| Commit message (Collapse) | Author | Age | Files | Lines |
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* Brings us current with AU_LINUX_ANDROID_JB_2.5.04.02.02.040.367
Camera: Fix deadlock due to mLock in pcb and takepicture
In non-zsl case of takepicture, we do streamoff for preview
stream which is waiting on preview callback thread to exit.
By that time the lock has already been acquired by takePicture.
So preivew callback will not exit until it acquires lock and
takePicture cannot continue until PCB call back is returned.
Fix: Avoid the mLock at services when both Preview cb &
Compressed cb are enabled.
Change-Id: I6c264928bf1540c7b51f1add65f9c3e968506e15
CRs-fixed: 479419
audioflinger: Fix the LPA-AudioEffects crash issue
- Issue:crash is observed during LPA playback on enabling
effects followed by plug-out->plug-in of wired headset
- Rootcause: while deleteing the effectchain in deleteEffect
EffctChain is being unlocked after clearing the chain
which leads to accessing the lock which might already deleted.
- Fix: first unlock the effectChain and then call clear
CRs-Fixed: 491774
Change-Id: I518ff086c5ad71486cd29142563145137ebc15b6
libstagefright: Fix for crash in sound recorder during device switch
-Crash seen in sound recorder during frequent insertion and removal
of wired headset
-During device switch some time Codec's input buffers are too small to
accomodate buffer read from source. Omx codec doesn't read the fix size
buffer from source, during device switch scenario sometime buffer read
from source exceeds input buffer size so it goes in error state which
leads to crash.
-Increasing the input buffer size fix this issue
Change-Id: Id15378670880d0c3c0bd4408841b28be963549a0
CRs-Fixed: 488449
libstagefright: Fix for FPS drop issue during A-V playback.
Issues:
-The AAC decoder was not updating the timestamp when EOS is reached.
-Logic to smoothen the real time update in AudioPlayer uses system
time. This introduces corrupt timestamp during EOS.
Fix:
-Update the timestamp in AAC decoder when EOS is reached.
-Extrapolate realtime using system time in AudioPlayer when EOS is
reached. Cap the value to realtime if extrapolated time becomes greater
than realtime.
CRs-Fixed: 384183
Change-Id: Ice54501436431d2527fcd3d710d65d9732fcffdd
libstagefright: Reset buffer size value with SurfaceTexture
- OMXCodec explicitly sets the decoder output buffer size using the
native window perform API. (to accomodate extra-data)
- This size is reset only when the SurfaceTexture is destroyed.
- Unless reset, this size will be assumed for all output buffers
if the SurfaceTexture is re-used.
CRs-Fixed: 337660, 432309
Change-Id: I28aed12ad02adeac61caffbb00e3082640a5f6d4
audio: Add support for tunnel mode recording
- Add support for tunnel mode recording.
Change-Id: I95cdfff729affd784141487521c9f2f714221d11
audio: Add support for non-pcm VOIP vocoders
- Add support for non-pcm VOIP vocoders
- non-pcm vocoders use AUDIO_SOURCE_VOICE_COMMUNICATION
as inputSource. Add check to verify inputSource and
then configure framecount accordingly
Change-Id: Ia38da4f6ba0ee40c794d3c97325327cdb7dcb32a
CRs-Fixed: 467850
frameworks/av: Add metadata mode changes to LPAPlayer
-Seek to EOS was causing playback to hang for 3 seconds before
switching to the next clip.
-This is because the lpa driver works on period size. Partial
buffers are not handled.
-Add support for metadata mode changes to LPAPlayer to support
partial frames.
CRs-Fixed: 458904
Change-Id: I8673756b54ae7bca18855d326c85ae1064652514
libstagefright: Add support for WMA in ACodec
- WMA support is not there in ACodec
- In the case of wma format, since not getting the complete information of
wma version so instead of allocating the component in onAllocateComponent
function it will create in onConfigureCompoenent function.
bitspersample is find as "bsps" from AMessage while configuring the
WMA10PRO and WMALOSSLESS format
CRs-Fixed: 453951
Change-Id: I98baa701dbf8a5c012f4be5e83831c0be2111dcc
libstagefright: Flush the pending buffers when EOS is received
For the use case where the first frame in the buffer is EOS, decode
the aac config frame buffer to update the sample rate and channel
mode and flush out the buffer.
