking write fed by the fast mixer.
Patch set 3 changes:
- Add more implementations of NBAIO interfaces:
added SourceAudioBufferProvider, MonoPipe, MonoPipeReader.
- Added Format_sampleRate and Format_channelCount.
- Extract out the roundUp() method.
- Respond to most comments from previous code review.
- The new classes are untested.
Patch set 4 changes:
- Fix bugs in MonoPipe::write() and MonoPipeReader::read()
- Fix bug initializing mFrameBitShift too early
- renamed roundUp() to roundup()
- Fix Android.mk
- Add LOG_TAG an LOG_NDEBUG, use ALOG_ASSERT and utils/Log.h instead of assert
- Fix build warnings
- Move constructor and destructor bodies from .h to .cpp
- Line length 100
- Following naming conventions for #include double-include protector macros
- Include what you use
- More NBAIO logging
- MonoPipe write can be blocking
Patch set 5 changes:
- Address code review comments
- Use a static library so unused implementations don't take memory
- Comment out libsndfile dependency
- Remove debugging LOGV and LOG_NDEBUG
Patch set 6 changes (would be 6 at old location, actually 2 at new location):
- Address code review comments on patchset 5
- For MonoPipe, allow the full pipe to be used, no need to omit one slot
- Don't do atomic releasing stores unless needed
Still to do:
- I'm not happy with the Pipe class names
- Update build/ for new static library?
Change-Id: Ie6c61f05ce06b676b033be448a8ef9025a2ffcfd
ndroid.mk
udioBufferProviderSource.cpp
udioBufferProviderSource.h
udioStreamInSource.cpp
udioStreamInSource.h
udioStreamOutSink.cpp
udioStreamOutSink.h
ibsndfileSink.cpp
ibsndfileSink.h
ibsndfileSource.cpp
ibsndfileSource.h
onoPipe.cpp
onoPipe.h
onoPipeReader.cpp
onoPipeReader.h
BAIO.cpp
BAIO.h
ipe.cpp
ipe.h
ipeReader.cpp
ipeReader.h
ourceAudioBufferProvider.cpp
ourceAudioBufferProvider.h
oundup.c
oundup.h
|
| deeb1282621f3177ad667360b40eef8e4fedb298 |
25-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
AudioMixer default track parameters
Move the default initialization of track parameters from the AudioMixer
constructor to getTrackName(). This fixes a bug where the defaults were
only valid the first time a name was allocated.
Also added a switch case for parameter FORMAT.
Change-Id: I07abd3aaf7f4efe9825a761d2bc662aaee2c6db0
udioMixer.cpp
|
| a1472d9883e35edd280201c8be3191695007dfd4 |
30-Mar-2012 |
Marco Nelissen <marcone@google.com> |
Make AudioTrack/AudioRecord handle more than 2^32 frames
b/6160363
Change-Id: I471815012c6a113ec2c4dd7676e8fa288a70bc76
udioFlinger.cpp
|
| a011e35b22f95f558d81dc9c94b68b1465c4661d |
30-Mar-2012 |
Eric Laurent <elaurent@google.com> |
implemented synchronous audio capture
Added the infrastructure to support the synchronization of playback and
capture actions on specific events.
The first requirement for this feature is to synchronize the audio capture
start with the full rendering of a given audio content.
The applications can further be extended to other use cases
(synchronized playback start...) by adding new synchronization events and
new synchronous control methods on player or recorders.
Also added a method to query the audio session from a ToneGenerator.
Change-Id: I51f1167290d9cafdf2fbcdf9e4785156973af44c
udioFlinger.cpp
udioFlinger.h
|
| b83d38feeeb88a8a2a6219e1fca2480b5a14fb0d |
26-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "IAudioFlinger::createTrack and openRecord flags"
|
| d12c68ad699ce0ed822a4d4db86e8e02324c6b03 |
23-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Revert "AudioFlinger does not need libmedia any more"
This reverts commit c920dee060ac69684be33210ee44b99a5fc3e8b2
ndroid.mk
|
| 4f5da11df06802856bf526f16563df7d5755f653 |
22-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger does not need libmedia any more
Change-Id: Ifd2c61882109ec36ca68072a2bf6506e08c8cf34
ndroid.mk
|
| c5c49398584f2399af905a931e556ed6e0a29cd4 |
21-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Clean up Track constructor"
|
| ccf89b54f973f11082150d02ed957f7e967fbc8b |
21-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioMixer can be configured for fewer max tracks"
|
| 63c1faa8dea7feb90255d31ef2a133d8f2818844 |
21-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Update comments"
|
| 7153494670bdac8b650cb10b8b1838651e0ca418 |
21-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Whitespace"
|
| a26ff6f22f4e86d09514c2819237bd9748455018 |
21-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "new doesn't fail on Android"
|
| 9a5f413a21b320fea0607e653bc75b1a4f0e7a2e |
21-Mar-2012 |
Eric Laurent <elaurent@google.com> |
am a48285c4: am 165ee4c5: am 14958e21: Merge "audioflinger: fix issue with camcorder and A2DP" into ics-mr1
* commit 'a48285c4f22ffc43f1771ebd1ff35dcec48db2c7':
audioflinger: fix issue with camcorder and A2DP
|
| 5c94b6c7689a335e26a86e8a0d04b56dc627738e |
21-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
AudioMixer can be configured for fewer max tracks
Change-Id: I371b17cef071d083eecf35cd3627a3adff907a33
udioMixer.cpp
udioMixer.h
|
| f99590187e2e3f1cf6f093063170edec269cac5d |
19-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Clean up Track constructor
The 'thread' parameter can never be NULL.
Use constructor initialization list when possible.
Make more members const.
Only put the relevant code under "if (mCblk != NULL)".
Add comment about track name leak.
Change-Id: Ib963390a69bed1999638cc982a759edd1d5f4712
udioFlinger.cpp
udioFlinger.h
|
| ea7939a079b3600cab955760839b021326f8cfc3 |
14-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Whitespace
Fix indentation, and add blank lines in key places for clarity
Change-Id: I57a0a8142394f83203161aa9b8aa9276abf3ed7c
udioFlinger.cpp
udioMixer.cpp
|
| 2ee367e444e7b62e02bde8a2e47603a9ad342c6e |
20-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Add libmedia_native
Change-Id: I3ac357c78fb89f108d15c6e5b9fa317de0e9fb9a
ndroid.mk
|
| 17a736c3e1d062d7fc916329eb32aef8935614af |
14-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Update comments
Change-Id: I327663a020670d0a72ff57bd0b682e2ce0528650
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
|
| a03567676e8766828ff970b87e13bc4c97b23473 |
19-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
new doesn't fail on Android
Change-Id: I5079a3bf31097dd0807b2d806d5f8d3cff2077ab
udioFlinger.cpp
|
| a075db4ff9b086ac2885df77bb6da0869293df92 |
06-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
IAudioFlinger::createTrack and openRecord flags
createTrack and openRecord don't need the "old" flags parameter,
which was either audio_policy_output_t or audio_in_acoustics_t
shifted left by 16 bits. But they do need "new" flags, which
are defined by the application use case. Initially, the only
application use case flag is timed output, but others are planned.
For output, the audio_policy_output_t flags are passed to
AudioSystem::getOutput, which returns an audio_io_handle_t, and that
handle is then passed to createTrack. So createTrack doesn't need the
old flags parameter.
For input, the audio_in_acoustics_t flags are passed to
AudioSystem::getInput, which returns an audio_io_handle_t, and that
handle is then passed to openRecord. So openRecord doesn't need the
old flags parameter.
Change-Id: I18a9870911846cca69d420c19fe6a9face2fe8c4
udioFlinger.cpp
udioFlinger.h
|
| 9d7b4c074205609271f61e1a4741ac0c524a1795 |
19-Mar-2012 |
Eric Laurent <elaurent@google.com> |
am 14958e21: Merge "audioflinger: fix issue with camcorder and A2DP" into ics-mr1
* commit '14958e21c12f922d7501d32c3bec05109eb342d5':
audioflinger: fix issue with camcorder and A2DP
|
| 89d94e79dad032fb18ddc655e6068e4231d3f0aa |
17-Mar-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix issue with camcorder and A2DP
Some audio HALs do not support well a device selection of 0 (no device)
received on an input stream.
This can happen because of a problem in the audioflinger code that handles
the forwarding of the output device selection to the record thread for use by
the pre processing modules that need it. If the output device is 0 (meaning
no op, which happens when stopping playback over A2DP) audioflinger could not
detect it was an output device selection and would forward it to the input
stream (see AudioFlinger::setParameters() and RecordThread::checkForNewParameters_l().
Issue 6179641.
Change-Id: Idae534521866538e0d12ba259a2834f402a922e2
udioFlinger.cpp
|
| b12153e36ae5e514d96c77d660b327d1a8f5000f |
16-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Add libmedia_native
Change-Id: Ib8cff8abd73723b793f08da99ad59549f219e0e7
ndroid.mk
|
| 33b383948e8f270bff30378476f00dce289004eb |
13-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dependency on audio_* location
Change-Id: I4bc66115fcb9ba22b057bd72db3f561dcb18a0d8
ndroid.mk
|
| dfaf549e3e310bc22444f4404b19f4907b24c286 |
15-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioFlinger playback thread CPU measurement in Hz"
|
| 190a46f7c84e160386610c0c4cecb9767fb5503b |
06-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger playback thread CPU measurement in Hz
Log statistics on CPU usage in Hz in addition to wall clock time
Use CPU statistics for all playback threads, not just MIXER
(but they are disabled by default by a compile-time debug macro).
ThreadCpuUsage library:
- Move statistics out of the library and leave that up to the caller
- Add API to determine a CPU's frequency
Change-Id: Ia1011123146e641fcf210ef26e78ae2b4d3b64ad
udioFlinger.cpp
|
| 3b229ed97c0dfc85a8cf881341e29e595e0edea7 |
14-Mar-2012 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: more info in dumpsys"
|
| 612bbb57c59397a540e96f06bdd16e437a583af5 |
14-Mar-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: more info in dumpsys
Added TID and io handle to AudioFlinger threads dump.
Change-Id: Ib1a856f3bad55c73e4c395b5e59d57435f4b9a4c
udioFlinger.cpp
|
| d3cee2f0f649c01e1153d593cbe723887b8e0ba0 |
14-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Break circular dependency on media player service
Bug: 6165157
Change-Id: I3c85bbcaf31f3cb9a009e273f7b6284015eb3bd8
udioFlinger.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
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
udioResampler.cpp
udioResampler.h
udioResamplerCubic.h
udioResamplerSinc.cpp
|
| fd83fbf5ead098070bae674b20e6f87f45ab5d4c |
13-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Clean up assertion checks"
|
| aa4397f07c43bd83bc3100b749401dc3d15e7622 |
13-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Fix bug where mMixerStatus was set to IDLE
Change-Id: I55df1738fb7ba17ba6caeea6a17557526eac17a7
udioFlinger.cpp
|
| 5798d4ebf236357a4b13246f40e52b90a34d09a4 |
08-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Clean up assertion checks
Use ALOG_ASSERT instead of assert.
Use compile-time asserts where appropriate.
Fix typo in an ALOGV.
Change-Id: I58f1c1ffc14319a022c88b5a88b8d0368660da8b
udioFlinger.cpp
udioMixer.cpp
|
| 73f4bc33e0d458933460250a47c64aa868c05f97 |
09-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Inline applyVolume() into threadLoop_mix()
Also the declaration of applyVolume in PlaybackThread was dead.
Change-Id: I4b1a9848d07d3d7f340baea05b17f667c78df868
udioFlinger.cpp
udioFlinger.h
|
| 470aa50c36089fbe0427557f7cf4464dd26a1c52 |
12-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove unnecessary friend declarations"
|
| 2d6ef93773465cd3e66146fac35050a472c589f7 |
12-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove virtual from methods that don't need it"
|
| 66fcab972e9218d47c58a915f391b2f48a09903a |
24-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge dup code at thread entry and param change
This CL is mostly just cleanup, but there are a couple of fixes marked
"FIX" below.
Merge the duplicate code that was at the beginning of threadLoop() and
after a parameter change. cacheParameters_l() is now called at entry to
threadLoop() and after any parameter change. It re-calculates all values
that are derived from parameters, and caches them in instance variables.
updateWaitTime_l():
- FIX activeSleepTime depends on mWaitTimeMs, which was initially set
to infinity. updateWaitTime_l() was not called at entry to
threadLoop(), so activeSleepTime was not set correctly before the
first parameter change.
- FIX reversed the order of calls after parameter change
for the same reason so that updateWaitTime_l() is called before
calculating values that are derived from wait time.
- marked it private since now it's only called from DuplicatingThread
Change-Id: If2607d2ed66c6893d910433e48208a93c41fb7e9
udioFlinger.cpp
udioFlinger.h
|
| 18868c5db2f90309c6d11e5837822135e4a0c0fa |
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_policy_output_flags_t consistently
This affects:
- IAudioFlinger::openOutput
- AudioTrack::AudioTrack
- AudioTrack::set
- apps that call these
Change-Id: I26fb281bac6cb87593d17697bc9cb37a835af205
udioFlinger.cpp
udioFlinger.h
|
| d69549665d412f1f6ebad48ad8cd05133ada8728 |
09-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Replace hard-coded 3 by FCC_2 to simplify searches"
|
| 688aac7675f18bdd7bff13334759e20bc4e6c390 |
09-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Remove virtual from methods that don't need it
Change-Id: I30e17e61aae25b036436c0e270313c80c43e5f06
udioFlinger.h
|
| 1998661fdb6b0b5ae103e047e3d653c5da1b99e3 |
09-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Remove unnecessary friend declarations
Add comments to the remaining friends, so we know what is left if we
decide to remove them later.
Change-Id: I1de929257dc4700960f77902cda3d303177c72cf
udioFlinger.h
|
| 083c154162c88a9f63aeaa10a4b52dd454bda9ff |
09-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Don't ask policy manager about invalid stream type"
|
| b3dfcf483a96816fe2e949dd86fed26e52ad7404 |
09-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioPolicyService InputDesc minor cleanup"
|
| a19ffb656616feec70613ba67ddfe15a504a4e76 |
09-Mar-2012 |
Eric Laurent <elaurent@google.com> |
Merge "audio policy: use audio_devices_t when appropriate"
|
| 81872a2ce65653ae980621ba2907da56a83ba9a7 |
08-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
AudioPolicyService InputDesc minor cleanup
Add a non-default constructor to set the mSessionId, and make mSessionId const.
Remove explicit clear on mEffects - it is automatically cleared by the destructor.
AudioPolicyService::setPreProcessorEnabled:
- parameter is const *
- use an alias instead of making a Vector copy
Destructor doesn't need to be virtual since there are no subclasses.
Change-Id: Ibc3c3bea8259839430b1cf5356186c7d96f1082f
udioPolicyService.cpp
udioPolicyService.h
|
| 53d76dbe7c55821e89d9da02e7a563f7fb45de87 |
08-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Replace hard-coded 3 by FCC_2 to simplify searches
Change-Id: I92881d04e8378307f849fb343071a58d181a68b4
udioFlinger.cpp
udioFlinger.h
|
| 6374252107fd6539397598195ea6defd5870fafb |
08-Mar-2012 |
Eric Laurent <elaurent@google.com> |
audio policy: use audio_devices_t when appropriate
Change-Id: I1b3a5879e81c789fb53d356af3d3a1ee2dca955f
udioPolicyService.cpp
udioPolicyService.h
|
| fec279f5a0bfaa2a42e91ab6dfa0282baeee308b |
08-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Mixer status cleanup
Use mPrevMixerStatus for DirectOutputThread also.
Remove the MIXER_CONTINUE logic and use MIXER_IDLE instead.
Rename the field mixerStatus to mMixerStatus.
Rename local variable back to mixerStatus.
Change-Id: I0a8145fc856c6c5ff8b784b6176ef3c4d8eb7408
udioFlinger.cpp
udioFlinger.h
|
| b071e9bc248865ef87a339044c0c5cbabfac175c |
08-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Cleanup DirectOutputThread::mActiveTrack
Rename activeTrack to mActiveTrack.
Release the reference earlier, at the end of threadLoop_mix().
This allows the field to be made private and to
move the declaration from PlaybackThread to DirectOutputThread.
Change-Id: I02be7a254638f7d85e92aaf0002d20ca0092a5c3
udioFlinger.cpp
udioFlinger.h
|
| 639dbee79140956c43926344c23af765f6e0c9a5 |
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Don't ask policy manager about invalid stream type
Change-Id: If50fbff9d34045d1398984da48da7e6428a74491
udioFlinger.cpp
|
| b279312a9038b9c5b9b05b31b1b1db86f748efd8 |
08-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "IAudioFlingerClient::ioConfigChanged param2 const"
|
| fa26a859bacb597587a8f49acfc3127351fa688c |
06-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Isolate references to outputTracks/mOutputTracks
Move all references to DuplicatingThread::outputTracks and
DuplicatingThread::mOutputTracks from the common threadLoop() into
virtual methods. This allows them to be moved from PlaybackThread to
DuplicatingThread, and to be marked private.
Also use vector assignment to copy mOutputTracks to outputTracks.
