| 07eee80b3422c0304491f24bbc415deb0badbefc |
|
22-Jun-2016 |
Andy Hung <hunga@google.com> |
Limit cached timestamp use if HW fails to provide them
On some devices, if the HDMI cable is disconnected during use,
cached timestamp info can become very old.
Bug: 29423451
Change-Id: Ieecf5e98bad14d00b4f5f14b032c2cf66ae94ceb
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 69488c4ef115c9de52c85f4fcae27c7774720298 |
|
17-May-2016 |
Andy Hung <hunga@google.com> |
AudioTrack: Fix timestamp startup condition
Suppress timestamps without valid kernel information or
a server update.
Update timing of server timestamp location to better indicate
when the server pull occurs.
Update write timing logic to suppress unneeded time calls
and to ensure proper throttling.
Bug: 28771691
Change-Id: I2cd25dad638ca1d0b1355d87d175d908f3bddb26
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 9ea77cdba8c422940adb57a790b44ac04fe0353f |
|
07-Apr-2016 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: Pass pid of process creating track or opening record
AudioFlinger allocates a Client heap for each unique pid.
If two applications use mediaplayer APIs, the same Client heap
is reused as the pid used is not the application pid but that
of mediaserver. With this change, the pid of the application
pid is used to decide the heap to be used.
Bug: 23525542
Bug: 28772898
Change-Id: I31a695b0b321eff6e2aca80c3bc4aeb3e1cd9ac7
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 6d7b119a416c9f10288051e562f294365e5d954c |
|
08-May-2016 |
Andy Hung <hunga@google.com> |
Improve timestamp jitter fix
Use last valid kernel timestamp on server side, not client side.
Bug: 28465713
Change-Id: I33590d6922980f288355f947bc56279245058429
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 7aa45dd7fc5c2bd959f1571231556ac06b7ec8ae |
|
03-May-2016 |
Andy Hung <hunga@google.com> |
Merge "AudioTrack: Fix timestamp jitter" into nyc-dev
|
| aebe9dc29988c94a774208ba659c4f156d274d79 |
|
02-May-2016 |
Glenn Kasten <gkasten@google.com> |
Normal tracks use triple buffering if resampling
Bug: 26230944
Change-Id: I40f6cd7147db1be95f297af406c77c6404a76f37
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| b01faa31462e19957ae3c0c3ada07c9576a9494f |
|
27-Apr-2016 |
Andy Hung <hunga@google.com> |
AudioTrack: Fix timestamp jitter
When HAL out_get_presentation_position returns error,
use server position adjusted for latency.
Bug: 28250436
Change-Id: I3dbcb9b7c4e56d34e7e161d1a02d8f64afd602b9
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| ea38ee7742e799b23bd8675f5801ef72f94de0f4 |
|
18-Apr-2016 |
Glenn Kasten <gkasten@google.com> |
Can now indicate the desired number of notifications (sub-buffers)
per fast track buffer when using the default buffer size.
There is no change for normal tracks, or fast tracks with a non-default buffer size.
Also fix related bugs:
Notification period was not set correctly when fast multiplier is not 1.
Incorrect warning when client adjusted notification frames.
Bug: 27819623
Bug: 28117362
Change-Id: Ifc2121e874f51718cd56ec04e4bd5f89f4963132
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 0c4a97e6e1470b2d8b8810792af8d66689020c51 |
|
13-Apr-2016 |
Andy Hung <hunga@google.com> |
Merge "NuPlayer: Reserve buffer for 8x max speed" into nyc-dev
|
| f2c87b3f88d987d2af6322e9763eb8224a2bae48 |
|
08-Apr-2016 |
Andy Hung <hunga@google.com> |
NuPlayer: Notify if drain message time is unexpectedly long
Bug: 27940058
Change-Id: Id8897c30e21b6b9de545a710e7766ffceb491455
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| ff874dc957f9ea70d87f4d627bf903e1fc86d58b |
|
12-Apr-2016 |
Andy Hung <hunga@google.com> |
NuPlayer: Reserve buffer for 8x max speed
Affects Photos with Bluetooth and 240x recording.
Bug: 26986060
Change-Id: I228223221f9811e05c274d64cd42ac104db04c36
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| e1e9846a786a0d77a13b44f8e798e03b4cba9d77 |
|
12-Apr-2016 |
Andy Hung <hunga@google.com> |
AudioTrack: Reset position on flush for extended timestamps.
Bug: 27940058
Change-Id: If91320669e2e84803e601137b1ce66b361b129f1
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| e13f8a603867280de9b02ce4839202a182556416 |
|
30-Mar-2016 |
Andy Hung <hunga@google.com> |
NuPlayer: Play out remaining audio data when recycling isn't possible
Don't flush it immediately.
Bug: 26530291
Change-Id: I0427633402fa251ec62e2b70e47a0a7b451d3ea2
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| ea2b9c07b34079f0dbd8610a511e006e69a15adc |
|
13-Feb-2016 |
Andy Hung <hunga@google.com> |
Implement client playback timestamps with 64 bit accuracy
Provide server timestamps if the HAL doesn't provide it.
Provide monotonic - boottime translation.
Bug: 17472992
Bug: 26682703
Bug: 27749434
Change-Id: I6c9b213d9f9284092e34d57f52870e02c72df62a
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 5fe2151459fd6c88958c5ebd71f28004d49d8b65 |
|
29-Mar-2016 |
Andy Hung <hunga@google.com> |
Merge "Read from proxy for PCM direct and offloaded track position" into nyc-dev
|
| 7a490e7ab39ef0b22baf24b03f2463c46dc068ed |
|
23-Mar-2016 |
Andy Hung <hunga@google.com> |
Read from proxy for PCM direct and offloaded track position
Bug: 27776737
Change-Id: I721bf2b4d46271bf84ad9fc29ff5c4ecd4759206
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 9b3359feabbf138bb965da5173434da78bfe0929 |
|
24-Mar-2016 |
Haynes Mathew George <hgeorge@codeaurora.org> |
libmedia: Initialize state vars to avoid possible crash
If playback or record track creation fails on server side,
some state is left initialized incorrectly.
This can cause a crash when the object is destroyed
because of a reference to a NULL object.
Initialize state vars properly in the initialization list
to avoid this behavior.
authored-by: Zhou Song <zhous@codeaurora.org>
CRs-Fixed: 992608
Change-Id: If8c3611e3229c0c1b14b81285e07b9357fec7658
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 79629f0716f04e79064256362be56730077b8c50 |
|
24-Mar-2016 |
Andy Hung <hunga@google.com> |
AudioTrack: Do not resume callback thread on start failure
Bug: 27673375
Change-Id: I1df1db51e75cee57f19824ac6506bf02000714fd
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| fc1337d079ba7e6ffd7ce07706cfeeeb355c0200 |
|
18-Mar-2016 |
Phil Burk <philburk@google.com> |
Merge "AudioTrack: fix stall if setBufferSizeInFrames() called before play()" into nyc-dev
|
| e8972b0a27fac327c997fceb20d9abae1dc1d899 |
|
04-Mar-2016 |
Phil Burk <philburk@google.com> |
AudioTrack: fix stall if setBufferSizeInFrames() called before play()
The server was waiting for a full buffer.
But the buffer was only getting partly filled.
So the stream was not starting.
The fix involves having the server look at the adjustable threshold.
Bug: 27505889
Change-Id: I5dbf686413e670dacbbecc9e0f838744e465f44f
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| d848eb48c121c119e8ba7583efc75415fe102570 |
|
08-Mar-2016 |
Glenn Kasten <gkasten@google.com> |
Use audio_session_t consistently
Bug: 27562099
Change-Id: I328d8226191386b163f2ace41233607294c50dcd
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 4d231dc0ee34380956c71cbe18a750e487a69601 |
|
12-Mar-2016 |
Eric Laurent <elaurent@google.com> |
audio: Fix race condition in AudioTrack underrun.
When audio flinger mixer removes an AudioTrack from the
active list in case of underrun, it is possible that the
client has written a full buffer just after the underrun detection and
is blocked waiting for more space to write. In this case, the client
will never detect the DISABLED flag and the track never be restarted.
Also implement missing DISABLE flag detection in server side audio tracks
(OutputTrack and PatchTrack).
bug: 27567768
Change-Id: I8d0753429d4113498258b1f61bd8ac5939a612f0
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| ae34ed2bcc9fa49a22fe7497327adb689ef27e8d |
|
28-Jan-2016 |
Haynes Mathew George <hgeorge@codeaurora.org> |
AudioTrack: Use original flags during track recreation
Also the OFFLOAD and DIRECT flags aren't denied by server.
authored-by: Pavan Chikkala<pavanc@codeaurora.org>
Bug: 27554925
Change-Id: I6fd6b129928af4513bb0cba271019288605310f8
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 7fd0422fbd17af3b24eb04421d37fce50f3826e2 |
|
02-Feb-2016 |
Glenn Kasten <gkasten@google.com> |
Sample rate 0 means a route-dependent rate
Remove check for primary _output_ [sic] sampling rate for fast capture.
Clean up AudioRecord handling of frame count and sample rate.
Clean up AudioTrack handling of notification period updates.
Make AudioRecord and AudioTrack more similar in order of operation, comments, and whitespace.
Bug: 25641253
Bug: 21019153
Change-Id: I24a6677945987fc39a9bf93f70357e4bc7410f98
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| eeecb980ff4c202d0a3c4b0bfe040dce2f73336d |
|
26-Feb-2016 |
Glenn Kasten <gkasten@google.com> |
Add use for audio_unique_id_t
Bug: 25641253
Bug: 21019153
Change-Id: I65dc128e760c245f3d90559635a8981b186c87d7
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 818e7a32ce3633980138aff2c2bfcc5158b3cfcc |
|
17-Feb-2016 |
Andy Hung <hunga@google.com> |
Implement server side playback timestamps with 64 bit accuracy
Provide server timestamps if the HAL doesn't provide it.
Provide monotonic - boottime translation.
Integrate record timestamps and playback timestamps together.
Bug: 17472992
Bug: 22871200
Bug: 26400089
Bug: 26682703
Change-Id: If1974f94232fcce7ba0bbcdf63d9e54ed51918ff
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 6ae5843c281301a9ffd1059d185620a9337e15a2 |
|
17-Feb-2016 |
Andy Hung <hunga@google.com> |
Enable FastTrack timestamps
Bug: 26413951
Change-Id: Id54c32cb0f0be83d443f6cf1007181ebc461c586
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 8ce8e264d0bf09ec131e4a7701fe1ebc3a678f8b |
|
16-Feb-2016 |
Eric Laurent <elaurent@google.com> |
Revert "Enable FastTrack timestamps"
This reverts commit b3a486fecddc9e9264fe63d356eca0812493db27.
Reverting while investigating the root cause.
Bug: 27186414
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| b0a144d705a021994628d136a170a6d2bd4282e7 |
|
12-Feb-2016 |
Phil Burk <philburk@google.com> |
Merge "AudioTrack: support ENCODING_IEC61937" into nyc-dev
|
| fdb3c07db5d44535eb8c3ec46dc78ad8446c01eb |
|
09-Feb-2016 |
Phil Burk <philburk@google.com> |
AudioTrack: support ENCODING_IEC61937
Set DIRECT flag.
Use audio_has_proportional_frames() instead of audio_is_linear_pcm()
where appropriate.
Bug: 24541671
Bug: 20891646
Bug: 26373761
Change-Id: Ia32036b18683b084d6c9887593df87397ea0afd9
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| b3a486fecddc9e9264fe63d356eca0812493db27 |
|
05-Aug-2015 |
Andy Hung <hunga@google.com> |
Enable FastTrack timestamps
Bug: 26413951
Change-Id: I99a69241a0bcd69f6bfebf785cac6d9ee1e2cd5f
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| d79072e9dff59f767cce2cda1caab80ce5a0815b |
|
06-Jan-2016 |
Glenn Kasten <gkasten@google.com> |
Remove TimedAudioTrack and associated code
Bug: 8278435
Change-Id: I095c1a4888e645e14d93b0b15fbef4524a831ca1
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| bbe98f009fc2c3181a8d4ab71b3e145f76c01755 |
|
17-Jan-2016 |
Phil Burk <philburk@google.com> |
Merge "AudioTrack: Add getUnderrunCount()"
|
| 27e583ba1d0d4fdc11d6eeee0c416e727045cbbc |
|
17-Jan-2016 |
Phil Burk <philburk@google.com> |
Merge "AudioTrack: add setBufferSizeInFrames()"
|
| 33ff89ba94a527e4293ee5349da01483252d5c83 |
|
30-Nov-2015 |
Phil Burk <philburk@google.com> |
AudioTrack: lower Java latency
Add AUDIO_FLAG_LOW_LATENCY to enable Fast track.
Change-Id: Ib1bff635afaa57af7754cb751b57c6afc4c2ab2b
Bug: 21019153
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| c0adecb800b197cb8c028513130ebabf2d0f37ba |
|
08-Jan-2016 |
Phil Burk <philburk@google.com> |
AudioTrack: add setBufferSizeInFrames()
Also add getBufferCapacityInFrames().
These can be used to dynamically raise or lower latency.
Bug: 21019153
Signed-off-by: Phil Burk <philburk@google.com>
Change-Id: I02ca7f6f5cc4e089fcd81cc8a2b6ff234e0381a8
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 2812d9ea3a3a33142dd8e23c9d949c498d6f7a12 |
|
04-Jan-2016 |
Phil Burk <philburk@google.com> |
AudioTrack: Add getUnderrunCount()
This allows an app to detect application-level output glitches.
Underrun counts survive track recreation.
Change-Id: I8eb14e92f6fc1007718a29b0666ab51ace30cdb8
Bug: 25641253
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 90e8a97dd5c9d391d7a2ea6a2290ed976f928379 |
|
10-Nov-2015 |
Andy Hung <hunga@google.com> |
Use modulo position variables in AudioTrack and AudioRecord
More type safety, plus correct treatment for sanitization.
Bug: 25569906
Bug: 25232421
Change-Id: Id852277b81a1792c5e67392cd74bc39cba7ed1ad
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 7095f876565f8c52bafe02e7c6ba04aa0c274dfa |
|
09-Nov-2015 |
Chad Brubaker <cbrubaker@google.com> |
Merge "Whitelist getTimestamp from integer sanitization"
am: 51cd9a441e
* commit '51cd9a441ed0acf6e3b9c221e9b7f9424073191f':
Whitelist getTimestamp from integer sanitization
|
| ec3806a8be1a014631b39df59edd596f8eb5be01 |
|
09-Nov-2015 |
Chad Brubaker <cbrubaker@google.com> |
Whitelist getTimestamp from integer sanitization
The timestamp math relies on modulo 32 bit math, whitelist this function
for now until we can find a better way to tackle this.
Bug: 25490803
Change-Id: I3e2b19db023b4622f5884933feab96363756e8c6
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 3845ba35a0bf364f30bd46029b89e7b5fff28793 |
|
28-Oct-2015 |
Dan Austin <danielaustin@google.com> |
Merge "Refactor setPlaybackRate to avoid benign unsigned integer overflow." am: a38594e92f
am: 9b698dcfb4
* commit '9b698dcfb41cbadd4a21d09eede0de672912959d':
Refactor setPlaybackRate to avoid benign unsigned integer overflow.
|
| 9b698dcfb41cbadd4a21d09eede0de672912959d |
|
28-Oct-2015 |
Dan Austin <danielaustin@google.com> |
Merge "Refactor setPlaybackRate to avoid benign unsigned integer overflow."
am: a38594e92f
* commit 'a38594e92fea7f88c5edad98a4992b5f1b688436':
Refactor setPlaybackRate to avoid benign unsigned integer overflow.
|
| e34eae2de7bd6dcba5dea132dabb09f63ac46adc |
|
28-Oct-2015 |
Dan Austin <danielaustin@google.com> |
Refactor setPlaybackRate to avoid benign unsigned integer overflow.
There is a check that results in a benign unsigned integer overflow.
This has been refactored to avoid the unsigned integer overflow.
Bug: 25327431
Change-Id: Ib112a7cd585c680f13e4bee3d7e9f45da7d66f1d
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 2eb225313de241d02979c66e2730983901ad05ea |
|
23-Oct-2015 |
Glenn Kasten <gkasten@google.com> |
Merge "Workaround for busy wait bug" into mnc-dr-dev am: d34cf6d95c
am: 6a92190261
* commit '6a9219026124f9075c03a861101cea985f01cdce':
Workaround for busy wait bug
|
| 606fbc1030d686bf1f8dc03d14a79342aca2ca41 |
|
23-Oct-2015 |
Glenn Kasten <gkasten@google.com> |
Workaround for busy wait bug
Underlying bug 25195759 is not yet fixed.
Bug: 24263351
Change-Id: I04b644246640054d38c8de79e0eb80fcbfef0e48
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| c67c65882f7cf1d3ae617bff41ff9c7ec8a0fd27 |
|
14-Oct-2015 |
Glenn Kasten <gkasten@google.com> |
am 5116febf: am cfda8c43: Merge "AudioTrack: Skip callback EVENT_STREAM_END on DEAD_OBJECT." into mnc-dr-dev
* commit '5116febf966875c19a0b137a3deef0ba0ebe42df':
AudioTrack: Skip callback EVENT_STREAM_END on DEAD_OBJECT.
|
| 39609a0a6d95d003b0f237e6c139b070b4df7071 |
|
04-Sep-2015 |
Andy Hung <hunga@google.com> |
AudioTrack: Skip callback EVENT_STREAM_END on DEAD_OBJECT.