Change-Id: I0354802cdbf61ac1ba0fecbbdf616705806b0f4a
CRs-Fixed: 459334
audio: Fix The Linux Foundation copyright
- Fix copyright format based on The Linux
Foundation copyright template
Change-Id: I100a5c86302d1a1a3d79543d95e242734daae746
media, audioflinger: check for divide by zero possibilities and err
When output stream is not available to audioflinger due to any reason
, sampleRate and frameCount have zero values when trying to create
new Audiotrack. This might result in divide by 0 situation.
Change-Id: Ic13cb51facb8497e68ab596abb027b44f496b907
CRs-Fixed: 478480
framewroks/av:Fix ANR at the end of video recording
- While doing video recording, when the recording
ends ANR observed while doing stress test for
many hours
- When the recording is stopped, audio HAL receives error
from driver and audio HAL propagates this error to
AudioFlinger. But AudioFlinger is not sending error
status to audio source to stop recording. Because of
this audiorecord thread keeps on waiting for buffers
which is resulting in ANR.
- To avoid indefinite wait, a timeout of 1 sec is set for buffer
in audioSource and after timeout, -ETIMEDOUT is returned
to recorder thread.
CRs-Fixed: 479968
Change-Id: I91aba6922086e711992d9d991dea9c35d33eaee9
audioflinger: Integrate SRS TruMedia
Change-Id: If61ae91556120ddd5f5ebcc6dbbfe6583c7df67d
audioflinger: Fix apply SRS effects if tones diabled in tunnel mode
For the use case of SRS post processing in Tunnel mode, the API's
of SRS are called only from write. With the huge buffering for
tunnel mode, once EOS is received there would not be further write.
With system tone enabled, the SRS API's are called during the
check for Parameters change through normal mixer thread.
With system tones disabled, SRS will not be applied after EOS as
no write and mixer thread would not be active.
Fix the issue by adding the Effects Thread for SRS in Tunnel mode.
Fix the compilation issue with ALOGV messages enabled
Change-Id: Ic7e62894840f786119dfe8ae471c5d24812917d7
audioflinger: Enhance LPA-effect logic to handle rapid config.
-Issue:Rapid Config events cause pops/glitches, raw data
playback.
-Rootcause1:Raw data leakage to DSP: applyEffectsOn() applies
effects chunk by chunk in a loop, if effects change during
this time the loop exits and this results in creation of
a buffer in which part of it is effects processed and rest
raw, this causes raw data to leak to DSP.
-RootCause2:Effectsthread directly works on the DSP buffers,
while DSP is rendering from there, so that effect application
is instantaneous and for this it gives the DSP buffers as
output to effects chain, this means that all the effects in
the chain update the DSP buffers one after the other, this
can create unpredictable rendering patterns.
RootCause1 and 2 combined seem to fragment memory with
parts of it with effects and parts with raw data etc.
-Fix1:Dont update DSP mem unless the effects are applied
completely on a buffer.
-Fix2:Effectschain will work on a temp scrath buffer
instead of DSP mem and when effects are applied
completely on this scrath buffer, memcpy this to DSP mem
with this DSP mem is updated in one shot.
-Remove repetetive logs which clutter the logcat if
msgs are enabled in audioflinger.
Change-Id: I9051e7b8531aa5c8cb3dcfafe0be3136a2cf0f9d
CRs-Fixed: 463880
frameworks/av: Update framecount and buffersize values
-framecount should be calculated based on mMaxBufferSize
returned from HAL
-update the buffersize with the value returned from HAL
CRs-Fixed: 482744
Change-Id: I90dd9c3ebbbc8a9f1f2f92c5347ae9cb01719e13
audioflinger: Fix the LPA-AudioEffects dead lock issue.
- Issue:Deadlock occurs when the LPA clips are subjected to
rapid next from BT device and simultaneously on/off the
audio effects.
- Rootcause:some times flinger thread processing
LPAPlayer/directtrack next deadlocks with the thread
working on effect configuration as both of them
contend for the audioflinger::mlock and effectmodule::mlock.
- Fix1:AudioFlinger::deleteEffectSession() not to acquire
audioflinger:mLock instead take the mLPAEffectChain.mlock.
- Fix2:ThreadBase::effectConfigChanged() not to acquire
audioflinger::mlock.