Change-Id: Ieb1cf1ad36b8a65143e61e6c92a65fb43427e5e2
udioFlinger.cpp
udioFlinger.h
|
| a13ad6e84db128eadf23b154d3346f0bb473a5f7 |
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Make applyVolume private to DirectOutputThread"
|
| d4513b09123deebf8023b73a82d2d46d35806cea |
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Make applyVolume private to DirectOutputThread
Change-Id: I7ca4a59505857cbd106b6f274c66e9580dead271
udioFlinger.cpp
udioFlinger.h
|
| a3707a280177e934a1e0a15660d9176663b7fc17 |
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Add comments about sequence for setting parameters"
|
| 1465f0c3df04c3166155a852a6a5c10069c1fd0a |
06-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge the calls to prepareTracks_l
Change-Id: I1dd759581333e2908d980180d44db7bf5ed6591d
udioFlinger.cpp
udioFlinger.h
|
| 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
udioFlinger.cpp
udioFlinger.h
|
| f8edf68a1e39da273eafe8c85bdbedc26636c4ec |
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Rename fields of AudioSessionRef"
|
| cbc52bfb3b51c81c945b1e35990324bde892829a |
01-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Add comments about sequence for setting parameters
Change-Id: Iffa59a34c3c47bdc1d3234cdcb4d8ff99c102825
udioFlinger.h
|
| 92b8360fe9c3174dc0edaaab4b746d8a3d3f831f |
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix indentation for re-organized code"
|
| 438b036deaf4413503567a75a2861c000d21da5b |
06-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Rename updateWaitTime since a lock is held
Change-Id: I9bb978cbd0debf5b21676467060f72eebafea3e6
udioFlinger.cpp
udioFlinger.h
|
| 952eeb27682a9b2ddfa761f24b6eb5e18fe5d814 |
06-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Fix indentation for re-organized code
Change-Id: I63471cebdbd095b7ad4e481611b785f9b02c7941
udioFlinger.cpp
|
| 012ca6b4f69fb24385025c0e84b8f816525a3032 |
06-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Rename fields of AudioSessionRef
Change-Id: I9f2a66094135c4ac6bec2d3e9db3ac5fbf988ede
udioFlinger.cpp
udioFlinger.h
|
| 000f0e39b4d0c88441297a05ab5f8da6832c1640 |
02-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
threadLoop merge
Change-Id: Id8e6330ac6be76f9c2debba94f856de87e2d98f7
udioFlinger.cpp
udioFlinger.h
|
| 3030fcefcfcfc48789612659206b957cd0489a54 |
01-Mar-2012 |
Eric Laurent <elaurent@google.com> |
renamed audio policy output flag.
Renamed AUDIO_POLICY_OUTPUT_FLAG_INDIRECT to AUDIO_POLICY_OUTPUT_FLAG_NONE
which is more appropriate.
Change-Id: Ia14d60397df0f2dcd9bea0186400a09da35bc104
udioPolicyService.h
|
| e8286332f3817a8b7cc4cfd8f6450a3913533660 |
29-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Shorten thread names"
|
| 3e07470f3b122097cacfe5b85cdb1359279a2f33 |
29-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Prepare for threadLoop merge - active tracks
Continued work on making the copies of threadLoop more similar:
- Remove alias for mActiveTracks in MixerThread and DuplicatingThread.
- Pull in declaration of activeTrack in DirectOutputThread.
- Remove redundant parameter of prepareTracks_l().
- Comment prepareTracks_l().
Change-Id: If1087c1902b454acec01ddfdd9f055f0ca7abf04
udioFlinger.cpp
udioFlinger.h
|
| bc4b08ba67a0245e092aee8f08ba30ef22d421bf |
29-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Update AudioFlinger comments"
|
| 91cda8bdf8358a8b977a44e016b60a70bc1a5ee9 |
29-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Pull in declaration of effectChains to inner block"
|
| 73ca0f5837d5448f7a5eb159a09cd0ebe82b4de9 |
29-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Pull in declaration of effectChains to inner block
Change-Id: I09eacf72124942abd604132b9f4e774b1236fcf3
udioFlinger.cpp
|
| c455fe9727d361076b7cead3efdac2d32a1a1d6d |
29-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
mSuspend comments and usage
Emphasize that playbackthread::mSuspend is a counter, not a bool
Change-Id: I7188e56814e1c54dbc65e560f3627f138257d644
udioFlinger.cpp
udioFlinger.h
|
| 688a64030834ea2f52cc9765676ddf6aa34df767 |
29-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Mark similar and different sections in threadLoop
Most of these comments will be removed after the threadLoop merge.
Note: the trivial change in assignments to mixBufferSize, and the
comments about "tracks to remove" is to make them all identical.
Change-Id: I3b1a33a7f2cd12ad557a1986bb71f6171161974a
udioFlinger.cpp
|
| d805b718b1fd2d5407ef665c8d4bb42e63dc71a9 |
25-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Update AudioFlinger comments
Add comments to enum mixer_state
Note side-effect of lockEffectChains_l
Fix a typo
Change-Id: Ibd51678bac2193201cbcbe081ff5664046fbc494
udioFlinger.h
|
| 480b46802bef1371d5caa16ad5454fce04769c57 |
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Shorten thread names
prctl(PR_SET_NAME) limits to 15 characters. Before we had names like
"Binder Thread #" and the counter was cut off :-( Also remove redundant
"thread" at end of name; it's always a thread.
Change-Id: I1f99c2730ba0787ed9b59c15914356cddf698e2f
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
|
| a3873833d518e032138cf70188b6f33cd7acec3d |
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Unlock effect chains in the middle of two if's"
|
| 3e9c3a1d34960cd258f294d31135ab6bf76179d5 |
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify removeNotificationClient"
|
| a17c820c556fddf7ddd96b82b3e9874e340ffafd |
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioFlinger const methods and parameters"
|
| cfbd62616ab2b12f0fee603658f04e5827cc7f8f |
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix theoretical race condition in addOutputTrack"
|
| fadb2c73fce479205432652530663e1e90fd546c |
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioBufferProvider comments and cleanup"
|
| 843a12d146bd64642bf85a4e56c274246e3893a6 |
27-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix tracking of hardware state for dump"
|
| e628d515888baadba75442128678e747e930ed58 |
27-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Make threadLoop() logs identical"
|
| d3cee0b1f77baa4fb7a049eb757e9f5006890726 |
27-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Move declaration of mixerStatus to inner block"
|
| a3b09254d44cd8d66ec947abe547538c4cfeaa89 |
20-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify removeNotificationClient
No need to check for presence of item before removing
(but we do lose the log of the previous value).
Change-Id: I2838430824de5f257f2ee15db0c22b1920c67d08
udioFlinger.cpp
|
| 02fe1bf923bbe5789202dbd5810e2c04794562e6 |
25-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger const methods and parameters
Change-Id: I93ec28024005ed23aa141518092a012a4a7c44c5
udioFlinger.cpp
udioFlinger.h
|
| c0b52836d07f823732f0ff98ca5ca9d7f5730cb8 |
24-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Make threadLoop() logs identical
Change the wording of the logs in the various copies of threadLoop()
to be identical. This will make it easier to merge them soon.
Change-Id: Idfa181e437738712c784dc7f746cac79f83d2931
udioFlinger.cpp
|
| 5d4eeeaf76ebe177b43e87b2a9df5e55e39021f0 |
24-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Move declaration of mixerStatus to inner block
mixerStatus was being declared (and initialized) too early,
which also resulted in a duplicate initialization. Moved
the declaration into the block where it is actually used.
Change-Id: Ifdcfefe362a5efe3493dd616cdb44645c6f9aed5
udioFlinger.cpp
|
| 37d825e72a6c606553a745da1212590a425996d3 |
24-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Pull out duplicated copies of silent mode check
Also fix the error handling for the property_get.
This is part of preparation for the threadLoop() merge.
Change-Id: I6405190ea18146d1271575e1dfe9f279e8f36b17
udioFlinger.cpp
udioFlinger.h
|
| 04743e99e71c0da012508c7119f414027654ee94 |
24-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Unlock effect chains in the middle of two if's
As part of the upcoming threadLoop() merge, this CL makes it clearer
what are the similar and different parts before and after unlocking
effect chains.
In each threadLoop(), the old code was:
if (sleepTime == 0) {
// A
unlockEffectChains(effectChains);
// B
} else {
unlockEffectChains(effectChains);
// C
}
The new code is:
if (sleepTime == 0) {
// A
}
unlockEffectChains(effectChains);
if (sleepTime == 0) {
// B
} else {
// C
}
Also this is slightly slower by one "if", it has the advantage of making
it much more obvious about what is done before and after the unlock,
and also to see the similarities and differences among the various
copies of threadLoop().
Change-Id: I7bf4369d2dcb072573ec43b7e52c637f0097dc00
udioFlinger.cpp
|
| 5ce96d97feafc6989f6141bb2633eae3d87ddf28 |
24-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Pull CPU statistics code out of threadLoop()"
|
| b6b740629c9f11535086e744465bada03f26df11 |
24-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Fix theoretical race condition in addOutputTrack
This is not a real race, because addOutputTrack was only called in two
places, and in both places there could be no other threads referencing
the DuplicatingThread instance.
Those two places are:
- the DuplicatingThread constructor, which is of course safe
- openDuplicateOutput - this is safe because it's called immediately
after the new DuplicatingThread, and there are no sp<> either in the
constructor or here which could cause onFirstRef() to do Thread::run().
But for safety in case addOutputTrack is ever called somewhere else,
or there are sp<> created earlier, it is safer to take the thread lock.
Change-Id: I1502d014fa37ec5dbf4bf40d3e2884af311cd5e9
udioFlinger.cpp
|
| 83efdd0fc08cd5aedf50b45741a8a87be8dc4b41 |
24-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Pull CPU statistics code out of threadLoop()
This is to prepare for the threadLoop() merge
Change-Id: I118c7d5c6b011b5d5b95ec7d63fb03feb166a9cf
udioFlinger.cpp
|
| 01c4ebf6b794493898114a502ed36de13137f7e5 |
22-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
AudioBufferProvider comments and cleanup
Add comments about which methods implement the AudioBufferProvider interface.
Simplified the definition of kInvalidPts. <stdint.h> is very hard to work
with, there seems to be no way to use it reliably to get INT64_MAX without
having a separate source file, which is ugly because it means kInvalidPts
is not a compile-time constant. So I just deleted AudioBufferProvider.cpp
and used a hard-coded constant instead.
Added a default constructor for Buffer so that the fields aren't random
(especially .raw which is used to determine if the buffer is valid).
Make the pts for getNextBuffer default to kInvalidPTS so code that
doesn't need a pts doesn't have to specify a value.
Rename the parameter to AudioMixer::setBufferProvider to make it clearer.
Change-Id: I87e7290884d4ed975b019f62d1ab6ae2bc5065a5
ndroid.mk
udioBufferProvider.cpp
udioBufferProvider.h
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
|
| 8abf44d2f2bcd20a2835570efe89d89c19db426a |
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Fix tracking of hardware state for dump
At end of AudioFlinger::onFirstRef(), the hardware status was being left
in wrong state. It should be AUDIO_HW_IDLE but was AUDIO_HW_INIT.
mHardwareStatus was being set to AUDIO_HW_OUTPUT_OPEN too early, and so
a return would leave it in the wrong state until next hardware operation.
Take the hardware lock for dev->get_parameters, and update mHardwareStatus
before and after.
Keep hardware lock only for the duration of the dev->set_parameters.
Rename two constants in enum hardware_call_state to have the prefix
AUDIO_HW so they follow the naming conventions.
Add comments.
Change-Id: I6c7450b11f9b13adaeef9cec874333e478a58fc0
udioFlinger.cpp
udioFlinger.h
|
| 5cf034d92d901169ca6e36c90475f40715827fcd |
21-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Remove TrackBase::mFlags
The bit-field TrackBase::mFlags was supposed to have track-specific
flags in the upper 16 bits, and system flags in the lower 16 bits.
The upper 16 bits of mFlags were initialized in the TrackBase
constructor from the flags parameter of IAudioFlinger::createTrack()
and IAudioFlinger::openRecord(), and the lower 16 bits were cleared.
However, the upper 16 bits of mFlags were never acccessed again.
So really there are no track-specific flags. I left the flags
in the parameter list of createTrack() and openRecord() but made a
note that these should be removed eventually as they are dead.
This leaves only the one system flag "step server failed". I replaced
the bit-field mFlags by bool mStepServerFailed, which is simpler and
slightly faster.
Change-Id: I6650f5487be72791b4a67d73adcd10ffa04e2aa5
udioFlinger.cpp
udioFlinger.h
|
| d6fd85a157ce2054b2304e6d171fa87ae09c363d |
22-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Avoid wp<>::unsafe_get() with a few exceptions"
|
| 9eaa55756c5b245970447019250ce852f5189525 |
20-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Avoid wp<>::unsafe_get() with a few exceptions
Avoid using wp<>::unsafe_get() except in a log, and other specific cases
when it's known to be safe.
Use more specific subclass types for parameters to avoid down-casts.
When a constructor or method parameter is "this" of an object that is
currently being constructed, it's better to use a raw pointer rather
than either sp<> or wp<>.
Using the raw pointer is safe, provided either:
- it is "this" of an object being constructed (which has sp<> refcount of 0),
- or the caller already holds an sp<>
The raw pointer is simpler and faster, and it avoids the problem of the
sp<> reference count being incremented and then decremented to zero on
scope exit, which would cause the object's destructor to run while the
object is still being constructed.
Also removed some dead code per a review comment.
Change-Id: I7375f64da3aec11b928c33cb01faff186252ef5e
udioFlinger.cpp
udioFlinger.h
|
| f063b49e95c28d63a58215ebda892a5fee4204cc |
18-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Fix build warning
warning: pointer of type 'void *' used in arithmetic
warning: enumeral and non-enumeral type in conditional expression
Change-Id: I7b8d626a636145ef648e3b5d0e77068216dd012e
udioFlinger.cpp
|
| 3b81acab52b7140c1b8b20be2d67be3e221637e7 |
28-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Remove bit fields to improve performance
uint16_t enabled is (mostly) changed to bool in a separate CL
Change-Id: Ied9f8c034b2479cee9a8778cee7b8ff92ae75b7b
udioFlinger.cpp
udioMixer.cpp
udioMixer.h
|
| 1b094ee8f7fe7eca65bf3d2f983ba95eef6db93d |
17-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify code"
|
| a111792f1314479c649d1d44c30c2caf70c00c2a |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify code
Use DefaultKeyedVector::valueFor to avoid extra test
Make local variables as local as possible
No double parentheses
No typedef for single use
No parentheses around indirect function call
No AudioFlinger:: prefix when not needed
Remove unnecessary casts
Remove block with only one line
Saves 128 bytes
Change-Id: I3a87430eeb01b81e7b81a1c38f6fdd3274ec48f3
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
udioPolicyService.cpp
|
| 29dcfcd66d884801e9907d04e81d407ee770802c |
17-Feb-2012 |
Mike Lockwood <lockwood@google.com> |
Merge "Put a bandaid on a segfault in timed audio track handling."
|
| fe5b3ba4b332d5fc9aa4f453434329b9f38768c2 |
13-Feb-2012 |
John Grossman <johngro@google.com> |
Put a bandaid on a segfault in timed audio track handling.
Add a bandaid to prevent a segfault which can occur while handling
timed audio buffers. There is a deeper problem which should
eventually be addressed, but for now this fix should prevent any
crashing.
The deeper problem is as follows.
When the AudioFlinger mixer gets data to mix from an AudioTrack, it
ends up getting a structure filled out which points into an IMemory
region owned by the AudioTrack. Unfortunately, this structure is not
holding a refcount on the IMemory which it points into. If the
IMemory refcount hits 0 and the chunk of RAM is retuned to the binder
heap it came from, there can still be a Buffer object being held by
the AudioFlinger mixer which points into the region of memory which
was retuned to the binfer heap. If AF reads from this buffer, it
could read corrupt data (if the region of memory gets handed back out
to a writer), or it could segfault (if the heap has been freed and the
pages unmapped). Similar problems could happen if AF attempts to
write to the buffer, heap corruption in one case, segfaulting in the
other.
In the past, this has not been an issue for AF, because tracks
allocate a single IMemory (which serves as a ring buffer) and the
IMemory lives for as long as the track lives. As an artifact of the
way the code came out, the mixer cannot be holding a Buffer structure
pointing into the IMemory which used to be owned by a track if the
track no longer exists. Tracks cannot come into or out of existence
during a mix operation, which is the only thing which makes this safe.
TimedTracks work differently, however. Timed tracks each allocate a
small binder heap, and then hand out IMemory instances broken out of
this heap. The heap lives as long as the track, so the worst which
could happen here is that a TimedTrack's IMemory gets returned to the
heap while there is still a buffer structure in flight pointing into
the memory region, then the region gets handed out again and
overwritten by new data causing the mixer to mix the wrong audio. The
timing to cause this to happen is very difficult to encounter, and you
to generate the timing conditions required, you need to be in a pretty
bad failure state where audio is already breaking up and skipping, so
its unlikely that anyone would notice (which is why I'm band-aiding
the segfault and letting the deeper issue slide for now).
In general, however, it might be a good idea to revisit this buffering
design. On principal, if someone is going to hold pointers into a
refcounted object, they should be holding a ref on the object at the
same time. Failure to do this will usually lead to a situation where
there are corruption or segfault issues, or to a system where the
refcounted object's lifetime must be implicitly managed very carefully
in ways which are usually non-obvious and are easy to break by new
engineers on a project.
Change-Id: Ib391075395ed0ef46a03c37aa38a82d09e88abeb
udioFlinger.cpp
|
| 9fda4b87441fe17d90d8144639c9de6d9022c3c0 |
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Fixed possible heap corruption in EffectDesc
"EffectDesc *effect = new EffectDesc(*effects[i]);" was relying on the
default copy constructor for EffectDesc, but the default copy constructor
does a member-by-member copy. This works OK for mUuid, but a member
copy of mName and mParams shares pointers. This could result in heap
corruption later on due to a double free. Changed to add an explicit
copy constructor that does a deep copy of both mName and mParams.
A malloc() and strdup() were being freed by delete, but the correct
matching API for these is free(). Fortunately our current memory runtime
implementation ignores the difference. Changed to use free().