Send only EVENT_NEW_IAUDIOTRACK to initiate teardown.
Bug: 23750452
Change-Id: Ib5dcd578b5b6a6a1d91a03e58ef8da0acd994243
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 7d7a11ca73c9684b63275e82b327f6f90fcc114e |
|
07-Oct-2015 |
Glenn Kasten <gkasten@google.com> |
am 9ea0680e: am b177e8a7: Merge "AudioTrack: Prevent stop() from reissuing last marker event" into mnc-dr-dev
* commit '9ea0680e4d77d21fb0698fec3ccd7779213eca62':
AudioTrack: Prevent stop() from reissuing last marker event
|
| 61be84167c1a11c8a030e00a2b79b2bc4fdf617f |
|
06-Oct-2015 |
Andy Hung <hunga@google.com> |
AudioTrack: Prevent stop() from reissuing last marker event
Avoid a duplicate marker event race condition (1 in 30 or less)
by clearing marker reached in start() not stop().
Bug: 24497521
Change-Id: I9520d063c7d173d2e642174bf60a2bfe75edf085
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 54b3c3c7acf18226f6abf788a220837bc5737677 |
|
29-Sep-2015 |
Dan Austin <danielaustin@google.com> |
am 6c4cf7fb: Merge "Fix benign overflow in AudioTrack"
* commit '6c4cf7fb4c8a48450f8290380710ddeb6a8cc625':
Fix benign overflow in AudioTrack
|
| 039c27aeaee809182fc1491311ca6dcbc1ec441b |
|
24-Sep-2015 |
Chad Brubaker <cbrubaker@google.com> |
Fix benign overflow in AudioTrack
two uint32_t's were being used in a computation that could be negative,
cast to int32_t before the subtraction to prevent incorrect overflow
detection.
Change-Id: I33c5ef79a0ebbba055daa0ea041b42229d0c3152
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| e9fa23fe940c287bf2f6e8b28f4cf9161818cb80 |
|
23-Sep-2015 |
Chad Brubaker <cbrubaker@google.com> |
resolved conflicts for 1f8749f2 to stage-aosp-master
Change-Id: Idda60a9d39f9be81e0c314f06fc080ad7fa35c7b
|
| 20b82e0cb5c615a4f88f245c12fec9ee0ec49894 |
|
22-Sep-2015 |
Chad Brubaker <cbrubaker@google.com> |
Fix benign unsigned overflow in AudioTrack
An unsigned overflow could occur in timestamp computation when
mTimestamp < mServer.
Change-Id: I9e01897478a6d481dd6fb7303d325617466497cf
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 97c9f4fd7bc31e2968d816402659ba1c64f10b42 |
|
12-Aug-2015 |
Eric Laurent <elaurent@google.com> |
NuPlayerRenderer: handle error when resuming an offloaded track
Make sure that an offloaded audio track is torn down if
start() returns an error in onResume().
This makes sure that a track invalidated due to a potential
audio path change while paused is re created on the correct
output.
Bug: 22256441.
Change-Id: I6c7bfec6e7322415daffc0451bab46d0c558522a
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 2c7d0f2013203588f9307166cad42902eb5a676f |
|
24-Jul-2015 |
Andy Hung <hunga@google.com> |
Merge "Fix restore position warning and static track adjustment" into mnc-dev
|
| d7bd69e4d1e74f25413721804d5290c53ea87f2b |
|
24-Jul-2015 |
Andy Hung <hunga@google.com> |
Fix restore position warning and static track adjustment
Remove unnecessary restoreTrack_l() position warning when
AudioTrack output device changes.
Also, fix rare getPosition() offset error if a static track is
continuously playing a long time (> 24 hours at 48khz sampling)
and its output device changes.
Bug: 22524122
Bug: 21699132
Change-Id: I5a35c13d62ff9f0702b09d5cd9362c730364b18c
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| a8d0890ca969ec40fdcf1cfcbeef85c0c379cfaf |
|
22-Jul-2015 |
Andy Hung <hunga@google.com> |
Fix race condition in AudioTrackThread::wake
Bug: 22533684
Change-Id: I2f46770dca44fc9dae41e067d3bec893c42a826e
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 55a2dcce3a409393dc51ab2eb433914331f2df51 |
|
23-Jun-2015 |
Ricardo Garcia <rago@google.com> |
Merge "Using centralized isAudioPlaybackRateValid to validate parameters" into mnc-dev
|
| 6c7f062d3149d6890daaee64828959ad6f61ea54 |
|
01-May-2015 |
Ricardo Garcia <rago@google.com> |
Using centralized isAudioPlaybackRateValid to validate parameters
Centralized validation code
bug: 20701446
Change-Id: I9d9941c7639c05b2afe069ff4f858c693c910bfe
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 6653c935d2518a713587b3887ca09aa09ebfa7fd |
|
08-Jun-2015 |
Andy Hung <hunga@google.com> |
Return DEAD_OBJECT if getTimestamp cannot restore track
Bug: 21699132
Change-Id: I90443f8674ed949e2546048b231be75cd6fe6615
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 1f1db8356b599bc40703c907fb69e6e539343532 |
|
08-Jun-2015 |
Andy Hung <hunga@google.com> |
Fix AudioTrack comments relating to use of restoreTrack_l()
Bug: 21699132
Change-Id: Ib0d029a5e28676aeffbbbafc88c52a17367a413c
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| c8e09c610fabc7390297ecb48e939acbdfe27325 |
|
04-Jun-2015 |
Andy Hung <hunga@google.com> |
Improve AudioTrack offload timestamp startup glitch detector
New or existing glitch behavior for Nexus 5 offload audio:
we receive several 0 timestamps,
then we get a stale timestamp (very large),
then a few ms later we get a correct nonzero timestamp.
We attempt to hide the glitch because the retrograde timestamp
correction makes the glitch "sticky".
Bug: 21633313
Change-Id: I39153af718c151f9435e7d315651a811f72743da
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| a7f03353d5f172016f324e2a01f301cca6794152 |
|
01-Jun-2015 |
Andy Hung <hunga@google.com> |
Compute sleep time when AudioTrack client callback returns no PCM data
Callbacks can go into a sleep-wait cycle if the client/app is unable to
deliver data. This can happen if the buffer is large, or if
the client/app cannot keep the buffer filled, or upon a stream end condition.
We improve the sleep time computation for AudioTrack PCM callbacks.
This minimizes the number of callbacks to NuPlayerRenderer.
Bug: 21198655
Change-Id: I4247798a6638def2f0d8f1b46f60323482065cb2
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 9f9e21ed9a342e0ca945818bdc6c0c4fb3bfb9bd |
|
01-Jun-2015 |
Andy Hung <hunga@google.com> |
Store server latency, sample rate, framecount information in AudioTrack
Bug: 21198655
Change-Id: I24590196642a97ea5d61bc6356f0aff782bd37d3
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| faeb0f291330134dc4468359a36e099aae508449 |
|
21-May-2015 |
Ronghua Wu <ronghuawu@google.com> |
libmediaplayerservice: try to open audio sink in offload mode in error.
Bug: 19061432
Bug: 21370108
Change-Id: Iaa757555ef37fd1ac87b6e2d5a9969bb58cc5ebc
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 296fb13dd9b5e90d6a05cce897c3b1e7914a478a |
|
01-May-2015 |
Eric Laurent <elaurent@google.com> |
Implement audio device callback
Add class AudioSystem::AudioDeviceCallback notifying
AudioSystem clients upon device selection change on a given
input or output thread.
Maintain a list of installed callback per I/O handle in AudioSystem
and call registered callbacks when an OPEN of CONFIG_CHANGED event
is received on IAudioFlingerClient::ioConfigChanged().
Add methods to AudioTrack and AudioRecord to add and remove device
change callbacks.
Add methods to AudioTrack and AudioRecord to query currently selected
device.
ioConfigChanged() events now convey the audio patch describing
the input or output thread routing.
Fix AudioRecord failure to start when invalidation is
handled by start().
Change-Id: I9e938adf025fa712337c63b1e02a8c18f2a20d39
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 4c5a3675ed6b429dfaa2707acf2b3faf2808baf2 |
|
01-May-2015 |
Phil Burk <philburk@google.com> |
AudioTrack: reduce retrograde motion spew
Change-Id: I96aced52b136ceea924aa3d9ef56374dd4c49784
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 3a474aa67fc31505740526dd249d96204c08bf79 |
|
25-Apr-2015 |
Lajos Molnar <lajos@google.com> |
stagefright: support setting/getting playback/sync config in MediaSync
Bug: 18249558
Bug: 19666434
Bug: 20057497
Change-Id: I5868b17423d7c20cfaf4a399f3eb67bfba440605
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 5aab9254a5c4f1951c12216a71550ab95b4adeaa |
|
30-Apr-2015 |
Phil Burk <philburk@google.com> |
AudioTrack: fix build
Using format %llu did not work for uint64_t on 64-bit systems!
Change-Id: I727094e460141b9da3e433ceaa95e069dad948bc
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| dc0ac804721efff054ec9720d53e93ecca86cf5e |
|
30-Apr-2015 |
Phil Burk <philburk@google.com> |
Merge "AudioTrack: fix spurious retrograde messages" into mnc-dev
|
| 8c7e6dac6f5eb38cef627dab92eac8b38513450c |
|
22-Apr-2015 |
Eric Laurent <elaurent@google.com> |
audio policy: session routes continued.
- Clear session routes when client process dies.
- Enforce the route only when the requesting session is active.
- Fix requested route not working if an output mix change is required
(e.g forcing to SPEAKER when the default route is USB or A2DP).
- Make sure all sessions sharing the strategy with a rerouted session
have the same route (needed for volume control consistency)
Change-Id: I0ab347a8fb97e73e2c5965374544c5f4fe509ef1
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 1b42097f38e72574ed853a35f4e8a66e4739c421 |
|
22-Apr-2015 |
Phil Burk <philburk@google.com> |
AudioTrack: fix spurious retrograde messages
The retrograde motion was confused by some positions coming from
the DSP on offloaded tracks. So the retrograde check was moved up
into AudioTrack.cpp. This allows us to take advantage of the checks
for invalid positions based on timing.
Bug: 2047891
Change-Id: Ifcad2349201443a7f1711347c203297100449536
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 983f2e02b342ef3ac4004471cbfd4cd8d5aecf7d |
|
22-Apr-2015 |
Ricardo Garcia <rago@google.com> |
Merge "Use AudioPlaybackRate to hold TimestretchBufferProvider parameters"
|
| 5a8a95de6dad1a3bcf3da5a37b35766e89086e13 |
|
18-Apr-2015 |
Ricardo Garcia <rago@google.com> |
Use AudioPlaybackRate to hold TimestretchBufferProvider parameters
Use this struct to handle the parameters for TimestretchBufferProvider all
across the system.
Add stretch mode and fallback mode to TimestretchBuffer Provider.
Change-Id: I19099924a7003c62e48bb6ead56c785cb129fba2
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 493404d8c396e15ed73455acd39aa0f1940996e2 |
|
22-Apr-2015 |
Eric Laurent <elaurent@google.com> |
audio policy: fix AudioTrack output device selection
Have AudioTrack::setOutputDevice() set CBLK_INVALID flag instead
of calling restoreTrack_l(). This allows restoreTrack_l() to be called in
a safe context.
Allow device change while the AudioTrack is active by forcing a new
device selection in startOutput() if the output route for this
session was changed.
Remove some warnings.
Change-Id: I2d921a63c9bfa0e122233645e2d6d39f95f5f17d
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| f1063e2659cad01ec334a265c31734c63c319cc4 |
|
18-Apr-2015 |
Lajos Molnar <lajos@google.com> |
libmedia: fix all warning, make warnings errors, use clang
Change-Id: Ic00d2c5d0bbb1605e96666e25c9ccc22bea6d3ff
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 26145643ce22d797b3b1675c82c47a2d8c79ecaa |
|
16-Apr-2015 |
Andy Hung <hunga@google.com> |
Work around lack of pitch adjustment in Timestretcher
Change-Id: I3e057b97c250a826133248b6aa1ff9ed677d88df
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| b23cabdb71b98dc5e852f5d06be271696fea92c7 |
|
14-Apr-2015 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioRecord::obtainBuffer() can return nonContig like AudioTrack"
|
| 551b5355d34aa42890811fc3606d3b63429296cd |
|
20-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
AudioRecord::obtainBuffer() can return nonContig like AudioTrack
And fix error handling in both AudioRecord and AudioTrack.
Change-Id: I4853d736cc6d2c05de5b18b0eac62716edfa62ec
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 8edb8dc44b8a2f81bdb5db645b6b708548771a31 |
|
27-Mar-2015 |
Andy Hung <hunga@google.com> |
Add playback rate to AudioTrack
Bug: 19196501
Change-Id: I6411e1d3ce652b711a71a6d9df020cb5f60d4714
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| aa9811945f575614b3482d09e4d969792701cebb |
|
21-Mar-2015 |
Paul McLean <pmclean@google.com> |
Adding explicit routing API to AudioTrack
Change-Id: I40c048c7644c46f4e4f7103875206c0785c4b1dc
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 5cbb5785a58655e21f5f7805274b8c78b986216b |
|
28-Mar-2015 |
Andy Hung <hunga@google.com> |
Reject sample rate changes for fast tracks
This is needed for a SoundPool fix.
Bug: 19970735
Change-Id: I46b69e059aabf0d06b16b31ea436bb71f7146750
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| bfd318402924414fbde4d31c9d23cc423f10b630 |
|
20-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
set() in AudioRecord and AudioTracks needs no lock
Change-Id: I9d61b3d117e9b21fbbfad772d8981a5abb8fafae
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 4c36d6f48f4eecacc148317f775f65961a4f03d2 |
|
20-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Logging
Move log of latency to where it is first computed.
Log more parameters when a fast track is denied by client.
Log session ID in destructor.
Remove obsolete ALOGV that were for debugging a particular problem.
Log more parameters of set().
Change-Id: I0156bceaf397f2faa053214572c57136705685ee
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 4408d8ab70d3385ac043d0f9a210a971c3a570c2 |
|
23-Mar-2015 |
Andy Hung <hunga@google.com> |
Merge "Fix position when restoring streaming audio track"
|
| 138d6f9b0b77d94773a8a29d3c5bdb78f38ee142 |
|
20-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Update AudioTrack code to match AudioRecord
session ID check
buffer error code check
Change-Id: I092a61adfd249122f9127dceedf67c0a3bdcfe7b
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 40bc906252974d0b389ae4a147232d0c9a97193f |
|
20-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
namespace does not need a closing semicolon
Change-Id: Ie8f9d42fc061f6d558f23b98414e04eb3d14b376
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| ae4b879d04c4b07c62241a8f507c8dce11e628e4 |
|
20-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Declare and initialize in one statement
Change-Id: Ie8593692907e67558f9d7654c9a4bf8cbce99bd8
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 7ccdaad38054474b1d84b7c37ceffd98e9df354d |
|
20-Mar-2015 |
Andy Hung <hunga@google.com> |
Fix position when restoring streaming audio track
Bug: 19802225
Change-Id: I19ca7836bc89627f6cead4961641b94c6b27b5f6
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| c6bd5db9d9cf4bba1649b5b7ddea2d23f5de23a9 |
|
10-Mar-2015 |
Eric Laurent <elaurent@google.com> |
fix flush on HW A/V sync tracks
Fix mismatch between track flags on client and server side
when HW A/V sync is requested.
The audio track was connected to a direct output by the audio policy
manager but the client failed to set the direct flag when creating
the track resulting in the direct flag not being set in audio flinger.
Bug: 19665934.
Change-Id: I0a5257b20bf76c9c4d89bc2ad9b44777b81f13bf
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 1dfe2f9c2d03fc8d0ed0cdfe0b9fb894bc0bcc11 |
|
09-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Allow TRANSFER_OBTAIN to be used for fast tracks
Change-Id: I8fa20c26f076567b38210af4a680fe1cb2eacee4
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| b46f394a85d704dd05287cf9bb77cf86e3c02a38 |
|
09-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
AudioTrack::obtainBuffer() now returns number of non-contiguous frames
Change-Id: I1f61d7e3d057c3254babe456b4aa0f6a1809da55
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 54a8a456c2b4b382b3ffe3d99e40703df79c0717 |
|
09-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
AudioTrack::releaseBuffer parameter is const *
Change-Id: I31ac20863107de05b83cd7f8cd7dd61bed98e456
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 3f02be2ceeaa4b67dc0b1a81aebcfa049276fad8 |
|
09-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
AudioTrack obtain/release buffer comments
Change-Id: I0eb178d1ab4eea7f72f4c55852c19b216ca55f5f
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 0e48d25606c82def035ad10a5b3923767a765cdd |
|
26-Jan-2015 |
Andy Hung <hunga@google.com> |
Change AudioTrack resampling buffers from 3 to 2
Move computation of minimum AudioTrack buffer size to server
for normal streaming PCM tracks.
Use server-side computation to exactly determine requirements
for the resampler to avoid triple buffering.
This reduces latency for normal audio tracks that require resampling,
and makes things consistent with the minimum buffer size.
Change-Id: I2f2ca0e599ee20e16559bc5c5dab61ed100da16c
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| abdb990953ffe94a9dc544aea0bed17ef7d5f484 |
|
13-Jan-2015 |
Andy Hung <hunga@google.com> |
Allow AUDIO_FORMAT_PCM_8_BIT AudioTrack buffers
Previously conversion to AUDIO_FORMAT_PCM_16_BIT was required.
Client shared memory for static tracks and AudioFlinger track creation
and buffer delivery now use native 8 bit PCM data.