Change-Id: I056c8297802f81644fa1371836db42bdbd3825fd
CRs-Fixed: 477511
libstagefright: Add support for High Frame Rate Encoding
- Based on kkeyhfr key value from meta data, add support in OMXCodec and
MPEG4Writer for HFR mode
- Assume normal mode recording if kKeyHfr is absent
- Increase bit rate for high frame rate (HFR) recording feature to reflect
the corresponding increase in frame rate
Change-Id: I0a69f8d9322a768677781d08dd910dc5772c5292
libstagefright: Support some userdefine properties
- support property to disable audio
- support property to change recorder profile mode
- support b frame encoding
Change-Id: I175decec83f6027cbd7988caf680f7fec2836f83
CRs-Fixed: 443327
libstagefright: Add support for H/W AAC decoder
- Currently, only software AAC decoding is supported.
- Add support for H/W AAC decoding by including it in the
list of available decoders and use it for decoding only
if the property 'media.aaccodectype' is set to 0.
Change-Id: I4bb9df1bd10bd8ee91e63dadd6c473fc4e29813a
CRs-Fixed: 449145
libstagefright: Move checks for creating new extractor to ExtendedExtractor
- Move all the checks and creation of the extended extractor
into ExtendedExtractor.
- Restrict creation of new extractor to the following conditions
o default extractor is NULL
o default extractor says the content is video only
or has an unrecognized audio stream
o the audio stream is a amr-wb (plus).
- This change is being added to avoid unnecessary creation of
two extractors thereby improving the startup latency.
CRs-Fixed: 462087
Change-Id: Ia87eca73c4f81d37697fa85fd4f7c8cc8d406104
[StageFright] Enable 4 channel support
This patches enables 4 channel WAV audio support and fixes invalid
data size in WAV header field if it exceeds the actual source size.
This patch is needed to support WebAudio in WebKit as some of the
chrome demos use 4 channel WAV audio and bogus header information.
Change-Id: I307026107ab4e4342b1c0d7bb64761a416fb2c65
audioflinger: Fix crash on LPA shutdown
* Decrement the refcount after unlocking the mutex
Change-Id: Ic3210700e0aaf5e8df78f85f501621a455058e24
libstagefright: Accept vendor specific NV12 colorformat from component
- Accept OMX_QCOM_COLOR_FormatYUV420PackedSemiPlanar32m color format
which is NV12 + 32 aligned stride and slice.
- This is different from vanilla NV12 which is 16 aligned.
Change-Id: I6de2ec3a78215dbcc28a6006b746e3e0afe69c3c
libstagefright: various fixes for avc_utils
- skip seq_scaling_matrix_present_flag assertion if checking for
interlaced property.
- correct interlace check to outside of if-block
Change-Id: Ia5854110feb1c56ddc86b312d2ba2dbb73d37804
CRs-Fixed: 445527, 445692
libstagefright: print stats at end of playback
- prints statistics before reset at the end of playback onto
logcat
- print statistics after each pause and seek
Change-Id: I68edcc3153a04209e7382e4d3fba0bf734f3e33f
CRs-Fixed: 457926, 447109
frameworks/base : Fix to play a specific Mp4 clip due to SYNCH_LOST_ERROR.
-Unable to play a Specific Mp4 clip.
-Mp3 playback is stopped if the Decoder errors out with SYNCH_LOST_ERROR.
-Ignore the frame with SYNCH_LOST_ERROR and play silence instead.
Change-Id: I6b94a83cf89e8bc6792d8ee3804042d629aa505b
Add checks before removing an active buffer in OMXNodeInstance
With this change, OMXNodeInstance will remove a buffer from it's
active list only if OMX_FreeBuffer returns successfully.
Change-Id: I685b39ac7ba762a2fc1b64d7f6c1efd391513598
libstagefright: Add interlaced video support
- Adds call to set output buffer size on the native window
Change-Id: If4a67b3f877bef557c46bb67b29d1e7051553335
audio: fix for AMRWB param overwritten issue
- Overwrite AMRWB params with default value only
when setParameters is not invoked
CRs-Fixed: 456459
Change-Id: I3fa6b56101ca408ed5b5b82707c6dc75a9d9f17b
audio: fix encoder parameters for AMRWB format
- AMRWB encoder only accepts SampleRate 16k and channel count 1.