EffectDesc and InputSourceDesc member fields were being torn down by
the code that does delete. Changed to do the tear-down in ~EffectDesc()
and ~InputSourceDesc().
Added constructor EffectDesc() with name and UUID parameters, rather
than having caller fill in the object after construction.
Made ~EffectDesc() and ~InputSourceDesc() non-virtual to save memory,
since they have no subclasses.
Change-Id: Ibb5cc2e6760d72e0c4cf537068ac4432c717bafd
udioPolicyService.cpp
udioPolicyService.h
|
| ef7740be67a4d7b6b033ebed59c3d4a9c74a2c18 |
09-Feb-2012 |
John Grossman <johngro@google.com> |
Fix a segfault in AudioFlinger.
Check the string returned by a HAL's implementation of get_parameters
for NULL before attempting to make use of it. That way, we won't
bring down the mediaserver because of a poorly written HAL.
Change-Id: Ic99d7b004520d7d6347842a681c0595e889b68ea
Signed-off-by: John Grossman <johngro@google.com>
udioFlinger.cpp
|
| 4ff14bae91075eb274eb1c2975982358946e7e63 |
09-Feb-2012 |
John Grossman <johngro@google.com> |
Upintegrate Audio Flinger changes from ICS_AAH
Bring in changes to audio flinger made to support timed audio tracks
and HW master volume control.
Change-Id: Ide52d48809bdbed13acf35fd59b24637e35064ae
Signed-off-by: John Grossman <johngro@google.com>
ndroid.mk
udioBufferProvider.cpp
udioBufferProvider.h
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
udioResampler.cpp
udioResampler.h
udioResamplerCubic.cpp
udioResamplerSinc.cpp
|
| 6dad4378f2a78d967defc8912ecf47f6ed117584 |
14-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix races related to volume and mute"
|
| d9b9b8d09e7471b0ffa21cfa9f944ef4ad300a71 |
14-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Update comments"
|
| 99e53b86eebb605b70dd7591b89bf61a9414ed0e |
19-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Update comments
We no longer put the filename at start of file.
Change-Id: Ic435b159a23105681e3d4a6cb1ac097bc853302e
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
|
| 8d6a2449a91f5116d7243ab039393195ebd663fe |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Use size_t and ssize_t with Vector
Use size_t with size() and ssize_t with indexOfKey(). Exception:
use ssize_t for backwards loops, and indices that are overloaded as a
marker or error code.
Change-Id: Ibf2a360af4539b72b09c818dda22ea2a0de92431
udioFlinger.cpp
udioPolicyService.cpp
|
| 6dbc1359f778575d09d6da722b060a6d72c2e7c5 |
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
AudioRecord and AudioTrack client tid
Inform AudioFlinger of the tid of the callback thread.
Change-Id: I670df92dd06749b057238b48ed1094b13aab720b
udioFlinger.cpp
udioFlinger.h
|
| 44deb053252a3bd2f57a007ab9560f4924f62394 |
06-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Factor out and speed up permission-checking code
Use the caching permission check for dump to save IPC.
Cache getpid() to save kernel call for other permission checks.
The C runtime library getpid() can't cache due to a fork
race condition, but we know that mediaserver doesn't fork.
Don't construct String16 on the stack.
Change-Id: I6be6161dae5155d39ba6ed6228e7683e67be34ed
ndroid.mk
udioFlinger.cpp
udioPolicyService.cpp
erviceUtilities.cpp
erviceUtilities.h
|
| 2b213bc220768d2b984239511cd4554a96bc0079 |
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
mAudioHwDevs and related cleanup
Inline AudioFlinger::initCheck and remove unnecessary lock.
Remove redundant check of mAudioHwDevs.size().
No need to lock mHardwareLock for each device separately
during initialization.
Use size_t not int to loop through Vector, since size() returns size_t.
Add missing hardware lock for get_mic_mute() and get_input_buffer_size().
Add comments.
Change-Id: Iafae78ef78bbf65f703d99fcc27c2f4ff221aedc
udioFlinger.cpp
udioFlinger.h
|
| b6333aa8317ce5162ab006c4baed6b0890936dc7 |
11-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify ThreadBase::exit() aka requestExitAndWait"
|
| 3a144d08b713e3c0f6b7b8e95bc42cef5886f4fa |
11-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Disable HQ resamplers for now until qualified"
|
| 0d9302d7830b46542821b3e5f3e4f96942bd3cb3 |
11-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Move header declarations around for clarity"
|
| 858df80948ee64f478782a6a6c06533ba1651ef1 |
11-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Camel case readability & private disconnect(bool)"
|
| 95a87908ddf240760e8e8b35a6cafcc149c7f33f |
11-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove aliasing"
|
| c8ad36bbb30e99e49026cba78e5e0f83db5cb0f6 |
11-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use mul from audioutils"
|
| 31f0acff69efd46897221d847becca79ab15105f |
11-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Mark fields const if only set in constructor"
|
| b28686f95daee16edeb5f39af2cd5274ac3dc99f |
06-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify ThreadBase::exit() aka requestExitAndWait
We can remove mExiting and use Thread::exitPending() instead.
The local sp<> on "this" in exit() is not needed, since the caller must
also hold an sp<> in order to be calling us. (Unless it was using a raw
pointer, but that would be dangerous for other reasons.)
Add comment explaining the mLock in exit().
Change-Id: I319e5107533a1a7cdbd13c292685f3e2be60f6c4
udioFlinger.cpp
udioFlinger.h
|
| 9d1f02d74fd395ec4de6861147da289423f0ab6f |
09-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Follow raw pointer and sp<> conventions
Unconditional delete for raw pointers.
Use "if (sp != 0)" not "if (sp.get() != 0)" or "if (sp != NULL)".
Use "if (raw != NULL)" not "if (raw)".
Change-Id: I531a8da7c37149261ed2f34b862ec4896a4b785b
udioPolicyService.cpp
|
| 0ba18ec1b343a8de70924f87630dd1f329b00fe6 |
10-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "No newline or space at end of ALOG format string"
|
| 12018d80add66f5558675614d73fa6549150806e |
10-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Move declaration of stream_type_t up earlier"
|
| 8b5980798ca06e57b1284e6e23fa220e1207bf41 |
10-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix typos in ALOG for pid vs tid"
|
| cdf2158f3b9498d6cd0eb228d8bee16e32399e16 |
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Disable HQ resamplers for now until qualified
This saves about 6500 bytes.
Change-Id: I87102fe561c95c19c9e615dea3de914f96639257
ndroid.mk
udioResampler.cpp
|
| 2f732eb768004c6362fae8a02c60b69c9400b032 |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Move header declarations around for clarity
Put IAudioFlinger methods in binder opcode order.
Move hardware call state closer to where it is used.
getMode() and btNrecIsOff() are private.
Change-Id: Ie50340b396c39c763f2b155cbc08da8a0d0f2424
udioFlinger.h
|
| 004f719467c498942c40de9f260be601ee45e630 |
30-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Mark fields const if only set in constructor
Change-Id: Iacd06bb9efaf708cf965033be1f2297b58f7f75c
udioResampler.h
|
| d198b61603d5fa9298edea4ddb5852ea45159906 |
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Remove aliasing
Code was aliasing mBuffer as buffer, but continuing to use both buffer
and mBuffer after that point. This was at best misleading, and at worst
could confuse the compiler into generating bad code. There was no
performance advantage to the alias, in fact removing it saves 16 bytes.
Change-Id: I55023ddba465d9be82f66745b088d18af658ac60
udioResamplerSinc.cpp
|
| 58123c3a8b5f34f9d1f70264a3c568ed90288501 |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Camel case readability & private disconnect(bool)
Change-Id: If66516ed2703e048c5e6ccc6cd431446a024f4a1
udioFlinger.cpp
udioFlinger.h
|
| 5b9ff43995f6a6b819d9ad37dd8cdc5ad4a088d7 |
09-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Use mul from audioutils
I verified that the disassembled output is identical.
Change-Id: I34a76f0842ebc4aef2c923e079e38d0bc1f98b5c
udioFlinger.cpp
|
| 23d82a9bc9a43b49ba684ba40875b91db310d3b9 |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Fix typos in ALOG for pid vs tid
Change-Id: I6dc70f137d0ff8a86427ab8882a81886e1de0782
udioFlinger.cpp
udioPolicyService.cpp
|
| 90bebef5669a9385c706b042d146a31dca2e5d9b |
28-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
No newline or space at end of ALOG format string
Change-Id: I0bef580cbc818cb7c87aea23919d26f1446cec32
udioFlinger.cpp
udioResampler.cpp
udioResamplerCubic.cpp
|
| 6637baae4244aec731c4014da72418d330636ae1 |
09-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Fix races related to volume and mute
Fix race conditions when setting master volume, master mute, stream
volume, stream mute for a playback thread, and when reading stream
volume of a playback thread. Lock order is AudioFlinger, then thread.
Rename streamVolumeInternal to streamVolume_l, comment, and use it to
implement streamVolume().
Code size reduction:
- Remove dead code: AudioFlinger::PlaybackThread::masterVolume, masterMute, streamMute.
- Change return type of non-binder methods that always succeed from status_t to void.
- Remove virtual from volume and mute methods that don't need it.
This change saves 228 bytes but decreases performance of binder operations
due to the added locks.
Change-Id: Iac75abc1f54784873a667d1981b2e08f8f31e5c9
udioFlinger.cpp
udioFlinger.h
|
| b7bf796b758e144f94f6ed4d16c21bf22a118fb3 |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Move declaration of stream_type_t up earlier
stream_type_t is used by AudioFlinger class, so it should be declared there.
This way we don't have to peek into PlaybackThread to get the declaration.
Change-Id: Ie08bab1604699214d1e8df2d48d3fbfbbc436e96
udioFlinger.h
|
| 02bbd20cece1785c223ac4ca2ddc635931a80673 |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Rename type() to streamType()
This avoids possible confusion with thread's type().
Also remove redundant cast "(audio_stream_type_t)".
Change-Id: I320b9177b6c267a102d215f002228bcf988c437a
udioFlinger.cpp
udioFlinger.h
|
| 98ec94c5854daccc3474758524e7f4adfe535ce0 |
25-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Combine duplicate code & document wp<> in mClients
Change-Id: Iea8cfe8e57563337fb2484a1246ef79d6ad3db18
udioFlinger.cpp
udioFlinger.h
|
| 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
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| dbfafaffe2e97eaf8d74ec6b6c468418a1ad2443 |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify destructors
Remove explicit clear() when the order doesn't matter.
Change-Id: I5931bc7ef5f681c7ce329aa9ec0a6e46d34a56c5
udioFlinger.cpp
|
| 5e92a7861196ddae14638d4b7a63fc4892b7ef59 |
30-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Effect UUID inputs passed by pointer are const
Change-Id: I1f5c338bcb7368e3dd8cd5f804b2e6d9fbe087f8
udioFlinger.cpp
udioFlinger.h
|
| 0a20fa9c41c96e31fa20e071074a4b6e7f6c41c3 |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use pid_t not int"
|
| b61ec89bb0c701b3bd06eb658f854230681f8b39 |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Don't double destruct audio_track_cblk_t"
|
| 63d2daed17ab749baa80bc808fb5083b688b771b |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioFlinger methods const and inline"
|
| 86feafe15b3f9609e1e9f64184688c2b6f2e4834 |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use virtual destructors"
|
| e616d4e6de6d53ddebbc3d7fb381af94589c2232 |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Improve performance for sp<> on stack"
|
| 1579d7948117e3e6541b0cfda02cc5234a3280ea |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use 0 not NULL for sp<> and wp<>"
|
| b5a66e15f371143f48665bbd462d75309582845d |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use bool instead of int"
|
| e98bbd36d67243fe987b09904956550a68af1cc7 |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Declare more IAudioFlinger methods const"
|
| 1a0ae5be3d1273cba12584b33830d859510fbf82 |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Don't double destruct audio_track_cblk_t
Fortunately audio_track_cblk_t doesn't have a destructor, but for clarity
remove the double destruction.
Also add warning not to add any virtuals to audio_track_cblk_t.
Change-Id: I70ebe1a70460c7002145b2cdf10f9f137396e6f3
udioFlinger.cpp
|
| bb001926447d0b7dc71ca8bb3c9856f3136d8f4c |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Use pid_t not int
Change-Id: Iad1c2fd4152e94080ad8c65c13ddf4519fc2ed27
udioFlinger.cpp
|
| d5e54f7a36daedc3b2a642d1499c262da04e6280 |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dead code
mFormat is unused in resampler
mClientTid is unused
local variable pid is unused in dump
Change-Id: Ib156e38029366620bfeff2a13e73471867155a5b
udioFlinger.cpp
udioFlinger.h
udioResampler.cpp
udioResampler.h
|
| f587ba5b991c7cd91e4df093d0d796bd419e5d67 |
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Declare more IAudioFlinger methods const
This is just documentation, as C++ method const-ness doesn't mean anything
for a binder API. Instead, here const means "no side effects".
Change-Id: Iaa9cd2fe477db10ae9a40cac4f79f0faa9b4e5e6
udioFlinger.cpp
udioFlinger.h
|
| 4c340c6521b634f159d2d6bc4e9359226fd8edf8 |
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use bool instead of int
The .h is not modified to avoid increasing data size.
Change-Id: Ide4a821a5b16424ffa03471dfff98dc3e9b5f751
udioMixer.cpp
|
| c59c004a3a6042c0990d71179f88eee2ce781e3c |
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger methods const and inline
This saves 1063 bytes and probably improves performance.
Change-Id: I11cf0dfd925fbaec75e3d1b806852a538eae5518
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
udioResampler.h
|
| c19e22450e6e3d07594c935c7a9522e85e909e82 |
30-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use virtual destructors
It turns out to be just a comment, as all except AudioMixer are RefBase.
There are only a few performance-sensitive cases where it's worth thinking
about whether you need a virtual destructor, and the headache usually
outweighs the benefit.
Change-Id: I716292f9556ec17c29ce8c76ac8ae602cb496533
udioFlinger.h
udioMixer.h
udioResamplerSinc.h
|
| 7378ca506e4e20c2b2d4e94a131cf1b95831adb5 |
20-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use 0 not NULL for sp<> and wp<>
Change-Id: Id1f0c89acefaceed6cb9ca7c165fce895e46d85b
udioFlinger.cpp
udioPolicyService.cpp
|
| 787bae0578fbaab6219ebf23494866b224d01438 |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use audio_in_acoustics_t consistently"
|
| ff3c83e04ae459c4252b86dd4440b4e37ee92c16 |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use ToneGenerator::tone_type consistently"
|
| 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
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
|
| 87f155d6655b2d3b27e69281a29e85c6407e4d26 |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "For performance, return large objects by reference"
|
| f81e97e4ec8b01965a5b36987f886cf5001f71ff |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "No need to check a wp<> for 0 before promote()"
|
| 84afa3b51ac48f84ed62489529ce78cba7fca00e |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Constructor initialization and const fields
In constructors, initialize member fields in the initialization list
rather than constructor body where possible. This allows more fields
to be const, provided they are never modified.
Also initialize POD fields in constructor, unless it's obvious they
don't need to be initialized. In that case, put a comment instead.
Remove explicit clear() in destructors on fields that are now const.
Give AudioSessionRef a default constructor, so it's immutable fields can
be marked const.
Add comment about ~TrackBase() trick.
Initialize fields in declaration order to make it easier to confirm that
all fields are set.
Move initialization of mHardwareStatus from onFirstRef() to constructor.
Use NULL not 0 to initialize raw pointers in initialization list.
Rename field mClient to mAudioFlingerClient, and getter from client()
to audioFlingerClient().
Change-Id: Ib36cf6ed32f3cd19003f40a5d84046eb4c122052
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
|
| 5c0ad10b14ec2287f90f95912d98e66eef006e2a |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Cleanup thread types"
|
| 62da7fbd60bee2dd57f503126266e9f04311d400 |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Make AudioTrack control block volume field private"
|
| d45ee9d9d61af0791c7c3c51f8d4fe6794ef02a5 |
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix const sp<>& in parameter list and return value"
|
| d05397144be774f2f3623c754e865f51753e4e31 |
30-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
For performance, return large objects by reference
Change-Id: Ibf737018ef1d3c7d717584615dcb2d4ecdb50c99
udioFlinger.cpp
udioFlinger.h
|
| 090f01963e215f895020a31e22368cd44e086ce3 |
30-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Improve performance for sp<> on stack
Combine default constructor for sp<> immediately followed by assignment,
as the reference-counting is relatively slow. Also return sp<> directly
rather than via local variable, for the same reason.
Change-Id: If55931f1e407994f6591ddde41b53db72fb4fc40
udioFlinger.cpp
|
| 435dbe6c3ecd04bcb4bd80584064e287ebccd720 |
30-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Fix const sp<>& in parameter list and return value
EffectModule::addHandle and Client::heap() were declared incorrectly.
As a parameter, an sp<> should be & for efficiency, and for input
parameters it should also be const to protect the caller's value.
But as a return value, an sp<> should have neither const or &. The "e"
in "return e;" might be located on the stack, and if there is "&" then
the caller would see the address of a variable which no longer exists.
Also, an & would make it hard to do "return 0;".
A "const" without & is meaningless in the return type.
(In this particular case, the "e" is a member field, so it was safe.)
Change-Id: I3df5f294214eb15a9d4d596c6d5ef29de97b5c27
udioFlinger.cpp
udioFlinger.h
|
| e9dd0176933d6233916c84e18f3e8c0d644ca05d |
28-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Unconditional delete
Don't check that pointer is non-NULL before delete.