Change-Id: I485c07a4971fde9a25442bd43fed95019d39abcc
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 2d85f097d653b21d1ff2c34f0b732c674d20ccc2 |
|
07-Jan-2015 |
Andy Hung <hunga@google.com> |
Fix cumulative error in setPositionNotificationPeriod
Time for AudioTrackThread sleep due to periodic notification
needs to be computed from actual position.
External reported AOSP issue 65807.
Change-Id: Ic46b64588e6c5803afcd8252cde8837eb3fbf6ff
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 3c09c78aabcb9d314dfb400258fb99a37c7b50c9 |
|
30-Dec-2014 |
Andy Hung <hunga@google.com> |
Event driven wake for AudioTrackThread notification changes
Used for setMarkerPosition and setPositionUpdatePeriod.
Change-Id: I0d94b929438a5cd94b295d7c1884f876fae8b5e7
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 53c3b5fc1afe162a8669cb3d27d6bb14e1847e39 |
|
16-Dec-2014 |
Andy Hung <hunga@google.com> |
Improve AudioTrack EVENT_LOOP_END and EVENT_BUFFER_END accuracy
Fix timing computation to consider current position.
Fix EVENT_LOOP_END count accuracy (some could be dropped before)
for non-infinite loop static AudioTracks.
Change-Id: I255f692915b55ab6f08ccd6a2365fee3f5d0a4fe
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 4ede21d9c1f957baf5e561849ff9bbe4bcbefc20 |
|
13-Dec-2014 |
Andy Hung <hunga@google.com> |
Fix loop and position restoration in static AudioTracks
Allow restoration of loop and position.
Make position and loop synchronously readable.
Bug: 17964637
Change-Id: I8cfb5036e665f55fdff5c67d27e1363ce9a8665d
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 9b4615887c23548438fd0d8e3d8f04ac21912850 |
|
02-Dec-2014 |
Andy Hung <hunga@google.com> |
Fix loop and position setting in static AudioTracks
Allow independent setting of position and loop.
Bug: 17964637
Change-Id: I8b3bd97a244b932728b68da7684044f2636984a5
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 85aca658ac7d20584b0647427256df50a5f243ef |
|
17-Dec-2014 |
Eric Laurent <elaurent@google.com> |
am d33712d7: am 145cf5d8: Merge "audioflinger: implement pause/resume for direct outputs" into lmp-mr1-dev
* commit 'd33712d7ec5dcf427cc0be9b7d2ca1c99823c8e6':
audioflinger: implement pause/resume for direct outputs
|
| d1f69b0b17acbd96987ecb2f3378abd394d05903 |
|
15-Dec-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: implement pause/resume for direct outputs
Extend pause/resume support to direct output threads
(was only for offload threads).
If the HAL implements pause/resume, track pause/resume is forwarded to
the HAL.
Pause, flush, resume sequence is respected by executing the HAL
calls in the playback thread (same as offload).
Make sure the track flags on client side are consistent with the
flags on server side.
Bug: 17883772.
Change-Id: I89b360d69818f7a9204bd36e3ec63a79e106ecf1
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| b279f5adfa2fd6f22c87b3a7106fbb62a0b9a5d2 |
|
10-Dec-2014 |
Eric Laurent <elaurent@google.com> |
am c49d9cda: am 2dda40d6: Merge "audio policy: add support for custom mixes" into lmp-mr1-dev
* commit 'c49d9cda49dab8b7dd0376cf56476b3e1895d5c6':
audio policy: add support for custom mixes
|
| 275e8e9de2e11b4b344f5a201f1f0e51fda02d9c |
|
01-Dec-2014 |
Eric Laurent <elaurent@google.com> |
audio policy: add support for custom mixes
Add support for custom mixes in AudioPolicyManager.
Two methods are added to register or unregister a list of custom mixes
with their attributes and format.
getOutputForAttr() and getInputForAttr() first look for a match in
registered mixes before defaulting to normal output/input selection
Remote submix device connection disconnection now takes address into
account to identify the correspnoding custom mix.
Bug: 16009464.
Change-Id: I3f1c2a485a0fb71b1f984ed0adc9b68aa971e408
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| be837c328ae1ea2b193d05aaa3d4214c263b5b77 |
|
22-Nov-2014 |
seunghak.han <seunghak.han@lge.com> |
Track volume should be maintained even if track is restored
If track is restored when it set its volume to a non-unity value (such as mute),
the volume was forced to unity again. Now, the previous volume is restored.
Bug: 18570664
Change-Id: I79f34c6c7d6f535baba813fd3b8ba8e55cac8ad7
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| aef46aa8eff1f975ae5253a334cf0c978427ca86 |
|
02-Dec-2014 |
Glenn Kasten <gkasten@android.com> |
am bde85d05: am 5458756a: Merge "Track volume should be maintained even if track is restored"
* commit 'bde85d05414cdd3888071635dd05e9ff78679d5e':
Track volume should be maintained even if track is restored
|
| e6a9d65845032644b0865b6e3d0b5a820f35d9e4 |
|
22-Nov-2014 |
seunghak.han <seunghak.han@lge.com> |
Track volume should be maintained even if track is restored
If track is restored when it set its volume to a non-unity value (such as mute),
the volume was forced to unity again. Now, the previous volume is restored.
Bug: 18570664
Change-Id: I79f34c6c7d6f535baba813fd3b8ba8e55cac8ad7
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 61280541836947ce0b281b240ffa4c55bbf5bd55 |
|
27-Nov-2014 |
Eric Laurent <elaurent@google.com> |
am 390ef843: am f20c4356: Merge "AudioRecord: use audio attributes instead of audio source." into lmp-mr1-dev
* commit '390ef84373b7edd1397381f780ca235326357094':
AudioRecord: use audio attributes instead of audio source.
|
| 121143d5242a790d0bd01fe1b9cec5d28a1ba1d7 |
|
26-Nov-2014 |
Eric Laurent <elaurent@google.com> |
am bf14c62e: am 72bf901c: Merge "audio policy: new getOutputForAttr() prototype." into lmp-mr1-dev
* commit 'bf14c62e7973ed793a3b00c2c67beca4484c1c7a':
audio policy: new getOutputForAttr() prototype.
|
| 0e47b16ffcf78acb81b0ebf14d058db191f74af0 |
|
26-Nov-2014 |
Andy Hung <hunga@google.com> |
am 1e18795a: am d325005e: Merge "Fix AudioTrack loop mode to play audio from buffer start" into lmp-mr1-dev
* commit '1e18795a4847e9368baa7ecee146d3901b283cd0':
Fix AudioTrack loop mode to play audio from buffer start
|
| 72215491c60fbcdb9a2f0be782e24e39cca249c5 |
|
26-Nov-2014 |
Eric Laurent <elaurent@google.com> |
am 1a475921: am 223fd5c9: audio: new routing strategies and stream types
* commit '1a475921c0577a4650d1bbe40a85b732d1766939':
audio: new routing strategies and stream types
|
| caf7f48a0ef558689d39aafd187c1571ff4128b4 |
|
26-Nov-2014 |
Eric Laurent <elaurent@google.com> |
AudioRecord: use audio attributes instead of audio source.
Added AudioRecord constructor with audio attributes.
Replaced AudioPolicymanager::getInput() by getInputForAttr().
No new functionality for now.
Also:
- Fixed warnings in AudioPolicyManager
- Allocate audio session ID before calling getOutputForAttr() in
AudioTrack.
Bug: 16006090.
Change-Id: I15df21e4411db688e3096dd801cf579d76d81711
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 72bf901ca905ce154fe7866f06d25a01aadcf974 |
|
26-Nov-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audio policy: new getOutputForAttr() prototype." into lmp-mr1-dev
|
| d325005e4a3be45c57e950f8c18b329c968602ce |
|
26-Nov-2014 |
Andy Hung <hunga@google.com> |
Merge "Fix AudioTrack loop mode to play audio from buffer start" into lmp-mr1-dev
|
| e83b55dc29ca16092ba02f36f55fa6e0e37fd78c |
|
14-Nov-2014 |
Eric Laurent <elaurent@google.com> |
audio policy: new getOutputForAttr() prototype.
Update getOutputForAttr() prototype and group all logic
dealing with audio attributes to stream type conversion in
audio policy manager.
getOutputForAttr():
- specifies the audio session (for future use)
- returns a status code
- receives either stream type (for legacy) or audio attributes
- returns an updated streamtype
Remove logic dealing with legacy stream types to attributes conversion
from AudioTrack.
Use correct type for audio sessions in other APIs (startOutput() ...).
releaseOutput() specifies the audio session (for future use).
Bug: 18067208.
Change-Id: I1bfbe9626c04c7955d77f8a70aecfad2cb204817
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 680b795435a5a501a56610258dce5eda40a1725d |
|
12-Nov-2014 |
Andy Hung <hunga@google.com> |
Fix AudioTrack loop mode to play audio from buffer start
Bug: 18217633
Change-Id: Ica77acf0a32832d9b04eb657ef2f4f5329f8fbda
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 223fd5c9738e9665e495904d37d4632414b68c1e |
|
11-Nov-2014 |
Eric Laurent <elaurent@google.com> |
audio: new routing strategies and stream types
Added new routing strategies and stream type for internal use
by audio policy manager and audio flinger:
- One for accessibility to allow different routing than media
- One for re-routing (remote submix) in preparation of dynamic
policies
- Added stream type for "internal" audio flinger tracks used
for audio patches and duplication.
Bug: 18067208.
Change-Id: I88f884b552e51e4a49c29125e5a1204cf58ff434
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 9544a8e904955f4b96ec9f2d6f4edc2a72afd270 |
|
18-Nov-2014 |
Eric Laurent <elaurent@google.com> |
am bfa77b5c: am 8cd37d67: Merge "audio policy: do not prevent routing to A2DP when suspended." into lmp-mr1-dev
* commit 'bfa77b5c687f217e26267693306aaa6cc8a63672':
audio policy: do not prevent routing to A2DP when suspended.
|
| 29e6cecdd8562700ab625c5645ddd6a7753789f1 |
|
14-Nov-2014 |
Eric Laurent <elaurent@google.com> |
audio policy: do not prevent routing to A2DP when suspended.
Allow A2DP sink selection even if A2DP is suspended.
It is not because use of A2DP is temporarily disabled by SCO
selection that the policy rule routing a given use case
to A2DP should be altered. The source will be routed to A2DP
but it will remain silent until suspend condition is cleared
which is the purpose of suspend mode.
Also make sure that accessibility prompts are audible while in call
by forcing stream type to STREAM_VOICE_CALL.
Fix inconsistent logic between AudioTrack::setStreamTypeFromAttributes()
and AudioPolicyManager::getStrategyForAttr()
Bug: 18095151.
Change-Id: I1c75dd8288ed3dd494eb942060eac9c9660dca50
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 77e79a509d19ab50b1867fe3903b5a8fae450591 |
|
17-Nov-2014 |
Marco Nelissen <marcone@google.com> |
resolved conflicts for merge of e78a27ca to lmp-mr1-dev-plus-aosp
Change-Id: If10a9cc17245f95d5e10b1507445abbb4020670e
|
| f888020c6e2735624f2b2a30e72aca24e17b8b4d |
|
14-Nov-2014 |
Marco Nelissen <marcone@google.com> |
Update calls to IInterface::asBinder()
to use the new static version.
Change-Id: Ia7b10eb38ca55b72278bfd33d3bf647f338b4e6a
Conflicts:
media/libmedia/IAudioFlinger.cpp
media/libmedia/IMediaPlayer.cpp
media/libstagefright/CameraSource.cpp
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 0d6db582f2ccc9f8943c5f3965e2994b7d137158 |
|
13-Nov-2014 |
Eric Laurent <elaurent@google.com> |
remove AudioSystem::getOutputSamplingRateForAttr()
It is safer to query the output sampling rate once the output
have been acquired by calling getOutputForAttr().
Bug: 16009464.
Change-Id: Ib561facd19fba5359a6b837d75f1a1cd2dc51b29
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| d2d089fc86c62843992e7d5b371ee9227189a1e6 |
|
05-Nov-2014 |
Glenn Kasten <gkasten@google.com> |
Improve AudioTrack recovery from mediaserver death
1. Fix race condition in handling of binder death notifications.
AudioSystem has a mixture of APIs for both ordinary app clients,
and the AudioFlinger and AudioPolicy services within mediaserver.
Due to this mix of uses, it is possible for there to be "surprising"
sequences of calls on the call stack.
Previously, we used a single mutex for all global variables, but
this caused a deadlock. To avoid the deadlock, we unlocked the mutex
during the critical sequence of calls. But this was a a
crucial place where it should have stayed locked;
see Change-Id I315c1c5066f62b05e1c13b04fae1272b5fbce977
Now we use separate mutexes for the AudioFlinger, AudioPolicy, and audio port
related global variables. This allows us to correctly hold each mutex
throughout the atomic region, even when AudioFlinger calls AudioPolicy
via AudioSystem, or vice-versa.
2. AudioSystem::clearAudioConfigCache now clears the IAudioFlinger reference.
3. Make AudioSystem::get_audio_policy_service more like get_audio_flinger.
Bug: 18242291
Change-Id: I9761443d8337df5bf66d4ca2316a9fd0bd11be94
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 461b02d3eccefea48fa15c67a9cd3a5895bd2765 |
|
04-Nov-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audio: fix talkback why alarm is playing" into lmp-mr1-dev
|
| 03fcdcd0aa9758b6d882b18218fcfb9a1ccffdad |
|
04-Nov-2014 |
Eric Laurent <elaurent@google.com> |
audio: fix talkback why alarm is playing
Convert attributes usage "assistance accessibility" to
a stream type guarantying that the prompt is audible
and not muted by policy manager.
Bug: 17994328.
Change-Id: I89f53f2ce84b499b859336c0a1a3a0332ee670f5
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| d9cfeb447356cb6334379eaf5da1e49424eb5979 |
|
23-Sep-2014 |
Jean-Michel Trivi <jmtrivi@google.com> |
Support speaker routing for FLAG_BEACON
Add support for routing audio stream to the speaker when audio
attributes carry the FLAG_BEACON value:
- associate streams with FLAG_BEACON to the up-to-now unused
STREAM_TTS ("Transmitted Through Speaker").
The remapping happens in AudioTrack.
- AudioPolicyManager: define new strategy for handling the Beacon
behavior, that only goes to speaker. Define new volume curves
for STREAM_TTS that are not silence only for the speaker device
category. Keep ref count of other streams and never play Beacon
streams when anything else is playing: either don't start playback
if something else is playing, or unmute STREAM_TTS when it's
allowed to play and mute when it's done.
- Take into account muting latency when starting an output.
Bug 15415971
Change-Id: I26539c7ec1a486accd85bbeb1623e9c7d3a1192f
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| c2813e568aa476e03d159529c0af28c99536db8d |
|
17-Oct-2014 |
Andy Hung <hunga@google.com> |
Fix AudioTrack retrograde getPosition when restoring tracks
mReleased represents the number of frames written to the
track, but was cleared on start() causing a mismatch if the
client wrote to the track before starting. Moved the clearing
to entering the STATE_STOPPED or STATE_FLUSHED state.
Bug: 18017947
Bug: 18022276
Change-Id: I3788c98c4c3c4d9cc004378432797b3f3138e22e
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 7985dcb06e0c29d5cc12d0c0e17e03d5d863cf53 |
|
08-Oct-2014 |
Eric Laurent <elaurent@google.com> |
AudioTrack: fix AudioTrackThread crash upon exit
In case of error during createTrack_l(), there is a possibility
that a newly created AudioTrackThread is resumed by requestExit()
just after entering the Thread class _threadLoop() but before
entering the class threadLoop(). In this case, processAudioBuffer()
is executed once with mCblk == 0 and we assert.
Bug: 17894033.
Change-Id: I93d0d0d4dbf14bcb88dad23e6ad49d2ecd47badd
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| bb6c9a05840d924b502ce0f1868fca4881ada1ed |
|
25-Sep-2014 |
Eric Laurent <elaurent@google.com> |
audio: fix stream type for accessibility usage
Make sure that accessibility prompts are heard when
a ringtone is active by forcing stream type
to AUDIO_STREAM_RING when phone state is AUDIO_MODE_RINGTONE.
Bug: 17558149.
Change-Id: Ia3bead8052fca5cbf282c267f7b9b06014fef628
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| dfc34daba6d6cb923683d96689a0cb7c7006eee5 |
|
19-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
Reduce log spam from getTimestamp errors
Bug: 17576481
Change-Id: I07d99e7c42b7d4f0994b5a6d9c9fe8eb29fbbe40
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 7f1bc8af1c46695191bf7e2aba6467f3616629c0 |
|
12-Sep-2014 |
Andy Hung <hunga@google.com> |
Fix AudioTrack offloaded timestamp handling.
getTimestamp for offloaded tracks now returns WOULD_BLOCK for
situations where the timestamp is unavailable due to the
previous track still flushing or the timestamp is stale.
It is fixed for normal playback conditions.
Bug: 14659809
Bug: 17428188
Change-Id: Ic9ec07ccabc604236979769db5c4ea2dec252660
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 200092b7f21d2b98f30b800e79d152636f9ba225 |
|
16-Aug-2014 |
Glenn Kasten <gkasten@google.com> |
Clean up AudioTrack position and timestamp handling
Replace epoch concept by observing and accumulating server delta
positions. The advantage of using server deltas instead of absolute
values is that they (1) are not sensitive to 32-bit wraparound,
(2) are not sensitive to server behavior for stop(), and
(3) prepare for future 64-bit client positions without requiring 64-bit
positions on server.