Always overwrite AMRWB SampleRate and channel count to default values.
- AMRWB encoder accepts BitRate from 6.6k to 23.85k, only overwrite
AMRWB BitRate to default(23.85k)if setParameters() is not invoked
Change-Id: I75a96b54ef04bc59dab9074ec112071e62fd51aa
CRs-Fixed: 460931
stagefright: Add QCOM_BSP ifdefs for interlaced video handling
Change-Id: I856ae4a97f1bf13ab18d386b3486e742a4804b2a
Camera : Changes to support camcorder profiles.
Change-Id: I9c4bf14f273839fd36d5f52db0f215873e8291a0
av: Ifdef all the things!
Change-Id: If9dd6c6442e9d2ac9e55e48369f2da85f5f951f7
Camera: Add profiles for camcorder.
Change-Id: Icdaf1fae0018de1fb04f41125cfbe34a91b5eda7
libvideoeditor: use vWidth and vHeight for buffer allocation
- video editor detects crop information from decoder, crop
width and height will override metadata width and height.
- decoder is capable of sending crop information where
crop width and height are smaller than actual resolution.
- use actual metadata width and height for calculating
buffer size.
Change-Id: Id1d77c316e3892e6d51a00418052f256629f495f
CRs-Fixed: 452511
Add ifdefs around enhanced media types
Change-Id: I64b8853660ac4fe90ddb218b237f63b635cdb47b
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When audio sample rate which set to audio track is not the same with
the actual pcm data, noise will be heard. Fix the bug when write 8 bit
pcm samples.
AOSP commit: https://android-review.googlesource.com/#/c/59837/
Change-Id: Idcb0d7b0e9aaa250dd22b758c8337e23d1706049
Signed-off-by: Ming Zhou <b42586@freescale.com>
Signed-off-by: guoyin.chen <guoyin.chen@freescale.com>
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Needed for Samsung legacy camera libs.
Change-Id: If03d8525b55181ea20dc934dbcbfef85402c42c7
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Completely ignore directories with .noscanandnomtp files
in them. Placing a .nomedia file will still scan a
directory but exclude it from the media database. This
is so the file may still be presented for MTP purposes.
Placing .noscanandnomtp completely prevents the scan,
which saves considerable processing power and battery
life on systems with numerous media files, but prevents
them from being seen over MTP.
Change-Id: Ibff2a9f2525255a2ac34132eeee36734962fbdd7
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Change-Id: Idd5c7a0364710d54809ef5d4c7b2404b22dc4cf6
Conflicts:
include/media/IAudioFlinger.h
media/libmediaplayerservice/StagefrightRecorder.cpp
media/libstagefright/Android.mk
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- Handle new ADSP status parameter
- media/libmedia: Add new ADSP status audio parameter
- framework/av: Add handling of new key-pair value in
Audio Flinger
- Handle Tunnel mode SubSys Restart
- framework/av: Post SSR event to Audio Flinger
- media/libmedia: Post SSR event to AudioTrack
- media/libmediaplayerservice: Post SSR event to
MediaPlayerService
- media/libstagefright: Post SSR event to TunnelPlayer
Change-Id: I8c8385af45be91caf7d7160ab2e0236d6591b159
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https://android.googlesource.com/platform/frameworks/av into 1.1
Android 4.2.2 release 1
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Change-Id: Ibe42bfa73816bbfeb7e652d435254d0171b89727
related-to-bug: 7638150
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related-to-bug: 6870049
Squashed commit of the following:
commit eee2f3ba6bb7335f4e285632726db85645669929
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 15:02:01 2012 -0800
Make everything a lot less verbose by default.
Change-Id: I884d7a7901aa1e7d4ff590f065ca57a79d2af8b3
commit 6bbdb837ed5bd88008e45efb8faf595e4051ba26
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 14:34:46 2012 -0800
HLS now properly signals media time changes at discontinuities including
the start of playback (which may not necessarily be at time 0 if the playlist
is of type 'event' and hasn't completed yet).
Change-Id: I5ab747d024f9b8d0df72a4e06a12ebb29f62802e
commit 1555589832b1878a144a976a643e1af4d61f877c
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 14:32:28 2012 -0800
As part of a time discontinuity, clients of IStreamListener can now
signal the corresponding media time after the discontinuity, i.e. the first PTS
timestamp following the discontinuity will be considered equivalent to the
specified media time and media buffers timestamped accordingly.