Don't leave deleted member fields non-NULL, except in a destructor,
since it could be misleading in a dump or debugger. (mRsmpOutBuffer)
Change-Id: Ic0492a6b752f74a67f4c96dfb89ca2de4e69eecf
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
|
| 77c1119ea0b5cb32287088ceeeb7e3b6bd14a85d |
25-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
No need to check a wp<> for 0 before promote()
Also remove unnecessary wp<> local variable.
Change-Id: I620e67b5d559d28616f8e00609a525cfe19c5ddc
udioFlinger.cpp
|
| 3d2f877c1cb4e4ae4ddde7f57f4353de9341f11b |
28-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use ToneGenerator::tone_type consistently
Also remove defaults in startToneCommand(), they're not needed and the
default for tone type was nonsense.
Change-Id: I70fa8cee4f3dbb8c66ceb3719c8d3d2f447f05b9
udioPolicyService.cpp
udioPolicyService.h
|
| de9719b3ec71472e6bf75117152176af51d1a515 |
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_in_acoustics_t consistently
Change-Id: I0a9dd668fb2e57b1c3ece3190588194974b99062
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| fd267d7cf640225bb57f0ed1af44efc153275f6d |
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use audio_source_t consistently"
|
| a3a2cd4072aaa2d93c91251a786eb7323f8d2c27 |
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioStreamIn and AudioStreamOut"
|
| 6f5980b75df837231365d238c1b0d6f386363fbb |
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Declare methods in binder opcode order"
|
| 28f52c84c22e129063a576e1269a39ae0cc0bfb3 |
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use enum effect_state consistently"
|
| 114c458f2b80a252ec627add1d5fda2093c79068 |
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use enum track_state consistently"
|
| 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
udioPolicyService.cpp
udioPolicyService.h
|
| aed850d0d3b3c8cf3feaf1438076f33db2a60946 |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
AudioStreamIn and AudioStreamOut
These are immutable, so make the fields const.
getOutput() and getInput() methods are now const.
Change-Id: I128246ebd56ea50b3e542be43f2aa1bcb55f1373
udioFlinger.cpp
udioFlinger.h
|
| 23bb8becff20449a9b1647d5a1a99b14c83f0cce |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Cleanup thread types
Use type_t instead of int for thread types.
Initialize ThreadBase::mType in constructor and make it const.
Change-Id: I43d141388b9639e4783c30b97dbda5688bf7555f
udioFlinger.cpp
udioFlinger.h
|
| 90716c5728b37637b2d0a730a721bfc9fad299e0 |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Declare methods in binder opcode order
This makes it easier to compare interface and implementation.
Change-Id: Ie060e43dec348902abcf40f5a610cec639d6d0d3
udioFlinger.cpp
udioFlinger.h
|
| 29c23c3aee5ae799b3480dc6876a46c46b019710 |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use enum mixer_state consistently
Change-Id: I5b71ed20f939dfc4b98143334b7aa064d282f584
udioFlinger.cpp
udioFlinger.h
|
| 28243dd563fee1c82f0fff6cc27b5cbf21fa2585 |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use enum effect_state consistently
Also fix indentation
Change-Id: I393ef9e37ffceed5ad4a78df439726ae1fe139df
udioFlinger.h
|
| b853e986caf43408ad95b9014f194aadff385e3c |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use enum track_state consistently
Change-Id: Ie5ebb7befa092e1de1e4df9c6e2d51e6bcfd176a
udioFlinger.cpp
udioFlinger.h
|
| 9365ea9bf2e439b3e71abbabe22ce7382ebc4b3a |
25-Jan-2012 |
Eric Laurent <elaurent@google.com> |
am 535b0264: am 7eeaf3f0: Merge "AudioFlinger: refine mixer sleep time logic" into ics-mr1
* commit '535b0264a4cfa790e549bd9cd09980788f1375f4':
AudioFlinger: refine mixer sleep time logic
|
| 84e19873fde204d73628ba1b5ca9e3f5778574fa |
24-Jan-2012 |
Eric Laurent <elaurent@google.com> |
am 7eeaf3f0: Merge "AudioFlinger: refine mixer sleep time logic" into ics-mr1
* commit '7eeaf3f07aa6fb10639d9f96c1367eb98c3e8839':
AudioFlinger: refine mixer sleep time logic
|
| 7c5aea0a8d9b422999483f96a2566f77ff11abf2 |
24-Jan-2012 |
Eric Laurent <elaurent@google.com> |
am 41773d46: Merge "DO NOT MERGE Revert "Revert "AudioFlinger: mix track only when really ready (2)""" into ics-mr1
* commit '41773d46556aa47d4322ff89fdaf7d1345c2d1f2':
DO NOT MERGE Revert "Revert "AudioFlinger: mix track only when really ready (2)""
|
| 21e4b6ed00e814bffc70895847a4944d7a190020 |
24-Jan-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: refine mixer sleep time logic
When an AudioTrack is in underrun state, the AudioFlinger mixer will
sleep for a short period of time to give the app a chance to fill the
AudioTrack buffer. If the AudioTrack is still not ready during next mixing round,
the mixer will proceed with other tracks.
If an application keeps a steady underrun condition, the AudioFlinger mixer will
alternate between ready and not ready states. In the longer term this will cause the
audio HAL to underrun.
There is a mechanism to reduce the sleep period if the mixer is not ready several times in a
row but this mechanism is defeated by the alternating ready/not ready conditions.
The fix consists in only increasing sleep time if the mixer is ready for at least two
consecutive times.
Issue 5904527.
Change-Id: Id0139bca9be8c4e425ec6d428515c4d8f718e8c9
udioFlinger.cpp
|
| eaa0b5cc2f7723e9b25298126d0dcb48c56d5dac |
24-Jan-2012 |
Eric Laurent <elaurent@google.com> |
DO NOT MERGE Revert "Revert "AudioFlinger: mix track only when really ready (2)""
This reverts commit b918035d34422a2041b6ec8c09c566bb93345b40.
Change-Id: I093bcfa56ad54a080b930208b6b79169d33581fb
udioFlinger.cpp
udioFlinger.h
|
| 15dfda272eec983508b89fb8bc9ca6f2bb825496 |
24-Jan-2012 |
Justin Ho <justinho@google.com> |
am fee5a860: Merge "DO NOT MERGE Revert "AudioFlinger: mix track only when really ready (2)"" into ics-mr1
* commit 'fee5a860a8355cda071ff23644e943414ba7f65d':
DO NOT MERGE Revert "AudioFlinger: mix track only when really ready (2)"
|
| 7baf7894bc2f1a62440f381eeb50143f210a5d61 |
23-Jan-2012 |
Justin Ho <justinho@google.com> |
DO NOT MERGE Revert "AudioFlinger: mix track only when really ready (2)"
This reverts commit 71c4496a9757438afd30b4404824f296f6158a49.
Change-Id: Iff10c49ea728bb10023ddeb50a3b708db770fff2
udioFlinger.cpp
udioFlinger.h
|
| 335787fe43596f38ea2fa50b24c54d0823a3fb1d |
21-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Remove AudioFlinger dependencies on client
Change-Id: Ibb591e41a3ca5d7015e2b66b98b8fef5f415fb37
udioFlinger.cpp
udioFlinger.h
|
| 83d86538c4c479a9225c75ab27938e8f05abb9c8 |
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Make AudioTrack control block volume field private
This is part of the process of abstracting the control block
to make it easier to maintain.
Change-Id: Idb8f461e68dab3bcf268159cc0781651c6fb7094
udioFlinger.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
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| 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
udioFlinger.cpp
|
| ea3cc3bca949139e401b77f2ac0cce7ac6e76f8f |
20-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove dead setRingerMode(mode, mask)"
|
| a3db30f7c53936a3acb87d2c44807ee2f0cfa16e |
20-Jan-2012 |
Jean-Baptiste Queru <jbq@google.com> |
am 9d25b82d: Merge "Rename LOG_ASSERT to ALOG_ASSERT"
* commit '9d25b82d280c2d979d500e7da4447148f32f820b':
Rename LOG_ASSERT to ALOG_ASSERT
|
| d746737921074e2a6c39c52b06022c5166689df5 |
20-Jan-2012 |
Jean-Baptiste Queru <jbq@google.com> |
am 6df477be: Merge "Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF)"
* commit '6df477be186233e36fc370c4d2db6c1ed928a740':
Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF)
|
| daef36f5d4934bd055c694a8d54b86e2b50a6159 |
20-Jan-2012 |
Jean-Baptiste Queru <jbq@google.com> |
am a826f9e2: Merge "Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF)"
* commit 'a826f9e2c4f6329d8d48c927f6e942e78ffaf92f':
Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF)
|
| 3734cbca25c6d902677cfb5e59dff7a1cb17a45d |
20-Jan-2012 |
Jean-Baptiste Queru <jbq@google.com> |
am 4f367f33: Merge "Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF)"
* commit '4f367f3387887c538c81c34cc8becaea6fa5e430':
Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF)
|
| e744a90eb52bf9547848c08380cfb7ba7e63ffbc |
20-Jan-2012 |
Jean-Baptiste Queru <jbq@google.com> |
am ba7f0d2a: Merge "Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF)"
* commit 'ba7f0d2a03643ce429421b81febf18fd50473070':
Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF)
|
| ca05a87f4c69670a06bdee4b4f98bcdcd838beda |
20-Jan-2012 |
Jean-Baptiste Queru <jbq@google.com> |
Merge ee4618bc
Change-Id: Ie1dc6ad38e7c30636d80f6caef11cf6673144940
|
| 39fd8eb956dea32692d3d72b10b64a8597df0a40 |
09-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename LOG_ASSERT to ALOG_ASSERT
Change-Id: Ie2c7ea6560656d65bad791a61996174c75677517
udioResampler.cpp
udioResamplerCubic.cpp
|
| 9a8ded7348c5b2302dd27b285b395416bc842c49 |
06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF)
Change-Id: I1de629b4632a4b3187ca1a28d6416daccd35f924
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
udioResampler.cpp
|
| aa70226152d2084f85a96b52359dbc8476a86a45 |
06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF)
Change-Id: I8fbdfa7a7581f481968dbb65aa40f7042936d7cb
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
udioResamplerCubic.cpp
|
| 53feeb42c721e8fc9285e35e679906a951f3277c |
04-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF)
Change-Id: I26f76452ac49e2890b14d133c065493d8df0fb4a
udioFlinger.cpp
udioPolicyService.cpp
|
| 52546c0ef96aa3e7e21482e0f9b6e982557c8da9 |
20-Dec-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF)
Change-Id: I44f267700356967dc51e8f85ebf457dc85cfb229
udioFlinger.cpp
udioMixer.cpp
udioResampler.cpp
|
| 3812256de32e73e38ba16e50ac0451c10223d4eb |
20-Oct-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF)
Change-Id: I5321ebd12e9c6248a108529e82c4e1af2a4405e3
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
udioResampler.cpp
|
| aeeb7e219e34d2d657d829913659a4e10e976375 |
19-Jan-2012 |
Eric Laurent <elaurent@google.com> |
resolved conflicts for merge of 05683c85 to master
Change-Id: I7846b7da8c5813b7a9b1f3f71aede0229689ff0d
|
| 86905f445c96743b037eed6360afa956a85b019c |
19-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Add units to time periods"
|
| ca42b286a162f5cd6a4a916bfe429abe95780730 |
19-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use const const strings for arrays and pointers to strings"
|
| 22ecc912a87099cff8cadc424cd12f85c118673f |
09-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Add units to time periods
Change-Id: Ib980e2676cecf5d9b0c2e388da6dc8e370df8abb
udioPolicyService.cpp
|
| 8dad0e31c2366c501bd1d600261d0af35a6ca786 |
09-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use const const strings for arrays and pointers to strings
Change-Id: I0a073e17ebf6ad59e7edbedac104c6fa4bcaf9c9
udioPolicyService.cpp
udioPolicyService.h
|
| f1d4592d4c3c99ebab55559e164ff102e825283e |
14-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
For booleans, use ! instead of == false
Change-Id: Ibc115936d2d0b0b7744ebe9b52839ea5b42c4edd
udioFlinger.cpp
udioPolicyService.cpp
|
| 0b07b8085d7b837b4dd5f09e0c8c39408f6bdbf7 |
18-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dead setRingerMode(mode, mask)
Change-Id: Ia4cc8be8424a40b3dcb7ebd0264fdff4e5247f7f
udioPolicyService.cpp
udioPolicyService.h
|
| 2774144fa8283f1a7b43e17a53c97dec0c366dd3 |
18-Jan-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: mix track only when really ready (2)
This problem due to the way audio buffers are mixed when
low power mode is active was addressed by commits 19ddf0eb
and 8a04fe03 but only partially. As a matter of fact, when more
than one audio track is playing, the problem is still present.
This is most noticeable when playing music with screen off
and a notification or navigation instruction is played: in this case,
the music or notification is likely to skip.
The fix consists in declaring the mixer ready if all active tracks
are ready. Previous behavior was to declare ready if at least one track was
ready. To avoid that one application failing to fill the track buffer blocks other
tracks indefinitely, this condition is respected only if the mixer was ready
in the previous round.
Issue 5799167.
Change-Id: Iabd4ca08d3d45f563d9824c8a03c2c68a43ae179
udioFlinger.cpp
udioFlinger.h
|
| 04eaf3e5f20d7faa956c0d672024ffb5117f4c26 |
18-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Track volume cleanup"
|
| 63ad6aacc6ce6b729bf25f41376cfea731a2c1eb |
18-Jan-2012 |
Eric Laurent <elaurent@google.com> |
Merge "audio framework: manage stream volume per device"
|
| b1cf75c4935001f61057989ee3cf27bbf09ecd9c |
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Track volume cleanup
Always read and write track volumes atomically. In most places this was
already being done, but there were a couple places where the left and
right channels were read independently.
Changed constant MAX_GAIN_INT to be a uint32_t instead of a float.
It is always used as a uint32_t in comparisons and assignments.
Use MAX_GAIN_INT in more places.
Now that volume is always accessed atomically, removed the union
and alias for uint16_t volume[2], and kept only volumeLR.
Removed volatile as it's meaningless.
In AudioFlinger, clamp the track volumes read from shared memory
before applying master and stream volume.
Change-Id: If65e2b27e5bc3db5bf75540479843041b58433f0
udioFlinger.cpp
|
| 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
udioPolicyService.cpp
udioPolicyService.h
|
| ad0f6cc5e115ca167ff122c83451b46d85c590ac |
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dead setVolume() and mVolume[2]
Change-Id: I94b835434093e920432614eb5007101e87758f32
udioFlinger.cpp
udioFlinger.h
|
| 0696400a6bb9abbed62b3b9c6aa105495dc600a2 |
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use audio_mode_t consistently"
|
| 263709e7be37c7040aaef385bc5c9389a9b5f514 |
06-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Check stream type in AudioFlinger::createTrack
A bad parameter to AudioFlinger::createTrack could cause mediaserver to crash.
Other AudioFlinger stream type cleanup:
- Simplify range check for audio_stream_type_t
- Add comment about mStreamTypes array initialization.
Change-Id: Ia33aa1cce0fdd694b08d9288816ffc097a9543d0
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
|
| 3944e0326a286bcb931551e61e79c033b10d09d4 |
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix locking for mMasterVolume and mMute"
|
| 613882293184e575a44bff681a3decaefe889e69 |
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use size_t for frame size"
|
| 0107954f72153db747a3727dc1157e9236dfed90 |
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use audio_stream_type_t consistently"
|
| 9806710f5d6722cfc5783c7eca3512451a0f2035 |
13-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Fix locking for mMasterVolume and mMute
mMasterVolume and mMute are both protected by mutex in AudioFlinger class, but
there were two places where they were accessed without a mutex.
Also make AudioFlinger::mMasterMute private not protected.
Change-Id: Ia3897daeb5c50313df5bcc071824357526237f3e
udioFlinger.cpp
udioFlinger.h
|
| 05632a5fa4b88ca474294887fc92a9fcdf0e2352 |
03-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
AudioTrack and AudioFlinger send level cleanup
Add an API to control block for getting/setting send level.
This allow us to make the mSendLevel field private.
Document the lack of barriers.
Use 0.0f to initialize floating-point values (for doc only).
Change-Id: I59f83b00adeb89eeee227e7648625d9a835be7a4
udioFlinger.cpp
|
| b9980659501d0428d65d8292f3c32da69d37fbd2 |
11-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use size_t for frame size
except in the control block, where we don't have room.
In AudioFlinger::ThreadBase::TrackBase::getBuffer,
read the frame size from control block only once.
Change-Id: Id6c4bccd4ed3e07d91df6bbea43bae45524f9f4e
udioFlinger.cpp
udioFlinger.h
|
| 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
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| 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
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| e3a067f8bc98134941ee1a4da8c2a92a15aaa9cc |
11-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify range check for audio_mode_t"
|
| c40256146bee58bff09e1c16ef99ea06d31f89f9 |
11-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use correct type for hardware call state"
|
| 2ea3410d0d3d592ce30c3ba0ce3e0e63b1244057 |
11-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use consistent style of & reference for AutoMutex"
|
| 9770988e61961d34033fd2c12f0de85a267df68f |
11-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioFlinger new can't fail"
|
| 02b2d2dddc9be339cf7647d4ead53cecd1c51038 |
11-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix build warning"
|
| 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
udioFlinger.cpp
udioPolicyService.cpp
|
| c1dc1cb1d1eaf84e88669f1a5f22579a0d9237c2 |
09-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename LOG_ASSERT to ALOG_ASSERT DO NOT MERGE
See https://android-git.corp.google.com/g/157519
Bug: 5449033
Change-Id: I8ceb2dba1b031a0fd68d15d146960d9ced62bbf3
udioFlinger.cpp
udioResampler.cpp
udioResamplerCubic.cpp
|
| 5f29ca38b71506ad7c7cb9925efbddf588e9655b |
09-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "By convention const goes before the type specifier"
|
| 25b248eb52a0a16adaef6b79c9d92cb88b9a2bc2 |
04-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use cached reference to media.player service
This save unnecessary binder calls
Change-Id: I93a60efc54d9c8fb8fab706cd4477bbfd00ffec8
udioFlinger.cpp
|
| 879bc64e43d62f2690c77a3282cef2e45d1b76e9 |
09-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Replace loop by __builtin_ctz"
|
| febdbfec3b1ed0e20aa4f10bfdd82702d3e41f4b |
09-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "suspended() and isSuspended() are const"
|
| 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
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
udioResampler.cpp
|
| 4a6f02833d8421b7d9b20f89729d8bb03b8b8102 |
07-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger new can't fail
Change-Id: I7dae05a5ea1c962a9975386eab1fedbbe106ffba
udioFlinger.cpp
|
| a7d8d6fc5e102a08b262a1b78aa1abeeb097d9e4 |
06-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use consistent style of & reference for AutoMutex
AutoMutex, which is a typedef for Mutex::Autolock, is overloaded for
either a reference (&) or pointer (*) parameter, but we prefer to use
the reference form when the mutex is known at compile time.