Add comments to AudioTrack::getTimestamp() and friends
that the timestamp output parameter is undefined on error.
Don't allow getTimestamp to return a negative frame position after stop().
Accumulate the client released frames, which may be useful for a future API.
Bug: 11815245
Change-Id: I652940fa2db2f34a78c012a3ead0d9204fa29c6e
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| cd04484f4837b8ca0041d118286ab6a98e84fc75 |
|
07-Aug-2014 |
Andy Hung <hunga@google.com> |
Extend downsampling ratios greater than 2:1
Also improve robustness to choice of sampling rate or buffer size
such that increasing either by 10x does not cause overflow.
Bug: 12979141
Bug: 15933066
Change-Id: If7989bd745d1bee3bdf811b8b7c978543ccafb65
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| d9d7fa0873796ac661c44a7fcd6ad5ff697ff01f |
|
24-Jun-2014 |
Jean-Michel Trivi <jmtrivi@google.com> |
Support for audio attributes on audio output of media player
Change-Id: Iae4995c98e64add1ab9e6c8ae6501515032755f5
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 34fb29696b0f3abf61b10f8d053b1f33d501de0a |
|
19-Jun-2014 |
Mark Salyzyn <salyzyn@google.com> |
libmedia: 64-bit compile warnings
Change-Id: I600f062fa7148c01851023c1240c39939e648002
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| ab5cdbaf65ca509681d2726aacdf3ac8bfb6b3fa |
|
10-Jun-2014 |
Eric Laurent <elaurent@google.com> |
AudioTrack: add support for compressed audio
Add support for compressed audio playback by use
of an AudioTrack attached to a direct output thread.
Bug: 9428304.
Change-Id: I4a61be9cf0e31003ca85935d6e0ee38ca8192e03
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 5bd3f38638acab633d181359cc9ec27b80f84d43 |
|
14-Jun-2014 |
Jean-Michel Trivi <jmtrivi@google.com> |
AudioPolicyManager: return output for audio attributes
In AudioPolicyManager, support querying an output
or playback strategy for audio attributes,
instead of a stream type,
In AudioTrack creation, use the output returned
for the track's attributes.
Change-Id: I0fef05845ba676404775e2e338c10e6a96237268
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| faabb51ceef13bf1e3f692219ac410c1cd75d0de |
|
12-Jun-2014 |
Jean-Michel Trivi <jmtrivi@google.com> |
AudioTrack construction with audio attributes
Store audio attributes in AudioTrack class.
When an AudioTrack is "set" with non null audio
attributes, derive a stream type that reflects
the attributes.
When an AudioTrack is "set" without attributes,
and only has a stream type, derive default
audio attributes.
Change-Id: I322b91fa9a7e193118960c5e78cdddd85d66f9ad
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 9c68ac03f16cc32625f6965e869432eae4fded2d |
|
13-Jun-2014 |
Jean-Michel Trivi <jmtrivi@google.com> |
Merge "AudioSystem: remove unused stream type"
|
| 3bcffa136909c1fb6e88ee4efd12ccac18360a85 |
|
13-Jun-2014 |
Eric Laurent <elaurent@google.com> |
force command flush after releasing a reference to shared memory
After releasing a reference to a shared memory interface,
IPCThreadState::self()->flushCommands() must be called to ensure
that the corresponding reference is also cleared immediately in the
remote process. Otherwise the binder implementaiton will not
push the clear systematically causing the memory region not to be freed.
See issues 10711502 and 2801375.
Bug: 14057166.
Change-Id: If55f36f00d452d6cf00cf83bd2fba1c8f3abcb57
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| b7f24b101d43139b4c747129bfbc4ecf5c468b86 |
|
11-Jun-2014 |
Jean-Michel Trivi <jmtrivi@google.com> |
AudioSystem: remove unused stream type
The audio stream type is not used in the getSampleRate() and
getFrameCount() methods.
Change-Id: I3d065ae272bd039204cd323cdab9b60460034f2d
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| b4db73d022f3de3530bc2b3c9c831ccfdd1a2ead |
|
20-May-2014 |
Andy Hung <hunga@google.com> |
Merge "Use new channel count functions for audio masks"
|
| c56f3426099a3cf2d07ccff8886050c7fbce140f |
|
22-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Pass stereo gains as packed minifloat
This will allow (eventually) a greater dynamic range for gains.
However there are still a few remaining places in effects and mixer
that will also need to be changed in order to get the full benefit.
Also fixes a minor bug: was not checking for NaN in AudioTrack C++.
Change-Id: I63bce9e82e0a61546d8ff475fb94bcb700d99c96
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| e541269be94f3a1072932d51537905b120ef4733 |
|
16-May-2014 |
Andy Hung <hunga@google.com> |
Use new channel count functions for audio masks
Change-Id: Ia658ab4b6320d19fdb50f123c930918724ff0ef3
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| b773038277ba328696832b690b91ce6e95bd1a6c |
|
01-May-2014 |
Glenn Kasten <gkasten@google.com> |
Add client side support for more AUDIO_FORMAT_PCM_*
As much as possible, the checks for specific formats are done on server
side. The only exception is 8-bit, which needs a client side workaround.
Change-Id: I52fe086c039460c8dac009de03b67eb54c56a836
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| c4b88a8d0f524666bf0f390075c334d047a104f2 |
|
01-May-2014 |
Glenn Kasten <gkasten@google.com> |
Fix bug for direct track with PCM != 16-bit
The AUDIO_FORMAT_PCM_8_BIT format was being converted to
AUDIO_FORMAT_PCM_16_BIT on client side even for direct tracks.
That conversion was incorrect; it should only be done for mixed tracks.
Also remove checks for specific PCM formats in the generic part of
server side of createTrack. Those format checks should only be done by
the thread. This will allow direct tracks for PCM 8-bit, PCM 24-bit, etc.
Change-Id: If5b9fd79f8642ed93e2aeabcaf4809b2ed798978
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 877a0ac7e8677fa1cbd0e8565ad9d38ba0db5fc0 |
|
01-May-2014 |
Glenn Kasten <gkasten@google.com> |
Use AUDIO_INTERLEAVE_* constants
Change-Id: I6609fa75c9a57fc4ca3887d626a2f4fb8b6593a1
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 142f519aa1acd5804d111e60d100f170fed28405 |
|
26-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Use symbol AUDIO_IO_HANDLE_NONE from <system/audio.h>
Change-Id: Id6b1aa17558eb73e17f22b8eab6cd02e00a96dff
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| ecdca392ed40e89a6ef14b86446e952c35497ac4 |
|
26-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use symbolic constants from <system/audio.h>"
|
| 8c7b9fb41585fccb06573252823b929c82a40999 |
|
26-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove streamType parameter from AudioSystem::getLatency()"
|
| ebb80e4f9873cc1a5ee3f766323f622bb0c07ae5 |
|
26-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Remove dead code
Change-Id: I0878d11451c7bbbf96b59f5fe0cd97ba1f033aa9
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 241618f1b286f9e7e02fe61e96d9194e2e1e8a7a |
|
26-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Remove streamType parameter from AudioSystem::getLatency()
Change-Id: Ie7346e93436ddc215cad7d16be555dcb6c277d54
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 66e4635cb09fadcaccf912f37c387396c428378a |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Use symbolic constants from <system/audio.h>
AUDIO_INTERLEAVE_*
AUDIO_STREAM_MIN
AUDIO_SESSION_ALLOCATE
Change-Id: I31dd6f327204685e50716079ce21c4ba206dff11
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| c5a17425986b4ce3384e6956762c86018b49c4a0 |
|
13-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Remove name output parameter from createTrack
It was only used for one log.
A better solution will be a per-track unique ID.
Change-Id: Ia440e02ae4a5a4019a9a2d08970e1ee93ac4c3a3
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| c644345b0b8f5292a049c4cee2bb7a2a6159b884 |
|
11-Mar-2014 |
Jean-Michel Trivi <jmtrivi@google.com> |
Merge "AudioTrack non-blocking write"
|
| 3a90f2849d49bf65f2d6257fd6da30ae46d745fa |
|
10-Mar-2014 |
Marco Nelissen <marcone@google.com> |
Fix freeze on pause
isOffloaded() tries to lock mLock again. We should be calling isOffloaded_l()
b/13394633
Change-Id: I155be6fee937f894d8e6c974e593223ab6014ade
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 1c32a00b8750998562a8d76351ba5c6d2ed788c8 |
|
07-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify AudioTrack stream end and fix race"
|
| 7064fd2dcdfeafea53cd5a992bb78c413542f29f |
|
08-Jan-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
AudioTrack: When paused, return cached playback position
An offload output can be re-used between two audio tracks having
the same configuration. A timestamp query for a paused track
while the other is running would return an incorrect time.
To fix this, cache the playback position on a pause() and return
this time when requested until the track is resumed.
Bug: 12826612.
Change-Id: I324112ea9827e52fff53ef44cd8513c8d85a0bc4
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 720ad9ddb2ac6b55b0dfbfcd2d8360151d8ac427 |
|
04-Feb-2014 |
Jean-Michel Trivi <jmtrivi@google.com> |
AudioTrack non-blocking write
Bug 7531968
Change-Id: I6d0e79fa8cab5b6eb36bcc34977f4cf0d7eec8ea
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 33ca5cf8826f5fcd9c011e8d336138b62d982f2a |
|
28-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Make openRecord_l more like createTrack_l for fast tracks: part 1"
|
| bce50bfc3846ab008bafa75c5d3f29fd7b5395f7 |
|
28-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Fix type of AudioTrack/AudioRecord parameter frameCount
It's size_t consistently
Change-Id: I29638ef59ac773218025f2403a3508a307b487e0
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 838b3d8bafa4a781e277870dee4e0390165cff52 |
|
28-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Fix type of AudioTrack/AudioRecord parameter notificationFrames
It's uint32_t consistently
Change-Id: If8298c7e9aeea2b951fe47b675adbdf48d104846
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| c6ba823a397abf865b02f4f48fe18231f94d8e87 |
|
27-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Make openRecord_l more like createTrack_l for fast tracks: part 1
Use the transfer mode to distinguish use cases, as the presence of
a callback handler is not sufficient. For example, the track could be
configured for synchronous transfer with write() or read(), and also
have a callback handler for position updates. But that does not mean
the track can operate in fast track mode.
Change-Id: I2a7f1f0ca98e68efe180b524496985109d8ce291
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| c08d20b6a37122ebf116262c9372509ed060d4c1 |
|
25-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Simplify error handling after track creation
IAudioFlinger::createTrack and IAudioFlinger::openRecord
both guarantee that (status == OK) == (sp<> != 0).
Change-Id: I91cb4f7e843019efb65cace7ba146f7da7aa5b59
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 5f631515d098c29603cda88f7a7e7580a2d55b57 |
|
25-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Unify comments and whitespace between AudioTrack and AudioRecord
Change-Id: I5320a6b2d7f7077cb12d7da4f2ca30a940100bf2
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| e3247bf8dd4f8fa8dfa3a108260241ae4a967569 |
|
25-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Update channel fields at same place in AudioTrack and AudioRecord
Change-Id: I4b649feeef47e7968a9fa3a460217017ca9b05fe
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 86f04663032ddaa25110149d709bbf896ad83b02 |
|
25-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Add log at entry to set() in AudioTrack and AudioRecord
Change-Id: Ife23b88474c1d62c0cf682c1a310d951f2c0f54a
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| d457c970c8d08519cd77280a90b61ae1e342cfe3 |
|
11-Feb-2014 |
Marco Nelissen <marcone@google.com> |
Track pid for each session
so they can be properly freed.
Change-Id: I6f389035bc29e74e7c367c1c6d0252b180f666b3
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 1d6fa7af1288b550faabe4ec2cf98684236723db |
|
11-Feb-2014 |
Narayan Kamath <narayan@google.com> |
resolved conflicts for merge of 566be7c3 to master
Change-Id: I7b1cc71057b2bd4f771e7bcf508a8c3abd6017ce
|
| 377b2ec9a2885f9b6405b07ba900a9e3f4349c38 |
|
03-Feb-2014 |
Kévin PETIT <kevin.petit@arm.com> |
Make frameworks/av 64-bit compatible
Contains the necessary changes to make frameworks/av build and work
on a 64-bit machine.
Signed-off-by: Craig Barber <craig.barber@arm.com>
Signed-off-by: Kévin PETIT <kevin.petit@arm.com>
Signed-off-by: Ashok Bhat <ashok.bhat@arm.com>
Signed-off-by: Marcus Oakland <marcus.oakland@arm.com>
Change-Id: I725feaae50ed8eee25ca2c947cf15aee1f395c43
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 43bdc1de363a3c72c7dcf9c9a898bac109dc7cb5 |
|
10-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Fix build after merge from AOSP
Change-Id: I1cd3432c02db1600e49ade67e6a89e39e7a8d3ed
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 3523e8c40bc60af0c95d1aa71a51a13d69ec80a2 |
|
10-Feb-2014 |
Glenn Kasten <gkasten@android.com> |
am 4e12dade: am 4361749d: Merge "AudioTrack: Never try to use the fast path if resampling is required"
* commit '4e12dadeec10993d2f77cce96600e73ca0dd86e2':
AudioTrack: Never try to use the fast path if resampling is required
|
| cac3daa6332bf6d1f7d26adc4a9915f3d7992dd9 |
|
07-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Use printf format %#x for audio_format_t in logs
Change-Id: I1c611d1037685d52ccc84efe0fccd6413ec938e9
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 398f21348e5100289f6e5be30c8b5257fa04aaf9 |
|
31-Jan-2014 |
Martin Storsjo <martin@martin.st> |
AudioTrack: Never try to use the fast path if resampling is required
Unless AudioFlinger was built with FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
enabled, AudioFlinger would deny using the fast path (and internally
fall back to the normal codepath) when it realized that resampling
was required. Since the buffer size calculations within AudioFlinger
don't take resampling into account properly (see the calculation
below "AUDIO_OUTPUT_FLAG_FAST denied" in audioflinger/Threads.cpp,
just below the hunk that this patch changes), make sure AudioTrack
doesn't try to use the fast path if resampling is required.
This removes the possibility to enable
FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE in AudioFlinger since it
AudioTrack now won't even try to use the fast path for content
that requires resampling, regardless of the AudioFlinger configuration.
Change-Id: Icf0f8ad50bf0fdb84657f518c0120aa0535f23f9
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| df576995a5aad3428aeeef765387d650945c161d |
|
28-Jan-2014 |
Eric Laurent <elaurent@google.com> |
AudioTrack: fix obtainBuffer timeout calculation
AudioTrack::obtainBuffer() passes a pointer to a timeout
variable that has gone out of scope when calling
ClientProxy::obtainBuffer().
Same fix for AudioRecord.
Bug: 11968591.
Change-Id: I22af8f94fa2f8cc54ab5c25c89167c805e754317
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 363fb75db26698cbb50065506e0c80b61d1fbf92 |
|
15-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Remove the redundant parameters from createTrack_l()
AudioRecord::openRecord_l() code was refactored earlier to
remove the redundant parameters:
> Change-Id: I124dce344b1d11c2dd66ca5e2c9aec0c52c230e2
This changelist refactors AudioTrack similarly.
Change-Id: Iefd2bd662870ea81d04eff7b7c26f9c8b0dadd26
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 38e905b3cbba4da443d799b16999989781afc6d8 |
|
13-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Refactor code related to I/O handles to reduce chance for leaks
The AudioRecord input handle code was refactored earlier
to fix a potential handle leak, and to simplify the code:
> Change-Id: I124dce344b1d11c2dd66ca5e2c9aec0c52c230e2
This changelist refactors AudioTrack similarly,
and adds further cleanup of both AudioTrack and AudioRecord.
We attempt to implement the rules for referencing counting I/O handles,
but there is still the possibility of a handle leak if the client process
dies after allocating the handle reference but before releasing it.
That issue is being tracked separately.
Details:
- AudioSystem::getOutput() is now called within createTrack_l
- restoreTrack_l was missing offload info
now it has the info available,
but is not yet being called for offloaded tracks
- AudioTrack::getOutput() is now const
- Remove getOutput_l()
Change-Id: I44a0a623d24fc5847bcac0939c276400568adbca
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| f0002d142e6d24c5438600b2c259679de710f8ac |
|
24-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Replace control block frameCount_ by explicit in/out parameter"
|
| 1071915ad4ffb47825a946706f8fa73c0d22d7f8 |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Make copy of audio_offload_info_t for future use"
|
| 543140eb709a07bc386eb5f667a4a72e31dba56c |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Move up initialization of mFormat, mStreamType, and mSharedBuffer earlier"
|
| 0ce5ca5e113d58cbcb7cb48615d6dbbf3fbe7c66 |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Cleanup createTrack error handling"
|
| d0b33f63a9364364a827e77da26b112edf3c4d6e |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove dead code in AudioTrack::processAudioBuffer"
|
| a1ebc3b03d4dca534374c19e3c4f32ee687942e3 |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Improve logs in AudioTrack::getMinFrameCount"
|
| 32a7e0a5f7b000951b9adb80b17796f35b024816 |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Unify comments between AudioTrack and AudioRecord"
|
| b5ccb2d99b2af400c70777a6452f090ac14f5a96 |
|
13-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Make copy of audio_offload_info_t for future use
Change-Id: I515970aa0660418d5d1640fb1bf477e112c89bdd
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| dd5f4c8c4059f890e81b28b026a688febb4e1dd9 |
|
13-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Move up initialization of mFormat, mStreamType, and mSharedBuffer earlier
This will allow removal of most of the parameter list to createTrack_l.