Change-Id: I3515e6d6ab7959b2356a27fa3b04fd49e42cb31e
udioFlinger.cpp
|
| a4454b4765c5905f14186893b0688be375642283 |
04-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use correct type for hardware call state
Change-Id: Ic6d98b129e3ec653df1d8f7e829adf8dccb4f378
udioFlinger.cpp
udioFlinger.h
|
| 54c3b66444ebfb9f2265ee70ac3b76ccefa0506a |
06-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
By convention const goes before the type specifier
Change-Id: I70203abd6a6f54e5bd9f1412800cc01212157e58
udioFlinger.h
udioMixer.cpp
udioMixer.h
udioResamplerSinc.cpp
udioResamplerSinc.h
|
| f9a27779634ce3a01e5957f234cd04eba74fa07f |
06-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Fix build warning
Change-Id: Ic99608d0c14ed56c02f036e0bbaaae1b16bab8ba
udioMixer.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
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
udioResamplerCubic.cpp
|
| 98dd542ffcd3424b1da58bd273c0c1ff5b8c8960 |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Replace loop by __builtin_ctz
Using the builtin is faster on some platforms, for example on ARM it's
19 instructions instead of 13, and is O(1) instead of O(n). Of course,
track creation is an inherently slow operation, so this doesn't matter
much now. But if we add support for virtual tracks, then physical tracks
will be allocated/freed more frequently. Also just on principle ...
Change-Id: I3f590934092bd7a1869cbedbc7357928aa5cc8ff
udioMixer.cpp
|
| 8a08dcc0a5de19a904e77d5f31bed3dff9a59890 |
05-Jan-2012 |
Steve Block <steveblock@google.com> |
Merge "Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGE"
|
| a3a854868a80fd9b9b8720e06a172754943f9417 |
04-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
suspended() and isSuspended() are const
Change-Id: I04b95970b5a645b64e7e64fffd46d868354dda66
udioFlinger.cpp
udioFlinger.h
|
| 88592eccaf6afcddd5f985955be92fe25205c680 |
05-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove the notion of "active track" from mixer"
|
| a2a0a5d7d56baa831870f4bf2a0d942a477d92ef |
05-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use the standard CC_LIKELY and CC_UNLIKELY macros"
|
| f6b1678f8f508b447155a81b44e214475ab634a8 |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Use the standard CC_LIKELY and CC_UNLIKELY macros
Several source files privately defined macros LIKELY and UNLIKELY in terms
of __builtin_expect. But <cutils/compiler.h> already has CC_LIKELY and
CC_UNLIKELY which are intended for this purpose. So rename the private
uses to use the standard names.
In addition, AudioFlinger was relying on the macro expanding to extra ( ).
Change-Id: I2494e087a0c0cac0ac998335f5e9c8ad02955873
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
|
| 7ab41c9f773ba599646f1b0d00955c1be80f92fd |
05-Jan-2012 |
Eric Laurent <elaurent@google.com> |
resolved conflicts for merge of 1a4b9939 to master
Change-Id: I0c910d391a38a916d8431f7d1f5b82e39e1a66c2
|
| df64d15042bbd5e0e4933ac49bf3c177dd94752c |
04-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/156801
Bug: 5449033
Change-Id: Ib08fe86d23db91ee153e9f91a99a35c42b9208ea
udioFlinger.cpp
udioPolicyService.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
udioFlinger.cpp
udioMixer.cpp
udioResampler.cpp
|
| 071ccd5a9702500f3f7d62ef881300914926184d |
23-Dec-2011 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix clicks on 48kHz audio.
The calculation done in prepareTracks_l() for the minimum amount
off frames needed to mix one output buffer had 2 issues:
- the additional sample needed for interpolation was not included
- the fact that the resampler does not acknowledge the frames consumed
immediately after each mixing round but only once all frames requested have been used
was not taken into account.
Thus the number of frames available in track buffer could be considered sufficient although
it was not and the resampler would abort producing a short silence perceived as a click.
Issue 5727099.
Change-Id: I7419847a7474c7d9f9170bedd0a636132262142c
udioFlinger.cpp
udioMixer.cpp
udioMixer.h
udioResampler.h
|
| 9c56d4ae6212c21ce5fd71ed534eb195983a07c1 |
20-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Remove the notion of "active track" from mixer
This is a first step towards making the mixer more object-oriented.
Change-Id: Ifd445d0e471023a7f5c82e934736ffc95ba1b05b
udioFlinger.cpp
udioMixer.cpp
udioMixer.h
|
| 12d9702773c8a3242a44dd6528126c9ff0dade43 |
20-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove dead code"
|
| bbaf8673f1d1dd79d1b7f474ca7111da58e84aff |
20-Dec-2011 |
Eric Laurent <elaurent@google.com> |
Merge "audio effects: rename configure command"
|
| 3d5188bd6abe55898f10a0edf3c05aff8aa2ef67 |
17-Dec-2011 |
Eric Laurent <elaurent@google.com> |
audio effects: rename configure command
Renamed audio effect library interface command for audio format
configuration from EFFECT_CMD_CONFIGURE to EFFECT_CMD_SET_CONFIG.
This makes the naming more consistent with other exixsting commands
and allow adding a new command to get the configuration (EFFECT_CMD_GET_CONFIG).
Same change for reverse channel configuration renamed from
EFFECT_CMD_CONFIGURE_REVERSE to EFFECT_CMD_SET_CONFIG_REVERSE.
Implemented EFFECT_CMD_GET_CONFIG in exisitng effect libraries.
Change-Id: Ia7b1c620f13797fe5aceb3b0b4acbacce09fb067
udioFlinger.cpp
|
| 81a028fef62bcadf13fc8550067a3d29c918b3ca |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Remove dead code
Change-Id: Icf23f7f90fdeb660f4015f22cf239e6d05f5d03c
udioMixer.cpp
udioMixer.h
|
| f602a5f7f29e694a7da4c143737b74501c00477d |
17-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Use constants for 2 and 32"
|
| d1e672acd8fa1af899f85ee2321327237028adf8 |
17-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Extract out audio DSP code to utility library"
|
| bf71f1e7948406492376c6cbd5e6a30c8cb670e4 |
13-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Use constants for 2 and 32
Change-Id: If820dfd58b6df258570750610a07af99598d9e53
udioMixer.cpp
udioMixer.h
|
| b92bc47a05925c29efef0d66711b6d3f17f406e7 |
16-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Use switch in AudioMixer::setParameter"
|
| 788040c5189bbdaf567ce4b29ffd1db08ea1020c |
05-May-2011 |
Glenn Kasten <gkasten@google.com> |
Use switch in AudioMixer::setParameter
Replace series of if/then/elses by easier-to-read switch. Also return
void instead of status_t, since callers weren't checking it. Assert on
bad input parameters.
Change-Id: Ie1f0a297977b28501d20e1af819afed9b4750616
udioMixer.cpp
udioMixer.h
|
| 237a624f674800d2300806b115eee8c9bb7db033 |
16-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
deleteTrackName now asserts on bad input parameter
This is safe, as the input parameter is always track->name(),
which must be valid.
Change-Id: Iea8ea3a5706c27026335526ba8851030d00681f8
udioMixer.cpp
|
| 3b21c50ef95fe4e7ac3426ca14b365749e66ff08 |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Extract out audio DSP code to utility library
Change-Id: Ib8ce72028a7ea30e82baa518e381370e820ebbd0
ndroid.mk
udioFlinger.cpp
udioMixer.cpp
udioMixer.h
|
| 6d4bf6eda04888144ec59cba4816a2cbb79b9e5c |
16-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "setActiveTrack and setBufferProvider can't fail"
|
| b87c068727a15a3d3f0bfdcb758c76a097f5e869 |
16-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify enable/disable mixing"
|
| 079123ee3d2e20bbc17a7ddbd96ca46bed27898f |
16-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Improve resistance to leaks for ConfigEvent"
|
| 26fa039c3752eaaf74d1be53d9795f48e9f43de3 |
16-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Use NULL not 0 for pointers"
|
| fba380a0b2c62684ce4272a05165965bae29d6a1 |
16-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
setActiveTrack and setBufferProvider can't fail
Return void, not status_t, from setActiveTrack and setBufferProvider.
These methods returned status_t, but the callers never checked the
return value. Since these aren't externally visible APIs, they now
return void, and assert on bad input parameters.
Change-Id: I530ed29484596ae41e8659826ca425149c51c2a1
udioMixer.cpp
udioMixer.h
|
| 1c48c3c61970527b97892ab6a2daae8eaac26964 |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Simplify enable/disable mixing
The MIXING enum isn't needed, and now returns void instead of status_t.
Change-Id: Ibe4ec24081d75ad4ab78b9c7191fc9077959c4e9
udioFlinger.cpp
udioMixer.cpp
udioMixer.h
|
| f3990f2cc8fd824ae52a880a7b22248e1bdfb192 |
13-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Improve resistance to leaks for ConfigEvent
A Vector of pointers is risky, as there is no ownership (and the
ThreadBase destructor was not deleting them, so if there were any left
over at end it would leak). Replaced by a Vector of values.
Change-Id: Iddde72dc30134adfcf724dec26cbe0a742509b8c
udioFlinger.cpp
udioFlinger.h
|
| e0feee3da22beeffbd9357540e265f13b2119cbb |
13-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Use NULL not 0 for pointers
Change-Id: Iab3f9abbdab617dc5a599e657ec46a0b0a002eef
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
|
| 42968939dfce0954d6540011199045ec4ed7de80 |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix indentation and whitespace"
|
| a06a9a50b37d60e9c43c9de9f8ea3a8649cd5691 |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Improve AudioFlinger error logging"
|
| 53a166cabc956529d2336b04c26b9c03b550ab38 |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioMixer uninitialized simple fields"
|
| 0cfd8231e4c489392809bf44c174315df2690145 |
13-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
AudioMixer uninitialized simple fields
Change-Id: I47e3849f048d4d990c9634f9c8a2ca2de0339e93
udioMixer.cpp
|
| d879601ace079e3c0aed79cf3fa5fb4db6ad4a9f |
28-Oct-2011 |
Glenn Kasten <gkasten@google.com> |
Improve AudioFlinger error logging
Change-Id: I8ce9aff4038cd7fa0067600faa8080b137db1939
udioFlinger.cpp
|
| 2eda60a8485cfe70a60e72156beffdc470ecb093 |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Audio C++ comments"
|
| c5ac4cb3a5124860ccfc7e4ff66251c55a5595ca |
12-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Fix indentation and whitespace
Use git diff -w to verify.
Change-Id: Ib65be0a1ecf65d6cad516110604e3855bf68a638
udioFlinger.cpp
udioMixer.cpp
|
| c23bd9b5b9e4be9c395789810fdd8522296fc50c |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Use const char correctly in AudioFlinger"
|
| e5dfcd8c6792c4b64120fd03708729b70a887f2a |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Use units after all times"
|
| b299dc4ded29a226daac07f195d1558e660d2f9f |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove redundant clear()"
|
| 362c4e697d8e9c034e964ac7b40227e054491547 |
14-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Audio C++ comments
Change-Id: I84906ebb9dfcfa5b96b287d18364b407f02a30c1
udioFlinger.cpp
udioMixer.h
|
| d1d8f23dde239289ba8c0de0944baaca3586c880 |
08-Jul-2011 |
Glenn Kasten <gkasten@google.com> |
Fix includes
Remove unused #include dlfcn.h
Add #include String8.h
Forward class declaration is only for references and pointers.
In this case, we need the full class declaration,
and were incorrectly depending on another header file to do it.
Change-Id: Iff65cceb3982698cc1ebaeec405695fc190c03ff
udioPolicyService.cpp
udioPolicyService.h
|
| 91eb8bfbe253a6b6fe1aa23fb884a601c28991c4 |
13-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Remove redundant clear()
Change-Id: Ie5e4e63cbc8fa85ef50451dddf8f149fa864b132
udioFlinger.cpp
|
| ec1d6b5e17281a066d618f7fcd2b63b3ce11f421 |
12-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Use const char correctly in AudioFlinger
Use const char [] instead of const char * to eliminate unnecessary pointer.
Make the array audio_interfaces also const, in addition to the strings.
Change-Id: I31f33d1dcb9a657ee136f4280fd2d46492496831
udioFlinger.cpp
|
| 7dede876998ff56351d495ec3a798c1b131193e8 |
13-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Use units after all times
Change-Id: I48d3f29c37228b5d03189e4c9600824c9360cac9
udioFlinger.cpp
|
| 2013d4d159bfc29b4143d3b5fd4735f51a03684c |
06-Dec-2011 |
Eric Laurent <elaurent@google.com> |
am 5433e25f: am 7b6aff23: Merge "audioflinger: fix audio skipping over A2DP" into ics-mr1
* commit '5433e25f6ce013860ff2a074ad8d1158cc39ab91':
audioflinger: fix audio skipping over A2DP
|
| 162b40bbaf3c3a24f61a6636bef6f80a9c0a31dd |
05-Dec-2011 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix audio skipping over A2DP
The maximum sleep time allowed in the mixer thread when audio tracks
are enabled but not ready for mixing is derived from the latency
reported by the output stream.
This does not work for A2DP where the latency also reflects encoding, decoding
and transfer time.
Modified activeSleepTimeUs() to take A2DP case into account.
Issue 5682206.
Change-Id: I3784ac01fb6f836b5a6ce6f764fb15347586de35
udioFlinger.cpp
udioFlinger.h
|
| 926798f8c21ab002d9797ef8973852a2612c1f75 |
23-Nov-2011 |
Eric Laurent <elaurent@google.com> |
am f6422f5f: am 20398fac: Merge "audioflinger: reduce sleep time to avoid underrun" into ics-mr1
* commit 'f6422f5f5d04aab47f8f36a0ea92e2140bed0105':
audioflinger: reduce sleep time to avoid underrun
|
| 7cafbb32999049873d4746ba83bd20c88abe6ce6 |
23-Nov-2011 |
Eric Laurent <elaurent@google.com> |
audioflinger: reduce sleep time to avoid underrun
Progressively reduce the sleep time applied in MixerThread::threadLoop()
in case of consecutive application underruns to avoid starving the audio HAL.
As the default sleep time is longer than the duration of an audio buffer
we ended up writing less data than needed by the audio HAL if
the condition persisted.
Issue 5553055.
Change-Id: I2b23ee79c032efa945025db228beaecd1e07a2e5
udioFlinger.cpp
|
| c23e2f2464eb3748599d47af7d8986b856f3c179 |
17-Nov-2011 |
Glenn Kasten <gkasten@google.com> |
Bug 5522189 re-enable ARM opts but disable inline
PIE did not work together with inlining. This change combines (almost
all of) the performance benefits of ARM optimizations together with PIE.
Change-Id: I4594d33ae5a0a7bac327ae08e30fb35343a06256
udioResampler.cpp
|
| c4795ecad4e5a0b3ec54862a40c82ef1ba53cd59 |
14-Nov-2011 |
Eric Laurent <elaurent@google.com> |
am 25924f8f: am 030bb998: Merge "audioflinger: fix noise when skipping to next song" into ics-mr1
* commit '25924f8f6c0a4ca4a2eb257b72d9625f69d2525e':
audioflinger: fix noise when skipping to next song
|
| 544fe9b6e9325701df4ab8c1d29774fc13c4cf6c |
12-Nov-2011 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix noise when skipping to next song
When audio effects are enabled, a noise can be heard at the
beginning of the new song when skipping to next song in music app.
This is because some effects (especially virtualizer) have a tail.
This tail was not played when previous song was stopped because effects were
not processed when no tracks were present on a given session. This is to
reduce CPU load when effects are enabled but no audio is playing.
The tail was then rendered when the new song was started.
Added a delay before stopping effect process after all tracks have been removed from a session.
Issue 5584880.
Change-Id: I815e0f7441f9302e8dfe413dc269a94e4cc6fd95
udioFlinger.cpp
udioFlinger.h
|
| 6977ca7d5ffdbc1610a95c74653b1fbe6a665f32 |
10-Nov-2011 |
Eric Laurent <elaurent@google.com> |
am db7d79e6: am 2b7f91b9: Merge "Fix regression for SoundPool playback" into ics-mr1
* commit 'db7d79e6f1e1860a9bfe4756a03c753435fd0ddf':
Fix regression for SoundPool playback
|
| a47b69c6f7c6fe0044ebcb2d0790ce3548de56fd |
09-Nov-2011 |
Eric Laurent <elaurent@google.com> |
Fix regression for SoundPool playback
Commit 19ddf0eb introduced a problem with applications (like SoundPool)
relying on an underrun condition to detect end of playback instead of
stopping the track when all data is written.
AudioFlinger would keep waiting for new data in case of partial buffer
filling and never reach the underrun condition.
Added a mechanism to wait no more than once if not enough frames are present
in the track buffer.
Issue 5585490.
Change-Id: I131e605ff6070831a01ddf734e68459e3bf2354b
udioFlinger.cpp
|
| 3b86c964df855a9740c446e984309b719c3ec37c |
08-Nov-2011 |
Eric Laurent <elaurent@google.com> |
am f3a892ab: Merge "AudioFlinger: mix track only when really ready" into ics-mr1
* commit 'f3a892ab9347ce733b81ccb4913a91c586f8f367':
AudioFlinger: mix track only when really ready
|
| 3dbe3201479828e84abe02e1fdd0a5d414c0ddb8 |
03-Nov-2011 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: mix track only when really ready
The addition of low power audio playback mode made that audio buffer consumption
by audio HAL can now happen in bursts. This makes that requesting audio data
from an AudioTrack for mixing can happen at much shorter intervals than before.
This revealed an existing problem where AudioFlinger would consider a track ready
for mixing although not enough frames were ready to completely fill one output buffer,
thus creating short periods of silence.
The fix consists in waiting for enough frames to be ready in AudioTrack buffer before
declaring a track ready for mixing. This minimum is not applied when the track is stopped
to allow the buffer to be emptied completely.
Change-Id: I6d04f9b65db5af85b0b53f0a5674be7ec02f9e9f
udioFlinger.cpp
|
| 305f4b6dabe500b97062e72049a296d0699243d7 |
28-Oct-2011 |
Glenn Kasten <gkasten@google.com> |
Bug 5522189 temporary workaround
Change-Id: I21d91d7a24df7bb6e7fc3d0fbc4786d55391fc0e
udioResampler.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
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
udioResampler.cpp
|
| a85a74a8219c03f2b1d1ef98f3f02e55f89f89a3 |
19-Oct-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 381905: BassBoostTest CTS tests fail...
When AudioEffectTest is executed, an Equalizer is created
and enabled on a MediaPlayer session. Effects on the output
mix are therefore suspended.
Then the MediaPlayer is released with the effect still enabled.
In this case, Audioflinger::purgeStaleEffects_l() fails to restore
the suspended effects when the effect attached to the released audio session
is removed.
When subsequent tests are executed on output mix effects, these effects cannot be
enabled as they are still suspended.
Fixed purgeStaleEffects_l() to restore suspended effects if the effect removed is enabled.
Also fixed EffectHandle::disconnect() to only restore suspended effects if the disconnected
handle actually has control over the effect.
Change-Id: I67232e7c34680b0cc01abfd57d5d510a524e5d4f
udioFlinger.cpp
udioFlinger.h
|
| 5c4e818c39ac2d2739675fe907904a874f7623c5 |
19-Oct-2011 |
Eric Laurent <elaurent@google.com> |
Limit AudioFlinger log.
AudioFlinger logs a warning when a write to the audio HAL
takes too long to return. The threshold for this warning is
a rule of thumb based on the assumption that the audio HAL will consume
buffers at a regular pace.
The introduction of low power audio mode with larger buffers and writes
occuring in bursts makes that this threshold is often exceeded resulting
in excessive and misleading warnings.
The threshold is raised to remove unwanted warnings but we should reconsider
the usefulness of this warning altogether.
Change-Id: I5ef6898ea28d879cede3e47da542a64092a3cca4
udioFlinger.cpp
|
| ec35a1416472865dbebc22b10199ad718ed2cc95 |
06-Oct-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 5381089: problem with A2DP music volume
This problem only occurs when audio effects are present and
the music volume is applied by one effect engine.
When connecting or disconnecting A2DP, audio effects are moved from
one mixer thread to another. When removed from the source thread,
the effect is stopped but it is not restarted when added to the
destination thread.
This regression was introduced by commit 21b5c47e.
Change-Id: I4cc578d8d760ec65b185032b6fda98c739d331bc
udioFlinger.cpp
udioFlinger.h
|
| 9d18ec574f5d847a86a21594ac39394ef5b108fc |
27-Sep-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 5373658: memory leak in AudioFlinger.
Unlink PowerManagerService binder interface death recipient
in ThreadBase destructor.
Change-Id: Iab06ae9a8a6737bb002b6416a157b0fb50c11ad5
udioFlinger.cpp
|
| 6bffdb8b598a2399e57f6ca48660fb7bdb2490b5 |
23-Sep-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 5355047: Automated effect tests fail.
Fixed several regressions in automated audio effect tests due
to changes in effect framework and visualizer FFT output range.
- Do not suspend Volume effect on session 0 when effects are
enabled on specific sessions.
- Adapt energy detection thresholds to new visualizer FFT range.
- Leave more time for BassBoost and Virtualizer effects to ramp up
before measuring the effect.
- Removed second insert reverb left by mistake on the player session
in preset reverb test.
Change-Id: I7a1ad1372d783fa7900eb9dd1d3b47f54d8d766f
udioFlinger.cpp
|
| eb8b914ad9d4331e1cdf4346731770ce69fd0e77 |
16-Sep-2011 |
Nick Kralevich <nnk@google.com> |
AudioResampler: use relative instead of absolute labels
This change is needed to allow Android to compile with -fPIE
Bug: 5328392
Change-Id: I84d947975776800a7b79c6ac75a881af461a631c
udioResampler.cpp
|
| 60cd0a0d488e604d27fc7dbb02b00348693dfde2 |
13-Sep-2011 |
Eric Laurent <elaurent@google.com> |
Issue 4345021: Audio routed to multiple devices...
There is a possiblility that the condition on which RecordThread::checkForNewParameters_l()
waits after updating the command completion status is never signalled.
This happens if the thread executing ThreadBase::setParameters() has timed out waiting
for the status (for instance if the audio HAL takes too long to execute the setParameters()
command. Then the RecordThread is stuck forever.
The fix consists in waiting for the condition with a timeout in RecordThread::checkForNewParameters_l().
Change-Id: I7fc671bc2fc43ba4acb65a2beb33ee05742f091e
udioFlinger.cpp
|
| b76e90de3c64626fe07a68469d0a59a31c8efb6b |
30-Aug-2011 |
Eric Laurent <elaurent@google.com> |
Merge "226483: A2DP connected, but music out to speaker"
|
| 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
udioFlinger.cpp
udioFlinger.h
|
| bee5337da7659b3b7128622ba1f42618b11df5be |
29-Aug-2011 |
Eric Laurent <elaurent@google.com> |
Audioflinger: reverse logic of BT NREC indication
The interpretation of BT NREC by AudioFlinger to enable
or disable AEC and NS was wrong: NREC to ON (default) means
the phone (Audio Gateway) must enable local AEC and NS.
Change-Id: I88a264e7fc9831c43bbace4f6b585baec73f2006
udioFlinger.cpp
udioFlinger.h
|
| a1884f9e9ec3836683efd7eb333ee442e8bc9d56 |
23-Aug-2011 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: add check for audio HAL init failure
Do not call audio HAL functions on the primary HW interface
if it could not be initialized properly.
Change-Id: If54059c8fd188d6c1686f9e0439994fe9411478a
udioFlinger.cpp
|
| a5cc7cce9b8aee73b08f6532710e186c02fdd1c0 |
03-Aug-2011 |
Eric Laurent <elaurent@google.com> |
Issue 5081351: isWiredHeadsetOn() permission.
AudioManager.isWiredHeadsetOn() should not require permission MODIFY_AUDIO_SETTINGS.
Remove permission checks on all getters in audio policy manager as permission enforcement
is really usefull for setters.
Also deprecate AudioManager.isWiredHeadsetOn() which name and implementation are deceptive.
Change-Id: I38f8df7c26c0d417bf0e2b74e4c11c2d143f2ecd
udioPolicyService.cpp
|
| 0270b188aa3929cc512ec6869caba1d6b60cc08c |
12-Aug-2011 |
Marco Nelissen <marcone@google.com> |
Don't remove effects on session 0
b/5159902
Change-Id: Id6f29fb1c687069f7480dd81d4745a558f202226
udioFlinger.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
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| 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
udioFlinger.cpp
udioFlinger.h
|
| c3e6572e0ff535932b1f6ffb7bcf5acd891675fb |
08-Aug-2011 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger: protect input/output stream access"
|
| b8ba0a979067a4efb0b3819bf17770793e41c15e |
08-Aug-2011 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: protect input/output stream access
Some methods would not check that the output orinput stream of a thread
was still valid before calling functions on its interface.
This could cause a crash if those methods where called while the output or
input was being closed by another thread.
Make sure that the output or input stream pointer is cleared before closing the
stream.
Always check that the output or input pointer is not null before calling
functions at the stream interface.
Generalize the use of initCheck() method to verify that the output or input
stream is not null.
Change-Id: I9d9ca6b744d011bcf3a7bbacb4a581ac1477bfa5
udioFlinger.cpp
udioFlinger.h
|
| 59bd0da8373af0e5159b799495fda51e03120ea4 |
01-Aug-2011 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: disable AEC and NS with BT headsets
Disable AEC and NS when the Bluetooth SCO headset in use indicates it
implements those pre processings.
Change-Id: I93f3d10b0a27243d5dbff7182639576fc0c6d862
udioFlinger.cpp
udioFlinger.h
|
| 59255e4fc7d8ff52874b85b1988dc0785140cf81 |
28-Jul-2011 |
Eric Laurent <elaurent@google.com> |
Audio Effect Framework: add effect suspend/restore
Add the possibility for the effect framework to suspend
(temporarily disable process) and restore audio effects.
This feature will be usefull to disable pre processing under certain
conditions and better control coexistence of audio effects
on output mix and specific sources.
Change-Id: I79b195982cc48748d5708308fb1647b9c3c34cc6
udioFlinger.cpp
udioFlinger.h
|
| 6dbe883644940badc684957cfc381bfd115f205e |
28-Jul-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 5090721: audio record broken
Commit 6dbdc40 introduced a deadlock when exiting the
AudioFlinger RecordThread.
Change-Id: I1f63e54c5aeff05da4e4d028b53f734c62c78677
udioFlinger.cpp
|
| a7280a59259018d997896c043fd2db95f631f12e |
27-Jul-2011 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger: fix crash when deleting pre process."
|
| ec437d8d3db79459d7b19e1734e6fe309bd621e8 |
27-Jul-2011 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix crash when deleting pre process.
If a pre processing effect is detroyed while enabled and capture is active,
there was a possibility that the effect engine is released by the framework
while still processed by the audio HAL.
The fix consists in not releasing the engine in EffectModule::removeHandle()
but just flag the effect as being detroyed to avoid further calls to functions
on the engine effect interface.
The effect interface is then removed from the audio HAL safely in
EffectChain::removeEffect_l() while holding the EffectChain mutex.
Change-Id: I71fab30d9145062af8644f545a1f1d4d3e7e7f02
udioFlinger.cpp
udioFlinger.h
|
| feb0db689c17dced50afaee54c659f1676e2d505 |
22-Jul-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 4604090: notification sound interrupted.
The problem is that the audio HAL fails to acquire the wake lock when playing the notification.
This is because of a change that removed the mediaserver process form the system group for honeycomb.
The fix consists in requesting the wake lock from PowerManagerService when AudioFlinger mixer
wakes up.
A consequence of this change is that audio HALs or pcm drivers do not have to hold wake locks
anymore as in the past.
Change-Id: I4fb3cc84816c9c408ab7fec75886baf801e1ecb5
ndroid.mk
udioFlinger.cpp
udioFlinger.h
|
| 57dae99c9fcd6becd2b5ed0c53f277ba4d7dbcfc |
24-Jul-2011 |
Eric Laurent <elaurent@google.com> |
Added APIs for audio preprocessing
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
udioPolicyService.cpp
udioPolicyService.h
|
| 1d2bff0e588afe183a1baaae731519b4e957bbdb |
25-Jul-2011 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: add dump of audio pre processing.
Dump of media.audio_flinger service was only listing effects on output threads.
Moved the dump of effect chains from PlaybackThread to ThreadBase class so that
pre processings on RecordThread are also listed.
Change-Id: If8bc74023c12b9c2371f1b300743b156ceca7b87
udioFlinger.cpp
udioFlinger.h
|
| edc15ad8fcde12dc4f642d80d077239b1532eeca |
22-Jul-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 4988574: 8 bit PCM audio playback broken.
Fixed regression in audio track control block frame size
calculation introduced by commit c310dcb.
Change-Id: Ia731b946ae4e43316b98d80229e3b08a696e47d6
udioFlinger.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
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| e0b5bb23f0a26d248275d203885b820659da7320 |
16-Jul-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Log CPU usage"
|
| 4d8d0c30abfa4b8d75866d42094cc797e05068fa |
09-Jul-2011 |
Glenn Kasten <gkasten@google.com> |
Log CPU usage
Change-Id: Ie447e59be139153e526b7ad467c46c659d26816f
ndroid.mk
udioFlinger.cpp
|
| 81eedd1746be590d5d506dfb9ff23619f9d9898f |
13-Jul-2011 |
Mike Lockwood <lockwood@android.com> |
Remove some #ifdef HAVE_ANDROID_OS that were needed for the simulator build
Change-Id: I13d9f251f86c05ae5405f37adbf6b8e9660935ba
Signed-off-by: Mike Lockwood <lockwood@android.com>
udioPolicyService.cpp
|
| 5e0067b486c3862316aa1f293cf9690c0cf54bda |
12-Jul-2011 |
Jeff Brown <jeffbrown@google.com> |
Remove the simulator target from all makefiles.
Bug: 5010576
Change-Id: I04d722f258951a3078fe07899f5bbe8aac02a8e8
ndroid.mk
|
| 6c21234e3d254fc58227cd2063500a0dcd173689 |
08-Jul-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix build warning; LOG_TAG was misplaced"
|
| 5159c7ec97b575c0486159f0543d2da90c3c3b68 |
08-Jul-2011 |
Glenn Kasten <gkasten@google.com> |
Fix build warning; LOG_TAG was misplaced
Change-Id: I4fb0590f626445188f9053367707878b2fa0fa30
udioResamplerCubic.cpp
|
| 6d8b694d999e9be7d5dcc336535832a80fb6f61f |
24-Jun-2011 |
Eric Laurent <elaurent@google.com> |
Moved and renamed effect API header files
Moved specific effect header files to
system/media/audio_effects/include/audio_effects
and renamed to lower case (effect_xxx.h).
Change-Id: Icfc2264bfd013cab0395d7e310ada636b9fe3621
ndroid.mk
udioFlinger.cpp
|
| 671a636931295d9c33ffca74551a804479d01241 |
17-Jun-2011 |
Eric Laurent <elaurent@google.com> |
Added audio_bytes_per_sample() helper function
Change-Id: Ibfcd75c4c241a53d5f052c25ada091904991048a
udioFlinger.cpp
|
| 7394a4f358fa9908a9f0a7c954b65c399f4268e6 |
14-Jun-2011 |
Dima Zavin <dima@android.com> |
audio: update for audio/audio_policy header names/locations
Change-Id: I36c49352eee57559403cd1597f56a8485a360289
Signed-off-by: Dima Zavin <dima@android.com>
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| 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
udioFlinger.cpp
udioPolicyService.cpp
|
| 0d255b2d9061ba31f13ada3fc0f7e51916407176 |
25-May-2011 |
Jean-Michel Trivi <jmtrivi@google.com> |
Use channel mask instead of channel count for track creation
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
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
|
| 65580f9adf6c4d98449ad0716488f9fe3869aa5a |
28-May-2011 |
Eric Laurent <elaurent@google.com> |
Removed interface to load audio effects libraries
Removed unused functions allowing dynamic loading of audio effects libraries
from effects factory API.
Change-Id: I06cc5a51dc10aca87c7a8687bbb874babd711eca
udioFlinger.cpp
udioFlinger.h
|
| e1315cf0b63b4c14a77046519e6b01f6f60d74b0 |
18-May-2011 |
Eric Laurent <elaurent@google.com> |
New effect library API
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
udioFlinger.cpp
udioFlinger.h
|
| 162f7d15ac5c8c23d1c3de171239f3a4e6e06b2a |
23-May-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "HAVE_ANDROID_OS AUDIOFLINGER_SECURITY_ENABLED dead"
|
| 6708b9a3fb654f5623ba5a696288fdba310a5e1a |
13-May-2011 |
Eric Laurent <elaurent@google.com> |
Merge "Fix audio effect framework issues"
|
| b469b9490b3cd9e0f0466d9b9ab228f6c793b82e |
09-May-2011 |
Eric Laurent <elaurent@google.com> |
Fix audio effect framework issues
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
udioFlinger.cpp
udioFlinger.h
|
| 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>
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| 249c6a61f21bc90e25e4b77f18c98af1ac363e69 |
05-May-2011 |
Glenn Kasten <gkasten@google.com> |
HAVE_ANDROID_OS AUDIOFLINGER_SECURITY_ENABLED dead
Remove dead code from the days when Android ran in emulator.
Change-Id: Ibadbbde0538239ad9b2811a3a2e8f8a6d3b6389c
udioFlinger.cpp
|
| 5a61d2f277af3098fc10b2881babca16391362da |
20-Apr-2011 |
Dima Zavin <dima@android.com> |
audioflinger: don't do work in constructor, instead do it in onFirstRef
Change-Id: I22d9e01821816c3beb52b014330386c7fd2f0411
Signed-off-by: Dima Zavin <dima@android.com>
udioFlinger.cpp
udioFlinger.h
|
| 799a70e7028a4d714436c3a744a775acfbd31aae |
19-Apr-2011 |
Dima Zavin <dima@android.com> |
audioflinger: enumerate all the possible audio interfaces
Keep track of the primary interface that handles the master volume,
etc.