Also check for valid stream type at client so we can log a better error
message.
Change-Id: Ia7176896d47fbb49106119fca26d9de8e7efe859
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 0cde076ddb283c84c3801a2df4cc3df99bd1577f |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Cleanup createTrack error handling
Similar to cleanup done earlier for openRecord in this CL:
> Change-Id: I84897dd7d30b370640b54e928f230604b873cb68
Bug: 10888816
Change-Id: I804a47f898e0319a60a9bd58906bbb037e45cc25
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 70c0bfbe5ec88dcc3efa2bd8df26f36cff1cf03a |
|
15-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Improve logs in AudioTrack::getMinFrameCount
Change-Id: I8b478e6abdded26a43f32c131931939e9ae36fd7
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 7103d9671ded507fd3f6938297019a32f71ea5a0 |
|
15-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Remove dead code in AudioTrack::processAudioBuffer
Change-Id: I0baefe9dc08bb5ec45d34698fc764caa160789d2
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| ac1954cb9a480a409e8cee4193fc534a6a29fbb6 |
|
15-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Document AudioTrack mFrameCount and mReqFrameCount better"
|
| 2b2165c75790050810460c8de3f414876bce4c0e |
|
13-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Unify comments between AudioTrack and AudioRecord
Change-Id: I00a1025e2891a1c96218b3c2187eaddda6614ebc
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 23a7545c4de71e989c2d8ebf1d5b9dcf463c36a9 |
|
13-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Document locking rules for mFlags, and fix discrepancies
Change-Id: Id45ba544cc84133ed5e578fb4fd8a11b62211dc1
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 84fba38b60c2ddb02b9a787f56be88fc084b4a16 |
|
14-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix race in AudioTrack::getParameters()"
|
| b1bef51fec06e8bceec914d1b48b411d3ff2adff |
|
13-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Improve error logging for getOutputSamplingRate
Change-Id: I3b52402a663b27efe1d7c6a4f684521f33f3ff8f
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 2c6c5294388e251ebc1e00d6c8785190561c27fc |
|
13-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Fix race in AudioTrack::getParameters()
mOutput is protected by mLock.
Change-Id: Id02e627062855ca60f28bd8961b1d5f44939c727
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 396fabdb6efcdac5aea3d9f559d1beedf6a4cedc |
|
08-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Document AudioTrack mFrameCount and mReqFrameCount better
and remove unnecessary initialization of mFrameCount in set().
Change-Id: I9effeb0a6dd035ca02fe77f6992c55d9515b4df6
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 66a0467fdddada4caabd0f0a999fbb367fea7bee |
|
08-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Cleanup AudioTrack::getMinFrameCount error handling
Guarantee to return a non-zero frameCount for return status NO_ERROR;
Return the correct specific status_t if any of the AudioSystem APIs fail,
instead of the generic NO_INIT.
API change: getMinFramCount no longer defaults to zero on error, so
callers _must_ check the return status. This change makes
getMinFrameCount more like other APIs. All known callers were reviewed,
and they do check the return status.
Change-Id: I4a8342a75ee89a068c23c84b8380ed9d1b968507
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 74935e44734c1ec235c2b6677db3e0dbefa5ddb8 |
|
19-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Replace control block frameCount_ by explicit in/out parameter
in IAudioFlinger::createTrack and IAudioFlinger::openRecord
Change-Id: I09c644c80e92c8e744b1b99055988a2588b2a83d
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 7c7be1e05634d96d08210efb4bdeb012ffba440d |
|
20-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Fix some (but not all) unused parameter warnings
Change-Id: Ia99e23a0b46db3f3e6aa46f9018e63c14f4af369
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| dfeed91152cb54772b1411d7e82833ca4a48efcc |
|
27-Nov-2013 |
Eric Laurent <elaurent@google.com> |
AudioTrack: fix position callback after restore
When restoring an AudioTrack, the next position callback point
should not be modified and set ahead of current buffer head.
Otherwise, as frames are dropped, the new position is never reached
and an application relying on position callbacks to reload the buffer
would be stalled.
Bug: 11868603.
Change-Id: I93b2a311642a0c89944b78bcc0482d4ceed98ae4
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 5e49afd05566820517747b9a8071c99ec0918328 |
|
27-Jul-2013 |
Eric Laurent <elaurent@google.com> |
update offloaded audio track sampling rate
AudioPlayer must read the sampling rate from offloaded audio sinks
whenever a new time position is computed as the decoder can update
the sampling rate on the fly.
Change-Id: I997e5248cfd4017aeceb4e11689324ded2a5bc88
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 688ca6330a5e7d072f67a1791e464fa7dec08b72 |
|
17-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
am 5c474238: am 5893736d: am a67c7653: Merge "Increase kFastTrackMultiplier from 1 to 2"
* commit '5c474238bc579224ec8f7267d05651b0aa934100':
Increase kFastTrackMultiplier from 1 to 2
|
| a67c76538fff102cd44b2f38dd073df5f2a33605 |
|
17-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Increase kFastTrackMultiplier from 1 to 2"
|
| b5fed68bcdd6f44424c9e4d12bfe9a3ff51bd62e |
|
03-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Increase kFastTrackMultiplier from 1 to 2
Change-Id: I158f147295eebcea96e4047d7618069bc48bdd7d
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| b31788687736666e58b852e744a93f772d3f7148 |
|
27-Nov-2013 |
Eric Laurent <elaurent@google.com> |
AudioTrack: fix position callback after restore
When restoring an AudioTrack, the next position callback point
should not be modified and set ahead of current buffer head.
Otherwise, as frames are dropped, the new position is never reached
and an application relying on position callbacks to reload the buffer
would be stalled.
Bug: 11868603.
Change-Id: I93b2a311642a0c89944b78bcc0482d4ceed98ae4
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 6f59db12a64f4496866952a251122ccb77a36c6b |
|
27-Jul-2013 |
Eric Laurent <elaurent@google.com> |
update offloaded audio track sampling rate
AudioPlayer must read the sampling rate from offloaded audio sinks
whenever a new time position is computed as the decoder can update
the sampling rate on the fly.
Change-Id: I997e5248cfd4017aeceb4e11689324ded2a5bc88
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 8ba90326d683b035d99e24db669093e4602a7149 |
|
30-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
Remove default channel mask for AudioTrack
It doesn't make sense to have a default channel mask, since the caller
needs to know what format it will use when supplying data, and there
is currently no API to return the channel mask of an AudioTrack.
audio_is_output_channel() does not allow 0, so it will catch any
stragglers (I'm not aware of any).
Also move channel mask validation earlier.
Change-Id: Ia018ded8711455581a2a935f37432b049422d492
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 9cae217050aa1347d4ac5053c305754879e3f97f |
|
14-Jan-2013 |
Marco Nelissen <marcone@google.com> |
Assign blame for playback wakelocks.
Set a work source for the playback wakelock, so that playback is
counted against the requesting app instead of the media server.
Cherrypicked from master.
b/9464621
Change-Id: I7329f88a288a95a582a78005a1c3d16a5a611e31
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 462fd2fa9eef642b0574aa7409de0bde3fec8d43 |
|
14-Jan-2013 |
Marco Nelissen <marcone@google.com> |
Assign blame for playback wakelocks.
Set a work source for the playback wakelock, so that playback is
counted against the requesting app instead of the media server.
Change-Id: I7329f88a288a95a582a78005a1c3d16a5a611e31
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| abe0948c7b9aa32dceb82cebbfca52beb782754b |
|
17-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
am b2059ff3: am f8f15b05: Merge "Fix race condition in AudioTrack::pause followed by start" into klp-dev
* commit 'b2059ff384eee8ffb70a7ec8fc5570405201c734':
Fix race condition in AudioTrack::pause followed by start
|
| f8f15b05fe051009945c9042a1a9260280e0feb2 |
|
17-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix race condition in AudioTrack::pause followed by start" into klp-dev
|
| 598de6c701e989385eeffa7c5dfd61f0459a2631 |
|
17-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
Fix race condition in AudioTrack::pause followed by start
Bug: 11148722
Change-Id: Iec88f00c8510363d4418e4b8d5b34feb06ecf04d
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 771f1b6d196b7924707fc44177ac3a88fc31fa9e |
|
17-Oct-2013 |
Eric Laurent <elaurent@google.com> |
am 7f8c3973: am 120a8847: Merge "AudioTrack: fix head position after restore" into klp-dev
* commit '7f8c397378a7ee5abd395413be71388ad36d3ed2':
AudioTrack: fix head position after restore
|
| cc21e4f1e41dfa17e7e2bef995fcd22c45f6bcd0 |
|
17-Oct-2013 |
Eric Laurent <elaurent@google.com> |
AudioTrack: fix head position after restore
The head position transfered to the new track
by restoreTrack_l() must take into account the frames that
are dropped from the old track to avoid a non recoverable
offset in the playback head position returned to applications.
Bug: 11230062.
Change-Id: I51143a08b95e8f264ed709ae2054360315f2b8b1
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 7a72e2aca9a6c46a08bc605166f90915aaeb81cb |
|
07-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix AudioTrack pause followed by stop"
|
| 59fe010bcc072597852454a2ec53d7b0a2002a3b |
|
28-Sep-2013 |
Eric Laurent <elaurent@google.com> |
fix volume and effect enable delay on offloaded tracks
Volume: add a method to wake up the mediaserver playback
thread when a volume command is received on an offloaded track.
Effects: call effect chain process on offloaded playback threads
asynchronously from writes to allow effect state updates while
waiting for async write callback.
Bug: 10796540.
Change-Id: Id2747ae88783575d1d7ffd6fc86fbd054ab2c739
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 96f04886d1c1bfbc422e2be033ea66be83e42441 |
|
20-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Simplify AudioTrack stream end and fix race
Bug: 10994052
Change-Id: Ib2e38e7a600bcffef8cbc68c1722e40fbbc7ea67
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 397edb3377e5775f4df60afb8bf6d4711e5adc0e |
|
31-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Fix AudioTrack pause followed by stop
Now the stop is not a nop.
Bug: 10993355
Change-Id: Idfbfd6d14897574578b80648a16e0fc73765cb6c
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 9d2c78c4798ffd8c276c1bf0eaa0b34bc255a2da |
|
23-Sep-2013 |
Eric Laurent <elaurent@google.com> |
AudioTrack: fix music resume
Fix regression introduced by commit 5a6cd22 in AudioTrack resume:
the callback thread was not signaled if paused internaly.
Bug: 10895013.
Change-Id: Ic356b115132d6fccbcee2d9bb855e92671dc20c5
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 5a6cd224d07c05b496b6aca050ce5ecf96f125af |
|
20-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Fix slow AudioTrack and AudioRecord destruction
There were two causes for the slowness:
When thread was paused, it used nanosleep and sleep. These usually
run to completion (except for POSIX signal, which we avoid because it
is low-level). Instead, replace the nanosleep and sleep by condition
timed wait, as that can be made to return early by a condition signal.
Another advantage of condition timed wait is that a condition wait was
already being used at top of thread loop, so it is a simpler change.
The AudioRecord destructor was missing a proxy interrupt that was correct
in AudioTrack. This proxy interrupt is needed in case another thread
is blocked in proxy obtainBuffer.
Does not address the 1 second polling for NS_WHENEVER.
Bug: 10822765
Change-Id: Id665994551e87e4d7da9c7b015f424fd7a0b5560
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| ce8828a016b082f730152af2204b8ea3610dc1ec |
|
17-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Fix underruns when fast track denied due to SRC
OpenSL ES requests a fast track. If sample rate conversion is needed,
the request is denied by server, and a larger client buffer is used
to handle the higher latency of a normal track. However the client
notification period was calculated based on buffer being divided into
2 sub-buffers. That resulted in the notification period being too long.
The server pulls chunks that are smaller than half the total buffer.
So now the client uses 3 sub-buffers when there is SRC.
Also removed the 'defer wake' optimization because it was incorrect.
This optimization attempted to reduce the number of wakeups of client,
when server releaseBuffer knows that another releaseBuffer will be
following. But there is no way for the first releaseBuffer to predict
how soon the second releaseBuffer will occur. In some cases it was
a long time, and the client underran. So now the client is woken up
immediately if the total number of available frames to client is >=
the minimum number the client wants to see (the notification period).
Also fix bug where minimum frame count was not being used in the
calculation of notification period.
Bug: 10342804
Change-Id: I3c246f4e7bc3684a344f2cf08268dc082e338e2a
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| fe346c707f59d763ded93bc3d27b51f0c0408258 |
|
30-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Fix miscellanous AudioTrack::getTimestamp() bugs
Check that get_presentation_position is non-NULL before calling.
AudioTrack::getTimestamp not implemented for fast tracks.
Fix typo in Track::getTimestamp().
Fix bugs in AudioTrack::getTimestamp after stop:
- getTimestamp while stopped is not allowed.
- stop, start, getTimestamp now returns the correct value.
Change-Id: Ie8d9dc1f28d8927634e04175a68b147ffc2ea8eb
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 56b592240073b1ab116c6e23a9c4c34ef5ad80d8 |
|
30-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Add IAudioTrack::getTimestamp()" into klp-dev
|
| 865f6f248bf10189272020ebfdbf4827433caf0f |
|
30-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "New AudioTrack C++ API for audio timestamps" into klp-dev
|
| 42a6f422c09ca6a960673e0e805ddf71a9b51bef |
|
29-Aug-2013 |
Eric Laurent <elaurent@google.com> |
AudioTrack: fix write retries for compressed audio
When the amount of frames that can be written to the buffer
is less than requested, AudioTrack::processAudioBuffer() estimates
the time needed to free the missing amount of frames and asks the
callback thread to sleep.
This behavior is not possible for compressed audio and should not
be enabled for offloaded tracks.
Change-Id: I5b657283cfba06254c9ac0ea9b447467cce7eb61
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 53cec22821072719ee02c856e9ac2dda2496c570 |
|
29-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Add IAudioTrack::getTimestamp()
with dummy implementation in AudioFlinger::TrackHandle, and
implement AudioTrack::getTimestamp() using IAudioTrack.
Also document invariant that mAudioTrack and control block are always
non-0 after successful initialization.
Change-Id: I9861d1454cff7decf795d5d5898ac7999a9f3b7e
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| ce70374bf105f8a6160bf5dd70affea012b2a464 |
|
20-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
New AudioTrack C++ API for audio timestamps
This new API is intended to replace latency(), especially for A/V sync.
The application will receive periodic timestamp notifications. The period
is unspecified, but will likely be more frequent after a pause or stop,
set position, underrun, display on/off change, route change, or when audio
framework notices drift. It will be up to the higher level application
(e.g. Stagefright) to reconstruct a clock that updates more frequently.
The current latency() method doesn't indicate when latency changes
due to screen on/off state, route changes, etc.
Includes squahsed change-Id: I2082f8752040be0c234b1a6f1be2e269abf2ce7c
Dummy implementation of AudioTrack:getTimestamp()
Rename AudioTrack::Timestamp to AudioTimestamp.
Renaming and pulling up to a higher level allows more modules to use it.
Change-Id: Ibf7f6a207c3f8d8697f25ede2cd5200697fadb86
(cherry picked from commit dd69eb893867634fd169c03204a6ad7c74b351e7)
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| ec9a032c2e104ab1e3b5bf73e69dab1408ced0ad |
|
28-Aug-2013 |
Eric Laurent <elaurent@google.com> |
AudioTrack: fix unwanted underrun when restarting
When restarting an AudioTrack from stopped state, it is necessary
to force refresh of mRemainingFrames by processAudioBuffer() as
the last write before stop() could be partial. No doing so will lead
into unnecessary sleep before filling the non contiguous part
of the buffer returned by obtainBuffer() when processAudioBuffer()
is executed for the first time after start().
Change-Id: Id703f8dc092a6f07c905eee194054b4a978f979d
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| d054c32443a493513ab63529b0c8b1aca290278c |
|
12-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Move control block mName to createTrack() output
This is part of a series of CLs to clean up the shared memory
control block, by removing any fields that don't have to be there.
Change-Id: I6e51003a1293b6800258c31b22cff2eba42162e7
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 96f60d8f04432a1ed503b3e24d5736d28c63c9a2 |
|
12-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Rename control block flags to mFlags
Change-Id: I7b6d31e24531954ab1ecdf3ed56c19433700bd89
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| b1a270d1e926fb9a01b4265a7675ed0c2c8f4868 |
|
14-May-2013 |
Richard Fitzgerald <rf@opensource.wolfsonmicro.com> |
libmedia: offloaded playback support
- start() returns a status so that upper layers can
recreate a non offloaded track in case of error.
- Added states to handle offloaded tracks specific:
- waiting for stream end (drain) notification by
audio flinger
- allow pause while waiting for stream end notification
- getPosition() queries the render position directly from
audio HAL.
- disable APIs not applicable to offloaded tracks
- Modified track restoring behavior for invalidated
offloaded tracks: just send the callback and wait for
upper layers to create a new track.
- Added wait for stream end management in audio track client
proxy. Similar to obtainBuffer and should be factored in.
Change-Id: I0fc48117946364cb255afd653195498891f622bd
Signed-off-by: Eric Laurent <elaurent@google.com>
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| cff1b2359b8a302f19539c0bc5d25c98a52733fd |
|
17-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Add comments"
|
| fb1fdc9d6603aa228362e7349451f6455c9849c2 |
|
11-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Add comments
Change-Id: Ifbf3a46a4183c8abc0feee1c588953ab10303cc1
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| b0dfd4613225a3b2a17bdf8d85e89a4b04d65ef3 |
|
11-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Fix type error in AudioTrack::processAudioBuffer
It returned a bool instead of nsecs_t
Change-Id: If0c096dac411afc0a4142ec1e59c1fdd36d4867c
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 1ab85ec401801ef9a9184650d0f5a1639b45eeb9 |
|
31-May-2013 |
Glenn Kasten <gkasten@google.com> |
Include what is needed
Remove old includes.
Header files only include other header files that they directly need themselves.
Change-Id: Ic471386808d9f42ea19ccbd59cb50a5f83a89dd0
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| ad3af3305f024bcbbd55c894a4995e449498e1ba |
|
25-Mar-2013 |
Richard Fitzgerald <rf@opensource.wolfsonmicro.com> |
Public API changes for audio offload support.
NOTE: this does _not_ include all private member variables added
to classes as part of offload support. Only public/protected functions
and stubs functions/variables needed to make the changes buildable.
- isOffloadSupported() added to audio policy service
A stub implementation is required to build, this always returns false
- setParameters() added to IAudioTrack
A stub implementation is required to build, this always returns
INVALID_OPERATION
- CBlk flag for stream end
- Change AudioSystem::getRenderPosition() to take an audio_output_t
so caller can specify which output to query
- Add AudioSystem::isOffloadSupported()
This is fully implemented down to the AudioFlinger function
AudioPolicyServer::isOffloadSupported() which is just a stub
that always returns false.
- Add EVENT_STREAM_END to AudioTrack interface.
STREAM_END is used to signal when the hardware has actually finished
playing all the data it was sent.
- Add event type enumeration to media player interface AudioSink callbacks
so that the same callback can be used to handle multiple types of
event. For offloaded tracks we also have to handle STREAM_END and
TEAR_DOWN events
- Pass audio_offload_info_t to various functions used for opening outputs,
tracks and audio players. This passes additional information about the
compressed stream down to the HAL when using offload.
For publicly-available APIs this is an optional parameter (for some of
the internal and low-level APIs around the HAL interface it is mandatory)
- Add getParameters() and setParameters() API to AudioTrack
Currently dummy implementations.
- Change AudioPlayer contructor so that it takes a set of bitflags defining what
options are required. This replaces the original bool which only specified
whether to use deep buffering.
- Changes to StageFright class definition related to handling tearing-down of
an offloaded track when we need to switch back to software decode
- Define new StageFright utility functions used for offloaded tracks
Currently dummy implementations.
- AudioFlinger changes to use extended audio_config_t.
Fills in audio_offload_info_t member if this info is passed in when
opening an output.
- libvideoeditor changes required to add the new event type parameter
to AudioSink callback functions
- libmediaplayerservice changes required to add the new event type parameter
to AudioSink callback functions
Change-Id: I3ab41138aa1083d81fe83b886a9b1021ec7320f1
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Eric Laurent <elaurent@google.com>
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 9f80dd223d83d9bb9077fb6baee056cee4eaf7e5 |
|
19-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
New control block for AudioTrack and AudioRecord
Main differences between old and new control block:
- removes the mutex, which was a potential source of priority inversion
- circular indices into shared buffer, which is now always a power-of-2 size
Change-Id: I4e9b7fa99858b488ac98a441fa70e31dbba1b865
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| bb6f0a0bb413817d707cfb4c4a30847fda520787 |
|
04-Jun-2013 |
Glenn Kasten <gkasten@google.com> |
Fix underruns when sample rate != native sample rate
This forces a minimum of 3 application buffers when the sample rates
don't match, using the normal mixer and low latency HAL.
There is still an issue that the latency() varies depending on whether
screen was off or on at the time of creating the AudioTrack.
With screen on:
I/AudioTrack( 2028): afFrameCount=960, minBufCount=2, afSampleRate=48000, afLatency=50
I/AudioTrack( 2028): minFrameCount: 2646, afFrameCount=960, minBufCount=3, sampleRate=44100, afSampleRate=48000, afLatency=50
With screen off:
I/AudioTrack( 2817): afFrameCount=960, minBufCount=4, afSampleRate=48000, afLatency=84
I/AudioTrack( 2817): minFrameCount: 3528, afFrameCount=960, minBufCount=4, sampleRate=44100, afSampleRate=48000, afLatency=84
Change-Id: Ib45515edff2afcd672dda34881b658c800ffc25a
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 7c027248e1a4ccd5b22bc4deafb03e2d87ac8f38 |
|
26-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Consistent whitespace
Change-Id: I118cce68d3b777f9ec9b6bfb70367496422a40f2
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 61aa989a5768b9b0571ca14edaf15886509a0ec8 |
|
03-May-2013 |
Eric Laurent <elaurent@google.com> |
am 1641f328: am 4170f515: Merge "Remove timing jitter during startup of audio" into jb-mr2-dev
* commit '1641f328d4de3b2b61481607dc482ccaefe808a9':
Remove timing jitter during startup of audio
|
| a07f17ca46db04c9d5d9e7d6b2878db59ca2b9ea |
|
23-Apr-2013 |
Glenn Kasten <gkasten@google.com> |
Remove timing jitter during startup of audio
This fixes a regression introduced recently, that increased timing jitter
during the startup of the FastMixer and AudioTrack callback threads.
The regression was to make requestPriority() asynchronous as a way to
avoid an apparent priority inversion in system_server. This means that
the target thread could run briefly with the initial priority, before
the new priority takes effect.
This change removes the startup jitter for FastMixer, by making the
requestPriority() synchronous again for that case. It doesn't matter
that this restores the priority inversion involving normal mixer thread,
because it happens during startup of both threads.
The change also removes the startup jitter for the AudioTrack callback
thread, by having the target thread check whether the requestPriority()
has completed yet. If not, the target thread blocks with a timeout
until the priority boost finishes.
Finally, we now log an error message if the expected priority boost
doesn't happen.
Bug: 8698989
Change-Id: Id590e9a274b70ec1ba85b44a585ee37a22e41cbc
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 9c6745f128648f6e0144b74ee593911a9fa10d51 |
|
30-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Propose new interpretation for setPosition and setLoop
Add new API getBufferPosition to return position relative
to start of fixed buffer.
Change-Id: I7aca8e392d45b988545f07b36b5032691057b03e
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| d5681bc9a38fe4cd1d591e6ae62b9c68fb851041 |
|
22-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Start isolating control block accesses in a proxy"
|
| 8d6cc842e8d525405c68e57fdf3bc5da0b4d7e87 |
|
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Remove unnecessary parameter
Just get the parameter on server side
Change-Id: I433a63104dbb257e0d862be2ab61847cb36d1c15
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| e3aa659e9cee7df5c12a80d285cc29ab3b2cbb39 |
|
04-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Start isolating control block accesses in a proxy
The proxy object will eventually be the only code that understands the
details of the control block. This should make it easier to change the
control block in the future.
Initial set of control block fields that are isolated:
- sample rate
- send level
- volume
Prepare for streaming/static separation by adding a union to the control
block for the new fields.
Fix bug in handling of max sample rate on a track. It was only checking
at re-configuration, not at each mix.
Simplify OutputTrack::obtainBuffer.
Change-Id: I2249f9d04f73a911a922ad1d7f6197292c74cd92
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| a8190fc518b6769257896605f3aee091aeb60b50 |
|
04-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Split off the current control block to separate file
Prepare for a new implementation of step() etc.
Change-Id: I268421976ba577aa1fb5d7015de5441c05861190
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 083d1c1492d496960d5b28f4664ff02101736677 |
|
01-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Emphasize distinction between streaming and static
Update comments and improve error checks to match
Change-Id: I7370d6e59a7ef26dfb284a8b058d5ab2e0a42ccf
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 4bae3649d504d590a546717a8e49f96a30d9a745 |
|
30-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
flush() comments and checks
flush() is only useful for streaming mode.
It is a no-op if track is active or uses a static buffer.
Change-Id: I918ac181ffae3d16a0d67d8a7208f4aec61b5bd6
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| e4756fe3a387615acb63c6a05788c8db9b5786cb |
|
29-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
AudioTrack::mute() is unused so remove it
If ever needed again, it could be implemented on client side by forcing
a track volume of 0.
Change-Id: I88a9b4f675b6dca2948549414f9ec2c192d29269
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 01437b7cdaecf53acb46b50ff8b5d86b9d36eb20 |
|
29-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
AudioTrack inline short const methods
Change-Id: I142917edb454d510bbe545e94e6eaea30b650fae
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| a42ff007a17d63df22c60dd5e5fd811ee45ca1b3 |
|
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Clean up channel count and channel mask
Channel count is uint32_t.
Remove redundant mask parameter to AudioTrack::createTrack_l()
and AudioRecord::openRecord_l().
Change-Id: I5dc2b18eb609b2c0dc3091994cbaa4628062c17f
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| aa25c0321bd4b9eb46a4d1cb4ac5fa47af30eeb4 |
|
17-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge ""if" statements use curly braces per media style"
|
| 67ba2b6b8018ce7556cdd880dbf28d06a92ebb78 |
|
17-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Don't use control block frame count after create"
|
| b603744e96b07b1d5bf745bde593fb2c025cefcf |
|
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Don't use control block frame count after create
This is part of a series to clean up the control block.
Change-Id: I7f4cb05aef63053f8e2ab05b286d302260ef4758
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| d65d73c4ae74d084751b417615a78cbe7a51372a |
|
23-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
"if" statements use curly braces per media style
Change-Id: I130e7849fd1da7a0b7fe56c3c53919d26e3843b8
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 827e5f1237757aee78b677efcf0f7c44fd0dd3d8 |
|
02-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Don't explicitly log tid
If needed, it can be obtained with adb logcat -v threadtime
Change-Id: I91b3911d20f7bcfc3361db4052db21ff9181f1cf
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| e33054eb968cbf8ccaee1b0ff0301403902deed6 |
|
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Use size_t for frame counts
Also fix typo: bufferCount should be frameCount.
Change-Id: Ibed539504db75ef99dc21c8ff1bf2987122063a5
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 60a839204713e0f8258d082af83262b1eb33a6c3 |
|
21-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Clean up frame size in AudioTrack and AudioFlinger
TrackBase::mFrameSize, mChannelMask, and mChannelCount are now const.
Use TrackBase::mFrameSize instead of re-calculating frame size.
AudioFlinger only sees 16-bit PCM format, conversion from 8-bit is
now entirely on the client side. Previously a small part of the
responsibility was on server side also.
size_t is unsigned, so use %u in logs.
Fix theoretical bug where TrackBase constructor was over-allocating space
for non-linear AudioTrack or 8-bit PCM AudioRecord (probably benign).
Change-Id: I7cbbba0bf4dba29ea751d8af341ab8e5cbbdc206
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 3b16c766d1ae2cfd8487e8ffb2b23936fc0a8e17 |
|
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Use uint32_t for sample rate
Change-Id: Ie240b48fb54b08359f69ecd4e5f8bda3d15cbe80
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 22eb4e239fbe9103568147d566d7482e480350b8 |
|
07-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Update audio comments
Change-Id: I85d7d2f6381b251db5695202fec75128883a8662
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 83a0382dc17364567667a4e6135db43f5bd92efc |
|
12-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Move frame size out of the control block
This is part of a series to clean up the control block.
Change-Id: Ifab1c42ac0f8be704e571b292713cd2250d12a3f
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| bc0f6b92bba33ca9c2e76f2a520d290f055da6b2 |
|
12-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Fix regression for AudioTrack::write() 8-bit PCM
Bug: 7526532
Change-Id: I8ddd1f0e9d035b54401788dcc422591281dcd97a
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| b929e417853694e37aba1ef4399f188987b709d9 |
|
08-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Move buffers pointer out of the control block
This is part of a series to clean up the control block.
Change-Id: Ie474557db7cb360f2d9a0f11600a68f5a3d46f07
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| a47f3165f53c8e8fb8907a94de7417e2c3047eeb |
|
07-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify AudioTrack::restoreTrack_l()
Remove CBLK_RESTORING and CBLK_RESTORED from control block flags,
for AudioTrack only. They are still used by AudioRecord.
This is part of a series to clean up the control block.
Change-Id: Iae4798f5b527c492bdaf789987ff3a1dadd0cb37
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 659004c2949620d8adb29e1d950a2dd1c75ba9a9 |
|
08-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Add all-channel AudioTrack::setVolume() API"
|
| 755e8e1dcf36f282b727ff22d87d3c2fac70d30a |
|
08-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix call to restoreTrack_l() without lock held"
|
| d5ed6e88a9bea1879e41d7defaf1edea7c09f554 |
|
02-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Fix call to restoreTrack_l() without lock held
Also document lock order
Change-Id: I2c1f273a0a51fa79ee3dd766de8d23083e270051
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 864585df53eb97c31e77b3ad7c0d89e4f9b42588 |
|
07-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Remove CBLK_DIRECTION from control block flags
This is part of a series to clean up the control block.
Change-Id: I0265fece3247356b585d4d48fbda6f37aea8a851
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 8fa3ad0e4c77f7868348ad2a4cacc34c34e7c39e |
|
07-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove CBLK_FAST from control block flags"
|
| e0b07179a48ee50fda931d2aa1b3c751d167e4d7 |
|
07-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Remove CBLK_FAST from control block flags
This is part of a series to clean up the control block.
Change-Id: Ic881a3560d9547cb63fcc0cefec87aa3da480e0d
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 05d499958e4030938ed77a924ebdd9899f36752e |
|
06-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Remove unused fields in AudioTrack::Buffer
Change-Id: Iab75f6e2348d8b6d1f3cec95aeb3fcd5135dfb50
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| d2c38fc4d5dc742d7441444316849510dd2b7363 |
|
01-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Cache mCblk in local variable cblk
Use "iMem" for sp<IMemory>
Change-Id: I2f1fbbc517fbd77cfc92f6c3b1f253c26bae93b0
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 8e825a57202ce6cb2ddac67683ee5885334ce75d |
|
06-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify control block flag names"
|
| b1c0993b215c5c3eebd1c6bafc22bba23d57a70b |
|
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Add all-channel AudioTrack::setVolume() API
Add combined channel APIs setVolume to AudioTrack,
and remove obsolete getVolume.
Change-Id: I0c87bfdbff4f4292259fa33e65f67badbafd270b
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 847d05dc8fa144dcf8f4f435d6a6ac1727f00937 |
|
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Remove deprecated AudioTrack APIs
Change-Id: I88be6525f3e33df529c0c3cb701d12a484809477
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 9c5fdd83f9b9f49be35107971feb33528d60b945 |
|
05-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify control block flag names
Use only one symbol per flag
Change-Id: Ia3582e2134abd60c896d11337face65383e79c7c
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 9a08ebc8de71e260efb86cd1a04559b075b38ebc |
|
02-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Fix spurious wakeup waiting for new IAudioTrack
If there was a spurious wakeup while waiting for another thread to create
a new IAudioTrack, we assumed that the track has been created when it
might not have been.
Change-Id: I5f3999b4f7a06a00aabd65a746cc7222fff396ab
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 85ab62c4b433df3f1a9826bed1c9bec07a86c750 |
|
01-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Line length 100
Change-Id: Ib28fd7b9ce951a6933f006e7f8812ba617625530
/frameworks/av/media/libmedia/AudioTrack.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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 73431968e3c5cf420b5d63c54cf72d10ff7ee7bc |
|
13-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Workaround for track recreation bug
Ignore thread can call Java parameter to work around bug where
AudioTrack instance gets recreated when key clicks are enabled
and WFD gets turned on, with the wrong parameters.
Change-Id: Ia42c8704b46fe3ffea560b05b60939fa2e4b29e1
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| c3ae93f21280859086ae371428ffd32f39e76d50 |
|
30-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Update audio comments
Change-Id: Ie7504d0ddb252f7e4d4f99ed0b44cfc7b1049816
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| a997e7a7c5aa2fc7e95316218316f8b7b98786ba |
|
07-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Revert "Swap the order of creating IAudioTrack and thread"
This reverts commit 5d464eb0b8cffb994a754ff108795e858a882414.
It caused the wrong thread ID -1 to be passed to IAudioFlinger::createTrack().
Change-Id: Ic221d2bb4af572d3d2d752af19238c52f6728e3a
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 5d464eb0b8cffb994a754ff108795e858a882414 |
|
23-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Swap the order of creating IAudioTrack and thread
Simplifies the error recovery in case IAudioTrack fails.
Change-Id: I6aee41a2ac747a5689fb4836b04174e6107bf32f
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 28b76b334f92a15a2be3cc9e2f7d229a3275d1ac |
|
04-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_channel_mask_t for channel mask
Change-Id: I1c1896da48983aa9f1462a4b471f910498816f60
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 04cd0186305e2b59d23c9147787046c6662029cc |
|
25-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
getMinFrameCount error handling
Convention is for "get" APIs that directly return status_t and indirectly
return a value via a pointer, to return BAD_VALUE if the pointer is NULL.
Also indirectly return 0 for other errors.
Change-Id: I1599f20ecb26e9723f9fb384ffbf911ff3a2ce1c
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| d4070955e28ae62aa4be1657f9d32acde104bb86 |
|
12-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dead code in libmedia
Change-Id: I7d8201590cda29c9fa99662a4fdba222091febfe
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| d3a9ff4b725de612bf0354c035ba8f8564dbc6e8 |
|
21-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Move declarations of local variables to first use
Change-Id: I48b193a742b32b6746aa938b84dc405124a6a5c3
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 192cbbad773979a6fe3b5a0c223356de4fc3309c |
|
13-Jun-2012 |
Eric Laurent <elaurent@google.com> |
Fix audio track pause.