Change-Id: Ib0701fccff8d8783a99035a241ab7c8ec75c00ac
Signed-off-by: Dima Zavin <dima@android.com>
udioFlinger.cpp
udioFlinger.h
|
| 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>
ndroid.mk
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| 6b80e0be94d3f92ec4aa2b7cace816780d3f338d |
20-Apr-2011 |
Dima Zavin <dima@android.com> |
audioflinger: move legacy audio hw/policy out to libhardware_legacy
Change-Id: I4adcec73d3c08bcbe15bb19e1ba2ff18b195af45
Signed-off-by: Dima Zavin <dima@android.com>
2dpAudioInterface.cpp
2dpAudioInterface.h
ndroid.mk
udioDumpInterface.cpp
udioDumpInterface.h
udioHardwareGeneric.cpp
udioHardwareGeneric.h
udioHardwareInterface.cpp
udioHardwareStub.cpp
udioHardwareStub.h
udioPolicyManagerBase.cpp
|
| dab0bea38ab2c2d6b52907b7f5684834c4979251 |
26-Apr-2011 |
Eric Laurent <elaurent@google.com> |
am 48dca4de: am 969357e3: am 223995dc: Fix issue 4335692: HDMI media volume
* commit '48dca4de70890d324b5830a58bb9fa273164151a':
Fix issue 4335692: HDMI media volume
|
| 5bfdfd1e7b4514c6f6143b6b97ef37013f6e53a9 |
26-Apr-2011 |
Eric Laurent <elaurent@google.com> |
am 969357e3: am 223995dc: Fix issue 4335692: HDMI media volume
* commit '969357e31c93e168e7a4cb3b41c33a6ec26f4a26':
Fix issue 4335692: HDMI media volume
|
| 023eeac54c428d78b755fa3c856f20d17abb8a40 |
25-Apr-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 4335692: HDMI media volume
Forced music stream volume to max when not muted and output device is HDMI.
Change-Id: Ibd287cea8ae1d3f36fea6651a113bd5cf2dbad13
udioPolicyManagerBase.cpp
|
| 5820e52f9bf3c40839d33457975b6c9629ab1c79 |
12-Apr-2011 |
Brad Fitzpatrick <bradfitz@android.com> |
am 07c25b65: am 4430670f: am ca7ad44a: Merge "audioflinger: Enable ARMv5TE optimized resampler"
* commit '07c25b651216daac884bd503c02b788418165b50':
audioflinger: Enable ARMv5TE optimized resampler
|
| 4d0243980123378e9dd69022bf4ddca729d84126 |
12-Apr-2011 |
Brad Fitzpatrick <bradfitz@android.com> |
am 4430670f: am ca7ad44a: Merge "audioflinger: Enable ARMv5TE optimized resampler"
* commit '4430670f48f75661293371dab8db55865f5f56e0':
audioflinger: Enable ARMv5TE optimized resampler
|
| 0c0a1c0c37dbd2646a732da706d6777283c83e44 |
06-Apr-2011 |
Jim Huang <jim.huang@linaro.org> |
audioflinger: Enable ARMv5TE optimized resampler
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
udioResampler.cpp
|
| c30268b9d118309a0514bcf280a03ee69f81403f |
06-Apr-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Miscellaneous code cleanup in audio framework"
|
| 4bcae82f9b07d1a39956c45a6f5bec0b696c4dd1 |
04-Apr-2011 |
Glenn Kasten <gkasten@google.com> |
Miscellaneous code cleanup in audio framework
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
udioFlinger.cpp
udioPolicyManagerBase.cpp
|
| 38ccae2c0324daa305f3fe77d25fdf5edec0b0e1 |
29-Mar-2011 |
Eric Laurent <elaurent@google.com> |
New fix for issue 4111672: control block flags
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
udioFlinger.cpp
|
| 33797ea64d067dfeaacbfd7ebe7f3383b73961b5 |
17-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 4111672: AudioTrack control block flags
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
udioFlinger.cpp
|
| ce80563f6fb49b0c40281f9a527e288799b4a577 |
15-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Improvement for issue 3489986: BT SCO volume
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
udioPolicyManagerBase.cpp
|
| 4f46fed004b76d462ef859aa94a3c056b0e79b60 |
03-Mar-2011 |
Gloria Wang <gwang@google.com> |
Merge "- To track the usage of all audio output devices - 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()"
|
| 9ee159b79022b2e1a050acb3890ce948e99e9ccb |
24-Feb-2011 |
Gloria Wang <gwang@google.com> |
- To track the usage of all audio output devices
- 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
udioFlinger.cpp
|
| 1c4967155109ab083a4149347596f7ebd3662a67 |
03-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Merge "Fix issue 3388775."
|
| c0bff9b96152179650dc9508799a38e35ef2a321 |
01-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Merge "Fix issue 3479042."
|
| 532492cd60a3ee2613cd3d664c9f4581d9929bc5 |
01-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3388775.
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
udioPolicyManagerBase.cpp
|
| 243f5f91755c01614a8cafe90b0806396e22d553 |
01-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3479042.
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
udioFlinger.cpp
udioMixer.cpp
udioMixer.h
udioResampler.cpp
udioResampler.h
|
| 043b22dbe3b98a8fdd6b0ea77df4089ff44b6539 |
28-Feb-2011 |
Jean-Michel Trivi <jmtrivi@google.com> |
Fix bug 3376700 Volume too loud at lowest setting
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
udioPolicyManagerBase.cpp
|
| 586b21780680c4d19723b5a9484e55ec47464c5f |
11-Feb-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3400751.
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
udioPolicyManagerBase.cpp
|
| 727a6f859841a1d421b680f56d4d4ca5ff8c6e4b |
11-Feb-2011 |
Eric Laurent <elaurent@google.com> |
Merge "Fix issue 3425342."
|
| 12932bfdc7163e654d5ef25f1cb1802ded2704f1 |
11-Feb-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3425342.
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
udioPolicyManagerBase.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
udioPolicyManagerBase.cpp
udioPolicyService.cpp
udioPolicyService.h
|
| 8473bb556421ac08f4e03abd5d149852114c9238 |
07-Feb-2011 |
Brad Fitzpatrick <bradfitz@android.com> |
am be806fe8: am cc8f87e9: am f664d6f9: am b083d3b8: Merge "Initialize resampling buffer per track."
* commit 'be806fe8c1c7bb3ae70ae27dce41d672410af26a':
Initialize resampling buffer per track.
|
| 802da7bc462f6f9df6e681cc75c51dc0a056a4a1 |
04-Feb-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Bug 3366668 Use BinderService template"
|
| e08d819989d394ed171ba6f1e6d0fa14c29fc42d |
04-Feb-2011 |
Brad Fitzpatrick <bradfitz@android.com> |
am cc8f87e9: am f664d6f9: am b083d3b8: Merge "Initialize resampling buffer per track."
* commit 'cc8f87e9410dd4de9a2fda4738429e6c6087c789':
Initialize resampling buffer per track.
|
| e26308ad434c575b2985ac442d2990bb1bc19f5d |
04-Feb-2011 |
Brad Fitzpatrick <bradfitz@android.com> |
am b083d3b8: Merge "Initialize resampling buffer per track."
* commit 'b083d3b816378ef3b9dceb33b2c2e20510b2632b':
Initialize resampling buffer per track.
|
| 2151d7b8c2dd77c9887691db30396937be778141 |
04-Feb-2011 |
Yuuhi Yamaguchi <yuuhi.x.yamaguchi@sonyericsson.com> |
Initialize resampling buffer per track.
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
udioMixer.cpp
|
| 117cd9286424888c1c5bf202ebf1e08ae1e6affe |
04-Feb-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Bug 3366885 Remove LVMX switch"
|
| 8782cb4dc783cdcac8498148ab1894b4780c18df |
04-Feb-2011 |
Eric Laurent <elaurent@google.com> |
am 6f1bd261: am 9c0a1003: Merge "Fix issue 3371080" into honeycomb
* commit '6f1bd261b7fd86ac7817ca061dfb55b95150b836':
Fix issue 3371080
|
| db130fbd3ccd37e247e49494a84f8a9841ecd593 |
04-Feb-2011 |
Glenn Kasten <gkasten@google.com> |
Bug 3366885 Remove LVMX switch
Change-Id: I0bf98c6f85f00b3296874571e1c049dcc4e2fcca
ndroid.mk
udioFlinger.cpp
udioFlinger.h
|
| d2dcb08808aed82218b258726fa66f8e095a3f9c |
04-Feb-2011 |
Glenn Kasten <gkasten@google.com> |
Bug 3366668 Use BinderService template
Change-Id: I93d7f3fc9dc9b6a365723d8a51a73a5aabdb4f93
udioPolicyService.cpp
udioPolicyService.h
|
| 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
udioFlinger.cpp
udioFlinger.h
udioPolicyManagerBase.cpp
udioPolicyService.cpp
udioPolicyService.h
|
| 2a30be33a39b07bf4f945d9506ffda0fd076e6c2 |
29-Jan-2011 |
Jean-Baptiste Queru <jbq@google.com> |
Merge 0ef57993 from gingerbread
Change-Id: If10fee1ae387a8130356dd62fe678495402d5edf
|
| 144f78906f1eefaf8cbde9298a1422fcea094447 |
29-Jan-2011 |
Jean-Baptiste Queru <jbq@google.com> |
am 4eeb1047: Merge 13212f83 from gingerbread-plus-aosp
* commit '4eeb10470ffafe8c508027f363ac66b58da5bf00':
Fix issue 2988031.
|
| 2ef89309d41daac3d69fa78b31cd09cb2c262ca1 |
29-Jan-2011 |
Jean-Baptiste Queru <jbq@google.com> |
Merge 13212f83 from gingerbread-plus-aosp
Change-Id: I9a8ee0c7e7896aea85e7a7c18ee82927091cb670
|
| 5092d8c066a1a98343bbc16e33a6753e577b5e7f |
27-Jan-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 2988031.
Limit SYSTEM stream volume when a headset is connected and music is playing.
Change-Id: Ieb44ae5bb53ffa9cd5fe8e317798eed279b78df8
udioPolicyManagerBase.cpp
|
| bfa2f13fd3f463dbceea4d3a18c3124e70df0a05 |
15-Nov-2010 |
Jean-Michel Trivi <jmtrivi@google.com> |
do not merge bug 3370834 Cherrypick from master
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
udioFlinger.cpp
udioHardwareInterface.cpp
udioPolicyManagerBase.cpp
|
| 61dca6702ce50d937fc764c55bb0842b40dcd0e4 |
25-Jan-2011 |
Jean-Michel Trivi <jmtrivi@google.com> |
Bug 3376700 Add support in APM for stream-specific volume curves
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
udioPolicyManagerBase.cpp
|
| b517e006b51c95db9c44bf5f104f562d431ae597 |
20-Jan-2011 |
Eric Laurent <elaurent@google.com> |
do not merge - Fix issue 3371096.
EffectModule::process() was copying effect chain input buffer to
output buffer if no effect was active instead of accumulating it.
Change-Id: If4ca75601ea69a088d0f71d88aec53e90a1dec89
udioFlinger.cpp
|
| 73337489229cc9b50371c7a9fcd86e9f00ea46d0 |
20-Jan-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3371096.
EffectModule::process() was copying effect chain input buffer to
output buffer if no effect was active instead of accumulating it.
Change-Id: I2838af2e7b6654d0a76547625929a5453da68d02
udioFlinger.cpp
|
| 935752053ef2691dbb6d5a6d149e0e362c6e3c74 |
19-Jan-2011 |
Eric Laurent <elaurent@google.com> |
Tentative fix for issue 3362362.
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
udioFlinger.cpp
udioPolicyManagerBase.cpp
udioPolicyService.cpp
|
| eb0d0c48ebfcee5e8141ae25fe42a9eac6aae230 |
10-Jan-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3317627.
The fix consists in selecting the digital audio device (SPDIF/HDMI)
when available if the routing strategy is STRATEGY_PHONE.
Change-Id: Ie500ae92f5c01f2511988543852ba559c6e5994b
udioPolicyManagerBase.cpp
|
| 626c85f3a9ec8e34a1e4af31eddf118d75ee6416 |
17-Dec-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 3217707.
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
2dpAudioInterface.cpp
2dpAudioInterface.h
|
| 0b456a6a33de8cb7c641724579382a31a1beaa2c |
16-Dec-2010 |
Eric Laurent <elaurent@google.com> |
Fix speakerphone routing to analog dock
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
udioPolicyManagerBase.cpp
|
| f3a8d32fd475ec68d966fa6f12a4c975ba80b659 |
15-Dec-2010 |
Eric Laurent <elaurent@google.com> |
Change audio routing policy for HDMI
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
udioPolicyManagerBase.cpp
|
| b3fd981d5f8bbf629dd8a07b06b4b656b70cb944 |
10-Dec-2010 |
Eric Laurent <elaurent@google.com> |
Fix audio mode log.
Since the new audio mode IN_COMMUNICATION was added, the audio mode log
was broken.
Change-Id: I4fdafc3b98a1b0ceb55058a9e47fed99b3dbe6ad
udioHardwareInterface.cpp
|
| ed7c671a57f89d389c9d6cd67e0698f160f69916 |
01-Dec-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2641884: Bluetooth volume is dependent on in call volume.
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
udioPolicyManagerBase.cpp
|
| 094b148b8587a67c19585e2af2cc7b53868bfb89 |
01-Dec-2010 |
Eric Laurent <elaurent@google.com> |
resolved conflicts for merge of 0d28be68 to master
Change-Id: Iec5f810c366d3e1c14a6f6294b0aea4ffb30ae3e
|
| 075a1f660b77fe47b165db1aff97cdd05e300391 |
02-Nov-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 3142808.
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
udioPolicyManagerBase.cpp
|
| 32cb1ba657a1094bb815f4d39f3e4bf12ce8437e |
24-Nov-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 3225810.
Take a wake lock whenever A2DP output stream is active.
Change-Id: Ie50e6d4cb34c8a1ba97b301ef25e10aeb153d8f3
2dpAudioInterface.cpp
2dpAudioInterface.h
|
| f5aafb209d01ba2ab6cb55d1a12cfc653e2b4be0 |
18-Nov-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 3157123.
Use a Mutex wherever atomic operations were used in AudioTrack,
AudioRecord, AudioFlinger and AudioEffect classes.
Change-Id: I6f55b2cabdcd93d64ef19446735b8f33720f8dbc
udioFlinger.cpp
udioFlinger.h
|
| f1fb01a7f00b8da90a36268aba8584a872e99175 |
15-Nov-2010 |
Jean-Michel Trivi <jmtrivi@google.com> |
Add new audio mode 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.
Change-Id: Id744cd26520ea1d1a4795eabe6a1f0c58789af76
udioFlinger.cpp
udioPolicyManagerBase.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
udioPolicyManagerBase.cpp
|
| c643d77a099b0e86ad12de9b498f6f780e03184e |
09-Nov-2010 |
Jean-Michel Trivi <jmtrivi@google.com> |
Add recording source for voice communication
Add a recording source used to designate a recording stream for
voice communications such as VoIP.
Change-Id: I4091d67069b1a0170c1a5ca5e6acd51eb0aa08f9
udioPolicyManagerBase.cpp
|
| b235dee7be544a6c990ba4f8d89ab03e38c826f8 |
06-Oct-2010 |
Praveen Bharathi <pbharathi@motorola.com> |
Added support for dock headset observer
Change-Id: I06b2e65e3bfa10735e6c7fd3349afa9ae7d45292
Signed-off-by: Praveen Bharathi <pbharathi@motorola.com>
udioPolicyManagerBase.cpp
|
| 173e75df12ad5398f3650a30e430dcac0cbcf278 |
07-Oct-2010 |
Eric Laurent <elaurent@google.com> |
am ce2e2184: am 37947afe: Merge "Fixed AudioFlinger not always pausing tracks" into gingerbread
Merge commit 'ce2e2184bbc5530f4fac3220fdf3d1b3fc08a4c3'
* commit 'ce2e2184bbc5530f4fac3220fdf3d1b3fc08a4c3':
Fixed AudioFlinger not always pausing tracks
|
| af59ce2407fa4e6e5d8f1664a4df2daf1badd407 |
05-Oct-2010 |
Eric Laurent <elaurent@google.com> |
Fixed AudioFlinger not always pausing tracks
If the pause request is received before the AudioTrack buffer was
completelly filled and the track ready for mixing, the pause is
not executed: the track just underruns and stays in pausing state.
The fix consists in considering the track ready for mixing immediately
if pausing.
Change-Id: Ia6cb4703fee2126e41011a6400ea8eeb3a3e5456
udioFlinger.cpp
|
| 9ad6e8bd9fab860e639f6ce95479e4b2a9b03ff5 |
01-Oct-2010 |
Eric Laurent <elaurent@google.com> |
am de12c3cf: am 220ab887: Merge "Issue 3032913: improve AudioTrack recovery time" into gingerbread
Merge commit 'de12c3cf56e3f27b2efc60eeae8b5e422747f2b9'
* commit 'de12c3cf56e3f27b2efc60eeae8b5e422747f2b9':
Issue 3032913: improve AudioTrack recovery time
|
| 44d9848d6656777a18019223e0d35f2fcc67719a |
01-Oct-2010 |
Eric Laurent <elaurent@google.com> |
Issue 3032913: improve AudioTrack recovery time
This issue showed that when an AudioTrack underruns during a too long period
of time and is therefore disabled by audioflinger mixer, it takes an additional
delay of up to 3 seconds to recover.