AudioTrack::pause() should signal the control block condition
to release threads waiting for available buffers in obtainBuffer().
Otherwise the behavior relies on the timout on the condition
or the fact that audioflinger will mix a new audio buffer while executing
the pause.
Bug 6653769.
Change-Id: I5f8f73c471fe306070f30b814f32fd4b4dc1d575
/frameworks/av/media/libmedia/AudioTrack.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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| df839841d2db4cb8e2acb10205b3942622b3e7a2 |
|
31-May-2012 |
Eric Laurent <elaurent@google.com> |
Do not keep audio wake lock when apps underrun.
Do not automatically restart an AudioTrack after an underrun
if the callback is executed but no data is written by the app.
Bug 6541286.
Change-Id: I11e7ab8dc968d7ff087058fec68f44490d3a7731
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 093000f7d11839b920e8dfaa42ed1d09f48e24b8 |
|
03-May-2012 |
Glenn Kasten <gkasten@google.com> |
Don't allow AudioTrack frameCount to decrease
This is a workaround for bug that client can cache return value of
frameCount(), and is not notified when this value changes due to automatic
re-recreation of the underlying IAudioTrack.
A better long-term fix would be to notify clients when these kinds of
parameters change, and to fix assumptions in client code that they are
constant (e.g. in SoundPool and maybe obtainBuffer).
Also, once a fast track request is denied, don't request it again.
Bug: 6431187
Change-Id: I55b4ff30bbd9ed3a402e39452a38de52cdea53a9
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| f4022f90db5acb680870db8c1150b673cdd211d9 |
|
02-May-2012 |
Glenn Kasten <gkasten@google.com> |
Fix race in AudioTrack destruction
Bug: 6427369
Change-Id: Id3b4487406235b881f6f0b4b95c5a02a9b797e75
/frameworks/av/media/libmedia/AudioTrack.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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| a636433cbd09c0708b85f337ef45c0cdef3bcb4d |
|
19-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
Use C APIs instead of C++ APIs for policy
The C++ APIs are going away.
Note: we use tid == 0 which is not supported yet by the C APIs,
do not submit this until that is added.
Change-Id: I0e90789e6c81c69f2544e899c52421ea5d1342be
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| ca8b28013c0558a4a3323a1a0f58520277200086 |
|
23-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
Fix regression in AudioTrack::pause()
Bug: 6379646
Change-Id: I12b53bc4118499ddc73a53a981f3f56328140868
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 3acbd053c842e76e1a40fc8a0bf62de87eebf00f |
|
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Configure policy of mediaserver threads
Change-Id: Ifd825590ba36996064a458f64453a94b84722cb0
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 1948eb3ea6eee336e8cdab9b0c693f93f5f19993 |
|
14-Apr-2012 |
Eric Laurent <elaurent@google.com> |
Add support for deep audio buffers
Allow AudioSink to use deep audio buffering when the
source is audio only and its duration is more than
a certain threshold.
This helps improve battery life but implies higher
audio latency.
Change-Id: Ie79915b61c370292f05aabda9779356570e03cbb
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 0ca3cf94c0dfc173ad7886ae162c4b67067539f6 |
|
18-Apr-2012 |
Eric Laurent <elaurent@google.com> |
rename audio policy output flags
Change-Id: I27c46bd1d1b2b5f96b87af7d05b951fef18a1312
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 1a9ed11a472493cac7f6dfcbfac2064526a493ed |
|
21-Mar-2012 |
Eric Laurent <elaurent@google.com> |
audio policy: add configuration file
removed outputs to stream mapping cache in audio system: the output for a
given stream type must always be queried from audio policy manager as the cache
is not always updated fast enough by audioflinger callback.
removed AudioFlinger::PlaybackThread::setStreamValid() not used anymore if
stream to output mapping is not cached.
Change-Id: Ieca720c0b292181f81247259c8a44359bc74c66b
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 4a4a0959bca78e03e3c3f486ba17829c28314d8c |
|
19-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
AudioTrack client fast policy
Change-Id: I42ce691df3f586ac061b62237f35a263a0116f1f
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| a1472d9883e35edd280201c8be3191695007dfd4 |
|
30-Mar-2012 |
Marco Nelissen <marcone@google.com> |
Make AudioTrack/AudioRecord handle more than 2^32 frames
b/6160363
Change-Id: I471815012c6a113ec2c4dd7676e8fa288a70bc76
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| b83d38feeeb88a8a2a6219e1fca2480b5a14fb0d |
|
26-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "IAudioFlinger::createTrack and openRecord flags"
|
| 291f4d505aff81969e6666049d3cc3446f836af2 |
|
19-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Remove enforceFrameCount
It was only used to decide whether to issue a warning.
The warning was issued the first time track was created but
not at re-creation. Now it is a verbose message every time,
not a warning since it happens all the time with key clicks on A2DP.
Change-Id: I9d39f53c0a7eb84b666e55b1b76ff830cf8f37ba
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 63c1faa8dea7feb90255d31ef2a133d8f2818844 |
|
21-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Update comments"
|
| ea7939a079b3600cab955760839b021326f8cfc3 |
|
14-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Whitespace
Fix indentation, and add blank lines in key places for clarity
Change-Id: I57a0a8142394f83203161aa9b8aa9276abf3ed7c
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 17a736c3e1d062d7fc916329eb32aef8935614af |
|
14-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Update comments
Change-Id: I327663a020670d0a72ff57bd0b682e2ce0528650
/frameworks/av/media/libmedia/AudioTrack.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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| e53b9ead781c36e96d6b6f012ddffc93a3d80f0d |
|
13-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Whitespace and indentation
Fix indentation to be multiple of 4.
Make it easier to search:
sp< not sp < to
"switch (...)" instead of "switch(...)" (also "if" and "while")
Remove redundant blank line at start or EOF.
Remove whitespace at end of line.
Remove extra blank lines where they don't add value.
Use git diff -b or -w to verify.
Change-Id: I966b7ba852faa5474be6907fb212f5e267c2874e
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| d8d6185c978c9b27ca69e7791785e0983ed9e8b8 |
|
06-Mar-2012 |
Eric Laurent <elaurent@google.com> |
AudioTrack: relax check on minimum buffer size
Current AudioTrack implementation enforces that the requested audio
buffer size is at least corresponding the audio latency.
This requirement is too strong and leads to problems with current
stagefright and AudioSink implementations when playing over output
streams with long latency.
Ultimately, the AudioSink design should be changed to specify a minimum
buffer size in time or frames units but not in buffer count units.
Change-Id: I8ba603956f92ac49143a8249572665aa548f2f0f
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 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
/frameworks/av/media/libmedia/AudioTrack.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>
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| ed15977476a3d53103866e6d527fa3fb65d4166c |
|
14-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove dead code AudioTrack::getLoop"
|
| 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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 38f5d71e72f3b76c5b519614d27f051d53cd2712 |
|
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dead code AudioTrack::getLoop
Change-Id: I868329c52f31bc20125f068500d8f892b4ec9796
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 4f9b0c54011eb8fd2ccfb393c2dcd51cd07800e0 |
|
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove dead mutex in AudioTrack/AudioRecord thread"
|
| a5224f319e2ba4b51ddb4287705ccf8d4b8ecc51 |
|
04-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
AudioTrack declare more methods const
Change-Id: I4999e984460893961d0d8092cff17f3cf07d7214
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| a0d68338a88c2ddb4502f95017b546d603ef1ec7 |
|
28-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use NULL not 0 for raw pointers
Use if (p != NULL) instead of if (ptr)
Change-Id: Iaec3413a59ccbf233c98fcd918cc7d70ac5da9fa
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 0e265cf36d201a7ccc0238b5c60b50f43d1dc450 |
|
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dead mutex in AudioTrack/AudioRecord thread
The client callback threads had mutexes called AudioTrackThread::mLock
and ClientRecordThread::mLock. These mutexes were only used by start()
and stop(), and were unused by the thread itself. But start() and
stop() already have their own protection provided by AudioTrack::mLock
and AudioRecord::mLock. So the thread mutexes can be removed.
Change-Id: I098406d381645d77fba06a15511e179a327848ef
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 05bfe50e13793404a78c20c850d467d17734d496 |
|
20-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Move memcpy_to_i16_from_u8 to audio_utils"
|
| c813985abd8ba61e999b3505f6a332574f87a1be |
|
18-Jan-2012 |
Andreas Huber <andih@google.com> |
Temporarily restore AudioSystem/AudioTrack APIs with their former signatures
until we get updated prebuilts from vendor.
Change-Id: I8aae81d2513edca0ab268053a11c8c4206879e61
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 511754b5839fd9b09fc56b89ae007fbc39084a33 |
|
11-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Move memcpy_to_i16_from_u8 to audio_utils
This will make it easier for this kind of code to be optimized
for each target architecture.
Change-Id: I9efd27d6c0175b00b9a784353244805cec63c0b8
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 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"
|
| 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
/frameworks/av/media/libmedia/AudioTrack.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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| fff6d715a8db0daf08a50634f242c40268de3d49 |
|
13-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_stream_type_t consistently
At native level it was a mixture of audio_stream_type_t, int, uint32_t,
and uint8_t. Java is still int. Also fixed a couple of hard-coded -1
instead of AUDIO_STREAM_DEFAULT, and in startToneCommand a hard-coded 0
instead of AUDIO_STREAM_VOICE_CALL.
Change-Id: Ia33bfd70edca8c2daec9052984b369cd8eee2a83
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 4cc55d53d542c4f4ed645738cebb65b9e7eb6c44 |
|
12-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use audio_format_t consistently"
|
| 83bc7f3cf78b28a818417f40a4f0c00593993366 |
|
04-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
libmedia new can't fail on Android
Change-Id: Ie79dd5abb8078b35474bf0f1b3a6ff994a3a3360
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 5c7b3bcc88e2b472f1f7b416d89222714b96a567 |
|
09-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Clean up AudioTrack::mActive and stopped()"
|
| 29357bc2c0dd7c43ad3bd0c8e3efa4e6fd9bfd47 |
|
06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/#/c/157220
Bug: 5449033
Change-Id: Ic9c19d30693bd56755f55906127cd6bd7126096c
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 5ff1dd576bb93c45b44088a51544a18fc43ebf58 |
|
06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/157065
Bug: 5449033
Change-Id: I00a4b904f9449e6f93b7fd35eac28640d7929e69
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| e1c3962e268ffc12bfd1bd9ea84da1f135f36960 |
|
04-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_format_t consistently
Was int, uint32_t, uint16_t, and uint8_t with 2-bit bitfield.
Also replace 0 by AUDIO_FORMAT_DEFAULT and replace 1 by
AUDIO_FORMAT_PCM_16_BIT.
Change-Id: Ia8804f53f1725669e368857d5bb2044917e17975
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 9a2aaf927e56a4b4acab23ef16b3f133a9f48a63 |
|
03-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Clean up AudioTrack::mActive and stopped()
mActive is protected by mLock; volatile is meaningless on SMP.
Fixed a couple of places where mActive was accessed without a lock:
- stopped()
- processAudioBuffer()
Added stopped_l() for cases where we already hold the lock.
Made mActive a bool not int.
Moved down a lock in setPosition that was being acquired too early.
Change-Id: I73ff368e991c0db9f9472df0b3f96fd33fcc7311
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 42968939dfce0954d6540011199045ec4ed7de80 |
|
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix indentation and whitespace"
|
| 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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 1d334101f1289cf4c8967af6e78ac22619175982 |
|
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove unnecessary this->"
|
| 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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 91eb8bfbe253a6b6fe1aa23fb884a601c28991c4 |
|
13-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Remove redundant clear()
Change-Id: Ie5e4e63cbc8fa85ef50451dddf8f149fa864b132
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 9054897ab9ffb307fadae81b774a1fc61cb542e7 |
|
13-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Remove unnecessary this->
Change-Id: I72038f5d4568f0633d3e4ab90f4b67e2dd22c332
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| f0c495012bad92230604a9a12a907812ec49ee8f |
|
30-Nov-2011 |
Glenn Kasten <gkasten@google.com> |
AudioTrack::setVolume check range
Change-Id: Ie182bf0f741f1f49f68c02a1e7437a2a34d34fc5
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 879135196fd1c97deefc538c888037c56c2879a7 |
|
23-Jun-2011 |
Glenn Kasten <gkasten@google.com> |
Bug 4903178 Restore priority and cgroup on stop
On AudioTrack and AudioRecord stop or failed start, restore the priority
and cgroup of the caller to their previous values, rather than forcing
to NORMAL. Dependent on new thread APIs.
Also fixes bug where priority was set to AUDIO but cgroup not set.
Change-Id: Ib83893918fb4fdf57c6b87884b51038997a631d8
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 3856b090cd04ba5dd4a59a12430ed724d5995909 |
|
20-Oct-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/#/c/143865
Bug: 5449033
Change-Id: I0122812ed6ff6f5b59fe4a43ab8bff0577adde0a
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| cfe2ba6b01a258e39f9c215ffc7b750e0b68f708 |
|
14-Sep-2011 |
Eric Laurent <elaurent@google.com> |
Issue 5298399: Lost speech after a crash in gTalk.
Fixed problem in AudioTrack::restoreTrack_l() causing a permanent
failure if the IAudioTrack interface to AudioFlinger could not be
restored at the first attempt.
Change-Id: I039d4fe2dca8d3baf71f1a6c51119f27a67b6611
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 2267ba18d0d2b2d4bd7f5411821ad89ac2659a88 |
|
07-Sep-2011 |
Eric Laurent <elaurent@google.com> |
AudioTrack: extend callback thread sleep time
Do not force wake up the AudioTrack thread every 10ms if no timed
events (loop, markers..) have to be processed.
This will help reduce power consumption.
Change-Id: Icb425b13800690008dd07c27ffac84739e3dbba3
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 408b8dc3c0a364c6f6b4991d15da9e6bcc2b8008 |
|
06-Sep-2011 |
Eric Laurent <elaurent@google.com> |
Issue 5247986: Battery drain due to audio wakelock
The problem occurs when activating or deactivating A2DP connection
while SoudPool has a channel active. This can happen quite frequently now
that the UI sound effects are enabled by default.
If PCM data is remaining in the AudioTrack buffer when it is restroyed and
re-created on the new AudioFlinger output thread, this data is flushed.
As a consequence, no underrun or request for new data callback is sent to
SoundPool and the sound channel remains active for ever as the end of the
sample is never detected.
Change-Id: I13e0c11e4ce3f83bff7f58d347ca814b6a86712b
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 9f6530f53ae9eda43f4e7c1cb30d2379db00aa00 |
|
30-Aug-2011 |
Eric Laurent <elaurent@google.com> |
226483: A2DP connected, but music out to speaker
When the A2DP headset is connected, there is a possible
race condition when the audio tracks are moved from
the mixer thread attached to the speaker output to the thread
attached to A2DP output.
As the request to clear the stream type to output mapping cache in
the client process is asynchronous, it is possible that the flag
indicating to the client audio track to re-create the IAudioTrack
on the new thread is processed before the cache is invalidated.
In this case, the track will be attached to the old thread and
music will continue playing over the device speaker instead of being
redirected to A2DP headset.
Change-Id: Ib2ce1eb5320eaff83287b93779061bf4e7a330df
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| cd07594333cbe8b2c86c6609cce01a74d6cc33f8 |
|
26-Aug-2011 |
Jean-Michel Trivi <jmtrivi@google.com> |
Bug 4364249 Play position is 0 after flushing AudioTrack
AudioTrack::stop() is not synchronous, so a stop() followed
by flush(), which is synchronous, will not always report
a playhead position of 0 after being called.
This CL adds a flag to mark a track as flushed, and report the
correct playhead position in this state.
Bug 5217011 has been created to address the real issue in the
future, where flush could be made synchronous, to properly
address bug 4364249.
Change-Id: Icf989d41a6bcd5985bb87764c287f3edb7e26d12
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 3a34befc6fb04a4945a849e8bda8b84e4bf973fe |
|
02-Aug-2011 |
Marco Nelissen <marcone@google.com> |
Keep effects sessions active when the caller dies.
Don't remove effects until the session they are in goes away or all
AudioEffects have been explicitly released. This allows the control
panel process to die without stopping the effects.
Change-Id: I4496e5df080230ca1af149dec95c1309ab8ea888
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 68cbeba4e21aa53f52fb99b74dfa1910af31a3eb |
|
23-Jun-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix warnings for uninitialized local variables"
|
| d0965dde97f2815ae0a15fe6b40946f8a741a81e |
|
23-Jun-2011 |
Glenn Kasten <gkasten@google.com> |
Fix warnings for uninitialized local variables
Change-Id: Ic9b03b0fd215444e76c7b7bebb385f7831c557e0
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 671a636931295d9c33ffca74551a804479d01241 |
|
17-Jun-2011 |
Eric Laurent <elaurent@google.com> |
Added audio_bytes_per_sample() helper function
Change-Id: Ibfcd75c4c241a53d5f052c25ada091904991048a
/frameworks/av/media/libmedia/AudioTrack.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>
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| c6854100cea4fcd0f20cb2ac8235c02d1849b3a1 |
|
02-Jun-2011 |
Glenn Kasten <gkasten@google.com> |
Remove unnecessary level of priority indirection
Change-Id: I942d43973c20a7ace8b0d3f78b4da97e45e996c6
/frameworks/av/media/libmedia/AudioTrack.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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 64760240f931714858a59c1579f07264d7182ba2 |
|
11-May-2011 |
Dima Zavin <dima@android.com> |
update for new audio.h header location
Change-Id: Ic4c62c4037800802427eb7d3c7f5eb8b25d18876
Signed-off-by: Dima Zavin <dima@android.com>
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| fce7a473248381cc83a01855f92581077d3c9ee2 |
|
20-Apr-2011 |
Dima Zavin <dima@android.com> |
audio/media: convert to using the audio HAL and new audio defs
Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
/frameworks/av/media/libmedia/AudioTrack.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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 9b7d950f1f3b0c526712b713dbceb0e22762c015 |
|
21-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3483718: audio streaming and A2DP.