This fix adds a simple mechanism to recover immediately when the client application
is ready to write data again in the AudioTrack buffer
Also throttle warnings on record overflows
Change-Id: I8b2c71578dd134b9e60a15ee4d91b70f3799cb3d
udioFlinger.cpp
|
| 9e2e2d2bc86372bde4673c31ef416b2f29d87ac2 |
29-Sep-2010 |
Eric Laurent <elaurent@google.com> |
am b047e3cd: am aeb2c62e: Merge "Fix several audio effects problems." into gingerbread
Merge commit 'b047e3cdf24b32e26f366fcd4cd0eee8ae6c592c'
* commit 'b047e3cdf24b32e26f366fcd4cd0eee8ae6c592c':
Fix several audio effects problems.
|
| dac69110ed1073bf0a9827a3f78698896dd05d97 |
28-Sep-2010 |
Eric Laurent <elaurent@google.com> |
Fix several audio effects problems.
Fixed the following issues in LVM effect bundle wrapper:
- memory leaks in EffectCreate() in case effect creation fails at various stages
- Added saturation when accumulating to output buffer
- Fixed problems with enabled effects count when an effect is released while enabled
- Do not allocate temporary buffer for accumulation each time process() is called
Fixed the following issues in effects framework (AudioFlinger)
- Release effect synchronously in the library when deleted from effect chain
- Do not call the effect process function if no tracks are present in the same
audio session
Change-Id: Ifbd80a163415cfb3c0a337c12082853ea45d9c91
udioFlinger.cpp
|
| f43c92f8d7b1d81cc7248031970d68dc65579be0 |
24-Sep-2010 |
Eric Laurent <elaurent@google.com> |
am 692dfafe: am 880dfe4f: Merge "Fix issue 3007862" into gingerbread
Merge commit '692dfafe02d04cdbab5367546e166580c92e4d2e'
* commit '692dfafe02d04cdbab5367546e166580c92e4d2e':
Fix issue 3007862
|
| 84e9a10fde8a4ae3da4f88d7911c154933aa457f |
24-Sep-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 3007862
Removed a cross deadlock condition between audioflinger and audio policy
service mutexes.
Audioflinger::createEffect() locks audioflinger mutex and then calls
AudioSystem::getOutputForEffect() which ends up in
AudioPolicyService::getOutputForEffect() which locks audio policy service
mutex. If at the same time, the command thread in audio policy service is
processing a command(set volume, set route...), the mutex is locked and the
command will call one audioflinger method which in turn will attempt to
lock audioflinger mutex.
The fix consists in releasing audioflinger mutex before calling
getOutputForEffect().
Change-Id: Id44e7feb36e0a295731f6aa97cf32d022edd34d0
udioFlinger.cpp
|
| cc08406f74b67b1b7e532ef8ed534c0476ced636 |
23-Sep-2010 |
Eric Laurent <elaurent@google.com> |
am 11746caa: am 08959c63: Merge "Request permission for global audio effects." into gingerbread
Merge commit '11746caaa852984ff186bf5b8807e2c14cd7c1bc'
* commit '11746caaa852984ff186bf5b8807e2c14cd7c1bc':
Request permission for global audio effects.
|
| fac4895de4ae63928ff0cf2ccece106eb6d33f72 |
22-Sep-2010 |
Eric Laurent <elaurent@google.com> |
Request permission for global audio effects.
Applications creating an audio effect on the output mix must
have the MODIFY_AUDIO_SETTINGS permission.
Change-Id: I57d88533f91ad0d33680107d79abcec28f7263b5
udioFlinger.cpp
|
| d20b4cfde2848052cf93ed13df14d75347ec64fd |
13-Sep-2010 |
Eric Laurent <elaurent@google.com> |
am 4d987850: am bd2e9ec6: Merge "Fix volume problems with insert revert" into gingerbread
Merge commit '4d9878502f7661ed34540a485a5942d859e209c7'
* commit '4d9878502f7661ed34540a485a5942d859e209c7':
Fix volume problems with insert revert
|
| e0aed6ddcb4e3c301b80aa26706b6052dab42c41 |
11-Sep-2010 |
Eric Laurent <elaurent@google.com> |
Fix volume problems with insert revert
- Use a constant input level to the reverb engine and implement volume control in the
insert reverb. This avoids the volume spikes when an effect that was inserted after
the reverb is disabled or removed.
- Fix clicks (one silent buffer) at the end of the reverb disable period.
- Modified volume management in audioflinger so that the volume ramp is also done by
the insert effect if present when the track is paused (avoids clicks).
- Increased room level for all presets.
Also fixed problems with output stage session (-1):
- effect bundle wrapper was not designed to support session -1
- the permission check in audioflinger for using session -1 failed due to a wrong usage of
getCallingPid()
Change-Id: Id1ff51327263364bf71d3f2668fa5cde4311d84f
udioFlinger.cpp
|
| e5fe2601b25e3220c3c4cfb79695ddb8a2f817ac |
02-Sep-2010 |
Eric Laurent <elaurent@google.com> |
am 937056c1: am 32ec1ad1: Merge "Fix problem in AudioEffect::command() status." into gingerbread
Merge commit '937056c1a7ce71e6881305a31f76e62a76685f24'
* commit '937056c1a7ce71e6881305a31f76e62a76685f24':
Fix problem in AudioEffect::command() status.
|
| aeae3de947fa0b1e670c8472b32288962f97b4f5 |
02-Sep-2010 |
Eric Laurent <elaurent@google.com> |
Fix problem in AudioEffect::command() status.
The *pReplyData argument of the command() function was left unitialized by EffectHandle::command()
when command was EFFECT_CMD_ENABLE, EFFECT_CMD_DISABLE and EFFECT_CMD_SET_PARAM_COMMIT.
Change-Id: I91a19817ead2a8cfbdd8e2d77ca270c7ce9d5bd4
udioFlinger.cpp
|
| 1040127473ee48c1c3b8c279ae05d38dbfb6e4c2 |
01-Sep-2010 |
Eric Laurent <elaurent@google.com> |
am 55e79374: am f560ceab: Merge "Audio Effects: fix problems in volume control." into gingerbread
Merge commit '55e7937462d107b469a58651d3d51e15086f8e9e'
* commit '55e7937462d107b469a58651d3d51e15086f8e9e':
Audio Effects: fix problems in volume control.
|
| 8f45bd725549436eeacd12ee69349e2332ed8da5 |
31-Aug-2010 |
Eric Laurent <elaurent@google.com> |
Audio Effects: fix problems in volume control.
- Fixed click when re-enabling effect during the turn off phase:
make sure the effect states where effect is processed are the same
where volume control is delegated to effect.
- Fixed click when effect is deleted while still active: do not apply
volume ramp if an effect having volume control was just removed from the
effect chain.
Also fixed a crash when PCM dump is enabled in effect bundle wrapper.
Change-Id: Ib562f5cf75c69af75df0e862536262e2514493e4
udioFlinger.cpp
udioFlinger.h
|
| ea798dd9f37b34acfa3cf8d945dbe7f7e90e7196 |
31-Aug-2010 |
Eric Laurent <elaurent@google.com> |
am 44217848: am ae6bdc23: Merge "Fix issue 2952766." into gingerbread
Merge commit '4421784895a58bb7bcf90236a9e443b372b5b80e'
* commit '4421784895a58bb7bcf90236a9e443b372b5b80e':
Fix issue 2952766.
|
| c1c88e27a6f07a530fca184f515c16d51e46d56f |
28-Aug-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2952766.
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
udioPolicyManagerBase.cpp
|
| 69e0b8c5556df1d4db1a730db1c13409a96749c7 |
21-Aug-2010 |
Eric Laurent <elaurent@google.com> |
am 3a899b85: am 0b9a12bf: Merge "Fix issue 2929440" into gingerbread
Merge commit '3a899b85102f6d51585b3d46ced3038c8602f202'
* commit '3a899b85102f6d51585b3d46ced3038c8602f202':
Fix issue 2929440
|
| 25cbe0ecd6df8be7e40537c5d85c82f105038479 |
19-Aug-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2929440
Fixed regression introduced by change a54d7d3d7dd691334189aab20d23c65710092869 in audioflinger mixer thread:
When the output stream is suspended, the sleep time between two writes must match the actual duration
of one output stream buffer otherwise the playback rate is not respected.
Change-Id: Ic5bebe890290d1f44aeff9dd3c142d18e26fff2a
udioFlinger.cpp
udioFlinger.h
|
| a1a6cba3664a04d214b372526658ba227e4d5366 |
14-Aug-2010 |
Eric Laurent <elaurent@google.com> |
am f28608cd: am c8d0d630: Merge "Fix issue 2909189: System property ro.audio.silent no longer mutes system." into gingerbread
Merge commit 'f28608cdf4f6a0075d0811ccd32123baf4daec31'
* commit 'f28608cdf4f6a0075d0811ccd32123baf4daec31':
Fix issue 2909189: System property ro.audio.silent no longer mutes system.
|
| 571d49c1c316f5e07b74ed7b5df6bdec7cbc1a14 |
11-Aug-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2909189: System property ro.audio.silent no longer mutes system.
Fixed regression introduced by commit 2a6b80bc65c4782b5a7168b300e1dc5ec9f617ee:
master mute was not working if no effect chains were present on session 0.
Change-Id: I66d107e045d159cb94d29c7476fa1e12d92f2ae7
udioFlinger.cpp
|
| bc417c0531332635df9525df4683274a08fb987f |
10-Aug-2010 |
Eric Laurent <elaurent@google.com> |
am 1ece7d68: am 2a6b80bc: Fixed several audio effects problems.
Merge commit '1ece7d6873027068a3e7b055597045e9387b217c'
* commit '1ece7d6873027068a3e7b055597045e9387b217c':
Fixed several audio effects problems.
|
| 8569f0d3bf4c6787707e348a7cf73b9c4199cb32 |
30-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Fixed several audio effects problems.
- Fixed constant inversions in AudioEffect.java
- Do not return error when enabling an already enabled effect
- Update cached effect state in native AudioEffect class when effect is enabled/disabled by command() method
- Remove click when restarting effect during disable sequence
- Fixed problem in master mute management when volume control is delegated to effect.
Change-Id: I6df4ce9fcc54fdc7345df858f639d20d802d6712
udioFlinger.cpp
|
| e679114b6c7e087a59c2b2c3b55603c43da4ef38 |
29-Jul-2010 |
Eric Laurent <elaurent@google.com> |
am ba676a00: am a54d7d3d: Fixed underrun in audioflinger mixer.
Merge commit 'ba676a00c1b9ba59b30fe805b83d03d1768fffbd'
* commit 'ba676a00c1b9ba59b30fe805b83d03d1768fffbd':
Fixed underrun in audioflinger mixer.
|
| 60e182437228312cc28469a5b0dfde77ac848e1a |
29-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Fixed underrun in audioflinger mixer.
When all audio tracks have been disabled and the mixer is running idle before the output stream is placed in standby,
the mixer sometimes fails to write to the output stream on time to avoid underrun.
This is because the sleep period used to wait before the next write to output stream is too close to the actual buffer duration.
In fact this sleep time is not critical as if we write too early to the output stream, the kernel driver will wait for free buffers
from the audio DSP DMA and we will sleep anyways.
The fix consists in dividing the calculated wait period by 2 to increase the margin.
Change-Id: I5730887dc2ccce2a511bc858494a6f7da6b392a0
udioFlinger.cpp
|
| cc759794489c2c39aaef229fba4ace32e394bb81 |
29-Jul-2010 |
Eric Laurent <elaurent@google.com> |
am 45dc4f82: am 493941b8: Allow creation of an audio effect on a session with no audio tracks.
Merge commit '45dc4f82a00e52b12389b22a7cfbbee5609e8e28'
* commit '45dc4f82a00e52b12389b22a7cfbbee5609e8e28':
Allow creation of an audio effect on a session with no audio tracks.
|
| 39e94f8f723d445447fdee0822291e664b631f60 |
28-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Allow creation of an audio effect on a session with no audio tracks.
This is necessary to allow creating and enabling an effect attached to a particular player
session before the playback is started. As a matter of fact, the implementation of the mediaplayer
does not create the AudioTrack before playback starts.
Change-Id: I1266e8885f9d756acc949303321aaac0fbf83e34
udioFlinger.cpp
udioFlinger.h
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| ed0de3c123aaa86147993721d4794648a65aab75 |
28-Jul-2010 |
Eric Laurent <elaurent@google.com> |
am 6fa88882: am 62d83a0c: Merge "Audio effects: modified command() parameter types." into gingerbread
Merge commit '6fa8888254deb4961dc64c6ac03dc103c9cd8bcf'
* commit '6fa8888254deb4961dc64c6ac03dc103c9cd8bcf':
Audio effects: modified command() parameter types.
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| 25f4395b932fa9859a6e91ba77c5d20d009da64a |
28-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Audio effects: modified command() parameter types.
The type of the cmd, cmdSize and *pReplySize parameters of the effect control interface command()
function have been modified from int to uint32_t. This is more consistent with their role.
Change-Id: I84d289fc262d6753747910f06f485597dfee6591
udioFlinger.cpp
udioFlinger.h
|
| 8184a5734690b30f4de0f6b6c16daf1e089f67df |
20-Jul-2010 |
Eric Laurent <elaurent@google.com> |
resolved conflicts for merge of dd206093 to master
Change-Id: I21dd2321a4839d034d49092baccbf40986f17dae
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| 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
udioFlinger.cpp
udioFlinger.h
udioPolicyManagerBase.cpp
udioPolicyService.cpp
udioPolicyService.h
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| bbf6dcd2aa8eb877e6412aaade25a077f62a6284 |
19-Jul-2010 |
Eric Laurent <elaurent@google.com> |
am 3c6ea955: am 7203272f: Merge "Fixed problems in audio effect volume control." into gingerbread
Merge commit '3c6ea95553d3a0e416874c583a41efff3f6fbc57'
* commit '3c6ea95553d3a0e416874c583a41efff3f6fbc57':
Fixed problems in audio effect volume control.
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| f997cabca292d70d078ae828e21c28e6df62995f |
19-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Fixed problems in audio effect volume control.
Fixed the following problems in audio effect volume control in AudioFlinger:
- Make sure that the volumes returned by EffectChain::setVolume_l() are correct even is
no change is detected since last call
- Do not use isEnabled() to validate volume control but mState >= ACTIVE instead as the volume control
must be also active in STOPPING and STOPPED states.
Change-Id: Id62da3164fad500ee8a5efd6cd78c77e8fdcb541
udioFlinger.cpp
udioFlinger.h
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| 521bcb0ff5054e5be40f5194026da3463504ae3b |
16-Jul-2010 |
Eric Laurent <elaurent@google.com> |
am f16b3f10: am b0374a0e: Merge "Several improvements in audio effects volume control." into gingerbread
Merge commit 'f16b3f104f60738447db60bcf2c44c7610e17199'
* commit 'f16b3f104f60738447db60bcf2c44c7610e17199':
Several improvements in audio effects volume control.
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| cab112421da6e8eac19ffddbbe3d76067cffee78 |
15-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Several improvements in audio effects volume control.
- Fixed crash when deleting an effect chained before an effect having volume control
- Changed EFFECT_FLAG_VOLUME_CTRL to implicitely include EFFECT_FLAG_VOLUME_IND
(not need to set both in effect descriptor).
- Volume control changes from one effect to another if needed according to effect enable state
- EFFECT_CMD_SET_VOLUME is only sent when their is an actual change in volume
Change-Id: Ieebaf09157e2627366023569d95516646e03e26c
udioFlinger.cpp
udioFlinger.h
|
| c0b9c87b2581af430e143822918935d15ac69911 |
15-Jul-2010 |
Mathias Agopian <mathias@google.com> |
am c74034b6: am 24651682: added BinderService<> template to help creating native binder services
Merge commit 'c74034b6bf0940dc80c4d4efa1e677ad0d2dfd6c'
* commit 'c74034b6bf0940dc80c4d4efa1e677ad0d2dfd6c':
added BinderService<> template to help creating native binder services
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| 094c96da874d84eae46f76cc036fa7b7993eb999 |
15-Jul-2010 |
Mathias Agopian <mathias@google.com> |
am 78983a91: am 4055b780: Merge "move native services under services/" into gingerbread
Merge commit '78983a9133d3dd3f08b1ec462a7e2f9e7bfa9e2f'
* commit '78983a9133d3dd3f08b1ec462a7e2f9e7bfa9e2f':
move native services under services/
|
| 5462fc9a38fa8c9dff434cd53fa5fb1782ae3042 |
15-Jul-2010 |
Mathias Agopian <mathias@google.com> |
added BinderService<> template to help creating native binder services
Change-Id: Id980899d2647b56479f8a27c89eaa949f9209dfe
udioFlinger.cpp
udioFlinger.h
|
| 65ab47156e1c7dfcd8cc4266253a5ff30219e7f0 |
15-Jul-2010 |
Mathias Agopian <mathias@google.com> |
move native services under services/
moved surfaceflinger, audioflinger, cameraservice
all native services should now reside in this location.
Change-Id: Iee42b83dd2a94c3bf5107ab0895fe2dfcd5337a8
2dpAudioInterface.cpp
2dpAudioInterface.h
ndroid.mk
udioBufferProvider.h
udioDumpInterface.cpp
udioDumpInterface.h
udioFlinger.cpp
udioFlinger.h
udioHardwareGeneric.cpp
udioHardwareGeneric.h
udioHardwareInterface.cpp
udioHardwareStub.cpp
udioHardwareStub.h
udioMixer.cpp
udioMixer.h
udioPolicyManagerBase.cpp
udioPolicyService.cpp
udioPolicyService.h
udioResampler.cpp
udioResampler.h
udioResamplerCubic.cpp
udioResamplerCubic.h
udioResamplerSinc.cpp
udioResamplerSinc.h
|