The problem is that when switching from A2DP to device speakers or headset,
The AudioTrack binder interface to AudioFlinger must be destroyed and restored
to accomodate new buffer size requirements. Current AudioTrack implementation
did not restore properly the PCM buffer write index which caused a mismatch between
the written frame count in the mediaplayer renderer and the AudioTrack. The renderer
could then believe the AudioTrack buffer was full and stop writing data preventing the
AudioTrack to reach a bufffer full condition and resume playback.
The rendered was also modified to refresh the AudioTrack frame count (buffer size)
inside the write loop in NuPlayer::Renderer::onDrainAudioQueue() as this count can change
from one write to the next.
Also modified AudioTrack::obtainBuffer() to check for track invalidated status before
querying for available space in the buffer. This avoids writing to the old track's
buffer until full before detecting the invalidated condition and create a new track.
Change-Id: I16a857e464e466880847f52f640820aa271539ad
/frameworks/av/media/libmedia/AudioTrack.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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 1703cdfee717b1b312bf8979816a9e2f16a82e5d |
|
07-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3439872: video chat and bluetooth SCO
This change fixes the stability problems experienced when using
a bluetooth headset supporting both A2DP and SCO. Problems occur
when starting the video chat at which time the A2DP output is being
stopped to start SCO. At that time, active AudioTracks are invalidated
by AudioFlinger so that a new AudioTrack binder interface can be
recreated by the client process on the new mixer thread with correct parameters.
The problem was that the process to restore the binder interface was not
protected against concurrent requests which caused 2 binder interfaces
to be created sometimes. This could lead to permanent client deadlock
if one of the client threads was waiting for a condition of the first
created binder interface while the second one was created (as the AudioFlinger
would only signal conditions on the last one created).
This concurrent request situation is more likely to happen when a client
uses the JAVA AudioTrack as the JNI implementation uses simultaneously the
native AudioTrack callback and write push mechanisms. By doing so, the code
that checks if the binder interface should be restored (in obtainBuffer()) is
much more likely to be called concurrently from two different threads.
The fix consists in protecting the critical binder interface restore phase
with a flag in the AudioTrack control block. The first thread acting upon the binder
interface restore request will raise the flag and the second thread will just wait for
a condition to be signaled when the restore process is complete.
Also protected all accesses to the AudioTrack control block by a mutex to prevent
access while the track is being destroyed and restored. If a mutex cannot be held
(e.g because we call a callback function), acquire a strong reference on the IAudioTrack
to prevent its destruction while the cblk is being accessed.
Modified AudioTrack JNI to use GetByteArrayElements() instead of
GetPrimitiveArrayCritical() when writing audio buffers. Entering a critical section would
cause the JNI to abort if a mediaserver crash occurs during a write due to the AudioSystem
callback being called during the critical section when media server process restarts.
Anyway with current JNI implementation, either versions do not copy data most of the times
and the criticial version does not guaranty no data copy.
The same modifications have been made to AudioRecord.
Change-Id: Idc5aa711a04c3eee180cdd03f44fe17f3c4dcb52
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 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
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 2beeb50b1bba9e92f6cacfeca37fe9fa9d36ead1 |
|
16-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Added support for auxiliary audio effects to AudioTrack and MediaPlayer.
Added methods to AudioTrack and MediaPlayer java classes to enable use of
auxiliary audio effects. The effect can be attached and detached by specifying its
ID and the send level controlled.
Change-Id: Ie74ff54a453096a742688476f612ce355543b6f3
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| a514bdb58b5de4986679f72b7204b4764f7a2778 |
|
21-Jun-2010 |
Eric Laurent <elaurent@google.com> |
Added support for audio sessions in MediaPlayer and AudioTrack.
Audio sessions are used to associate audio effects to particular instances (or groups) of MediaPlayers or AudioTracks.
Change-Id: Ib94eec43241cfcb416590f435ddce7ab39a07640
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 33005a932c60a0780fe9b7307d5988df3d9f6c26 |
|
16-Jun-2010 |
Chia-chi Yeh <chiachi@android.com> |
media: add AudioTrack::getMinFrameCount().
Change-Id: If15162583d1d16d89f59be0793106afe77417f35
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| be916aa1267e2e6b1c148f51d11bcbbc79cb864c |
|
02-Jun-2010 |
Eric Laurent <elaurent@google.com> |
Issue 2667801: [Audio Effect Framework] AudioFlinger, AudioMixer AudioTrack modifications.
First drop of audio framework modifications for audio effects support.
- AudioTrack/AudioRecord:
Added support for auxiliary effects in AudioTrack
Added support for audio sessions
Fixed left right channel inversion in setVolume()
- IAudioFlinger:
Added interface methods for effect enumeraiton and instantiation
Added support for audio sessions.
- IAudioTrack:
Added method to attach auxiliary effect.
- AudioFlinger
Created new classes to control effect engines in effect library and manage effect connections to tracks or
output mix:
EffectModule: wrapper object controlling the effect engine implementation in the effect library. There
is one EffectModule per instance of an effect in a given audio session
EffectChain: group of effects associated to one audio session. There is one EffectChain per audio session.
EffectChain for session 0 is for output mix effects, other chains are attached to audio tracks
with same session ID. Each chain contains a variable number of EffectModules
EffectHandle: implements the IEffect interface. There is one EffectHandle object for each application
controlling (or using) an effect module. THe EffectModule maintians a list of EffectHandles.
Added support for effect modules and effect chains creation in PlaybackThread.
modified mixer thread loop to allow track volume control by effect modules and call effect processing.
-AudioMixer
Each track now specifies its output buffer used by mixer for accumulation
Modified mixer process functions to process tracks by groups of tracks with same buffer
Modified track process functions to support accumulation to auxiliary channel
Change-Id: I26d5f7c9e070a89bdd383e1a659f8b7ca150379c
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| d1b449aad6c087a69f5ec66b7facb2845b73f1cb |
|
14-May-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2553359: Pandora does not work well with Passion deskdock / Cardock.
The problem is due to a too big difference between the buffer size used at the hardware interface and at the A2DP interface.
When no resampling occurs we don't notice problems but the timing is very tight. As soon as resampling is activated, the AudioTrack underruns.
This is because the AudioTrack buffers are not resized when moving the AudioTrack from hardware to A2DP output.
The AudioTrack buffers are calculated based on a hardware output buffer size of 3072 bytes. Which is much less than the A2DP output buffer size (10240).
The solution consists in creating new tracks with new buffers in AudioFlinger when the A2DP output is opened
instead of just transfering active tracks from hardware output mixer thread to the new A2DP output mixer thread.
To avoid synchronization issues between mixer threads and client processes, this is done by invalidating tracks
by setting a flag in their control block and having AudioTrack release the handle on this track (IAudioTrack)
and create a new IAudioTrack when this flag is detected next time obtainBuffer() or start() is executed.
AudioFlinger modifications:
- invalidate the tracks when setStreamOutput() is called
- make sure that notifications of output opening/closing and change of stream type to output mapping are sent synchronously to client process.
This is necessary so that AudioSystem has the new stream to output mapping when the AudioTrack detects the invalidate flag in the client process.
Previously their were sent when the corresponding thread loop was executed.
AudioTrack modifications:
- move frame count calculation and verification from set() to createTrack() so that is is updated every time a new IAudioTrack is created.
- detect track invalidate flag in obtainBuffer() and start() and create a new IAudioTrack.
AudioTrackShared modifications
- group all flags (out, flowControlFlag, forceReady...) into a single bit filed to save space.
Change-Id: I9ac26b6192230627d35084e1449640caaf7d56ee
/frameworks/av/media/libmedia/AudioTrack.cpp
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| 54b1a0550411c2fd2084d82d28934d505c37349a |
|
20-Mar-2010 |
Mathias Agopian <mathias@google.com> |
libutils Condition are now PRIVATE by default
Condition must be initialized with SHARED for the old behavior, where
they can be used accross processes.
Updated the two places android that require SHARED conditions.
PRIVATE conditions (and mutexes) use more efficient syscalls.
Change-Id: I9a281a4b88206e92ac559c66554e886b9c62db3a
/frameworks/av/media/libmedia/AudioTrack.cpp
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| 867d2f6ce668968e463eb86b856d21525f12fd67 |
|
26-Jan-2010 |
Mathias Agopian <mathias@google.com> |
Simplify the MemoryDealer implementation
At some point the implementation became complicated because of
SurfaceFlinger's special needs, since we are now relying on gralloc
we can go back to much simpler MemoryDealer.
Removed HeapInterface and AllocatorInterface, since those don't need
to be paramterized anymore. Merged SimpleMemory and Allocation.
Made SimplisticAllocator non virtual.
Removed MemoryDealer flags (READ_ONLY, PAGE_ALIGNED)
Removed a lot of unneeded code.
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 6100d2d60517ff33ed8eb35d0b7ea63cde0831c9 |
|
19-Nov-2009 |
Eric Laurent <elaurent@google.com> |
Issue 2265163: Audio still reported routed through earpiece on sholes
This is a second attempt to fix the audio routed to earpiece syndrom.
The root cause identified this time is the crash of an application having an active AudioTrack playing on the VOICE_CALL stream type.
When this happens, the AudioTrack destructor is not called and the audio policy manager is not notified of the track stop.
Results a situation where the VOICE_CALL stream is considered as always in use by audio policy manager which makes that audio is routed to earpiece.
The fix consists in moving the track start/stop/close notification to audio policiy manager from AudioTrack to AudioFlinger Track objet.
The net result is that in the case of a client application crash, the AudioFlinger TrackHandle object (which implements the remote side of the IAudioTrack binder interface) destructor is called which in turn destroys the Track object and we can notify the audio policy manager of the track stop and removal.
The same modification is made for AudioRecord although no bug related to record has been reported yet.
Also fixed a potential problem if record stop is called while the record thread is exiting.
/frameworks/av/media/libmedia/AudioTrack.cpp
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| 2b584244930c9de0e3bc46898a801e9ccb731900 |
|
10-Nov-2009 |
Eric Laurent <elaurent@google.com> |
Improvements for issue 2197683: English IME key-press latency is noticeably higher on passion than sholes
This change goes with a kernel driver change that reduces the audio buffer size from 4800 bytes (~27ms) to 3072 bytes (~17ms).
- The AudioFlinger modifcations in change 0bca68cfff161abbc992fec82dc7c88079dd1a36 have been removed: the short sleep period was counter productive when the AudioTrack is using the call back thread as it causes to many preemptions.
- AudioFlinger mixer thread now detects long standby exit time and in this case anticipates start by writing 0s as soon as a track is enabled even if not ready for mixing.
- AudioTrack::start() is modified to start call back thread before starting the IAudioTrack so that thread startup time is masked by IAudioTrack start and mixer thread wakeup time.
/frameworks/av/media/libmedia/AudioTrack.cpp
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| 34f1d8ecd23169a5f299937e3aaf1bd7937578a0 |
|
04-Nov-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 2203561: Sholes: audio playing out of earpiece.
Create a new IAudioTrack interface to AudioFlinger when start() fails due to a broken pipe error.
Do the same if start fails due to the same error after time out in obtainBuffer().
Do not indicate that the AudioTrack is started to AudioPolicyManager if IAudioTrack start fails.
This avoids that an AudioTrack keeps a dead IAudioTrack after a media server crash.
Same modifications for AudioRecord.
Add a flag to ToneGenerator indicating that the callback thread can call Java. Without it, when the media server crashes and restarts, the AudioSystem error callback will crash in JNI if the IAudiotrack is created from AudioTrack callback thread.
/frameworks/av/media/libmedia/AudioTrack.cpp
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| 3302526f6276911b2dc40c731ea7fa0e7972d908 |
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04-Aug-2009 |
Eric Laurent <elaurent@google.com> |
Fix problem in AudioTrack with 8 bit PCM and direct output.
Do not perform 8 to 16 bit conversion in AudioTrack write() and processAudioBuffer() if direct output flag is set.
/frameworks/av/media/libmedia/AudioTrack.cpp
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| c2f1f07084818942352c6bbfb36af9b6b330eb4e |
|
17-Jul-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 1795088 Improve audio routing code
Initial commit for review.
Integrated comments after patch set 1 review.
Fixed lockup in AudioFlinger::ThreadBase::exit()
Fixed lockup when playing tone with AudioPlocyService startTone()
/frameworks/av/media/libmedia/AudioTrack.cpp
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| b07c28b90b2d2793be2b8878d813b607f3eebbb7 |
|
14-Jul-2009 |
Mathias Agopian <mathias@google.com> |
add a ctor to Mutex to specify the type, which can be shared. This is used by sf and af an soon will allow some optimization in the kernel for non shared mutexes
/frameworks/av/media/libmedia/AudioTrack.cpp
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| 349dba337e07e129f6ba49a132999f0b73fedbe3 |
|
07-Jul-2009 |
Eric Laurent <elaurent@google.com> |
am 88e209dc: Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR
Merge commit '88e209dcf8c2ebddda5c272f46d1bd5478bc639c'
* commit '88e209dcf8c2ebddda5c272f46d1bd5478bc639c':
Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR
|
| 573266210fb2b2e7d86fbd46d0dfe16763611d91 |
|
07-Jul-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR
Store sample rate on 32 bits instead of 16 bits in audio_track_cblk_t.
Removed sampleRate() methods from AudioTrack and AudioRecord: replaced by getSampleRate().
AudioTrack::setSampleRate() no returns a status.
/frameworks/av/media/libmedia/AudioTrack.cpp
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| 7562408b2261d38415453378b6188f74fda99d88 |
|
20-May-2009 |
Mathias Agopian <mathias@google.com> |
move libbinder's header files under includes/binder
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 1dd70b9f04961a06fcb73a97fca10a53b3245d3c |
|
21-Apr-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 1745312: Various cleanups in media framework
AudioTrack, AudioRecord:
- remove useless mAudioFlinger member of AudioTrack and AudioRecord.
- signal cblk.cv condition in stop() method to speed up stop completion.
- extend wait condition timeout in obtainBuffer() when waitCount is -1 to avoid waking up callback thread unnecessarily
AudioFlinger:
- remove some warnings in AudioFlinger.cpp.
- remove function AudioFlinger::MixerThread::removetrack_l() as its content is never executed.
- remove useless call to setMasterVolume in AudioFlinger::handleForcedSpeakerRoute().
- Offset VOICE_CALL stream volume to reflect actual volume that is never 0 in hardware (this fix has been made in the open source): 0.01 + v * 0.99.
AudioSystem.java:
- correct typo in comment
IAudioflinger, IAudioFlingerClient:
- make AudioFlinger binder interfaces used for callbacks ONEWAY.
AudioHardwareInterface:
- correct routeStrings[] table in AudioHardwareInteface.cpp
/frameworks/av/media/libmedia/AudioTrack.cpp
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| 2c22aeb65e801f663a754d043062f85e49f77739 |
|
25-Mar-2009 |
Jean-Michel Trivi <> |
Automated import from //branches/donutburger/...@140663,140663
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 1179bc9b0e3d17c984e8f4ad38561c049dd102fa |
|
19-Mar-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //branches/cupcake_rel/...@140373
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 8555d0867c3e8fe6cc5c7ad40af557fe6b92fa72 |
|
05-Mar-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //depot/cupcake/@136594
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 89fa4ad53f2f4d57adbc97ae1149fc00c9b6f3c5 |
|
04-Mar-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //depot/cupcake/@135843
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 15f767b960b38059a74a42a33e16d8df2aec8bc1 |
|
04-Mar-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //depot/cupcake/@135843
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| ad04d9201452001dbaac4349f084cc9316190b89 |
|
03-Mar-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //depot/cupcake/@132589
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 99ffda877980468a9ae31e013cd10fb3645df1b0 |
|
03-Mar-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //depot/cupcake/@137055
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 25658fd43d150a45fb37734a9f9f27f48bb5c133 |
|
19-Feb-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //branches/cupcake/...@132276
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 7a2146d5807030b2629f347736be5301b61e8811 |
|
13-Feb-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //branches/cupcake/...@131421
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 5e07b5774c8b376776caa4f5b0a193767697e97e |
|
11-Feb-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //branches/cupcake/...@130745
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| e5198b620a9a208ec59ea8457282404725f8ff6e |
|
20-Jan-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //branches/cupcake/...@127101
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| cce8bd12da6d8419a8770e4552a51ec297c250c2 |
|
10-Jan-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //branches/cupcake/...@125939
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 7b5eb023f8d87cca6d830ae6c11c6aadbe02aca8 |
|
18-Dec-2008 |
The Android Open Source Project <initial-contribution@android.com> |
Code drop from //branches/cupcake/...@124589
/frameworks/av/media/libmedia/AudioTrack.cpp
|
| 2729ea9262ca60d93047e984739887cfc89e82eb |
|
21-Oct-2008 |
The Android Open Source Project <initial-contribution@android.com> |
Initial Contribution
/frameworks/av/media/libmedia/AudioTrack.cpp
|