Change-Id: Id7db7679b7faa6efd6270620ff52e34e884f3e92
commit 5c24c605c073a11c426d025b1e7478fc1ad8365a
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 13:00:56 2012 -0800
NuPlayer sources now expose flags() and can announce
that duration may change (increase) dynamically, in which case duration
will be polled at 1 second intervals and communicated to the upper layers.
Change-Id: I45102909b7a19eed0dda576747e3814d742a0eea
commit ecb71de8e281e61971a2cd73e7161a97540bc357
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 12:57:47 2012 -0800
Stop caching duration in MediaPlayer, duration could increase dynamically.
Change-Id: I7bb2f16c0abe49debdf45c776d2266aa069d7791
commit 544aec5823e6d7a3e97e15b6b23546616bcd343e
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 08:46:28 2012 -0800
An attempt to add support for "event" style HLS playlists.
Change-Id: I3dfb2e801ecaff8f5d8bdb3a4fca1b18aeeb2c60
Change-Id: I48cf7f65a654d33f2f49ded74f8be22aed9e3b98
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auido: Add amr-wb+ codec to ACodec.
-Add an entry for amr-wb+ decoder in ACodec.
-amr-wb+ non tunnel will be enabled by default.
Change-Id: Ied8902eb83da29a3164eb99e88630570a43f681e
libstagefright: Create MP3 decoder libraries without OMX layer
- With the current MP3 OMX SW decoders, the decoding time
is increased w.r.t the libraries without OMX layer that are
present in GB. This increase in decoding time results reduction in
power savings in LPA mode.
- This commit is to remove OMX layer for MP3 to reduce the
power consumption in LPA mode
Change-Id: I835ab6d013a326f111e513586f884bacd5f7106a
audioflinger: EffectModules are updated with device change
Issue: Effects modules are not updated with the device change
information
Fix: 1) Add setDevice information to mLPAEffectChain
2) Remove the return after sending the device route information to
Direct track so that mixer thread is also aware of the device
change for EffectsChain
Change-Id: I82936cd47290946a5e4772e448669d81e0e4d6f5
libmedia : Add a NULL pointer check
- Print frame count in AudioTrack::dump() only if the control
block is valid
Change-Id: Icf594eb721b48795c43d7bd165f6086031ce6efd
CRs-Fixed: 435050
libstagefright: Query AudioSystem for suggested record mute duration
- AudioSource mutes a pre-defined duration (defined by kAutoRampStartUs)
at the beginning of a recording.
- Instead, query the audio system for any ongoing playback streams
and use its output latency to calculate the duration to mute the incoming PCM stream.
- This assumes all current playback threads will be paused once recording
is started.
Change-Id: Ie9b1d62e7be803ef1d8a59127b95c73e03fa5ce6
CRs-Fixed: 438149
libstagefright: Convert mono to stereo for LPA clips
- Sound effects are not supported for mono clips
- Repetative calling of effects_configure and effect_process for
mono clips is resulting in crash in the sound effects library.
- So, Mono clips are now converted to stereo by copying the left
sample to right.
- This is same as what Resampler does in Non-LPA Playback.
This commit is a port of fcc0647fab20ceaf1c07bc10bb243f14c48b114c
CRs-Fixed: 421639
Change-Id: Ie579c8d11afe3db8d42a35956e8bf23eeb88cfe6
audioflinger: Fix to set volume from MediaPlayer in Tunnel mode
Issue: MediaPlayer.setVolume does not have effect on Playback
volume in TunnelPlayer mode
Fix: the left and right volume parameters of setVolume are
hardcoded and defaulted in DirectAudioTrack. Updating the
parameters from the input arguments fixes the issue
Change-Id: I8a107ce57284b225b17d95fed0f69e3adc5fb131
CRs-Fixed: 441849
libstagefright: Enable Tunnel Decode for select formats
- Enable tunnel mode decode only if the audio mime type
matches a supported list.
Change-Id: I32afd83e5fda1e90cb671dd747f17cb83bb84fc1
CRs-Fixed:437651
framework/av:: Add support to decode mp3 data in mp4 container
- Added support to decode mp3 data in mp4 container packed as mp4a
atom and .mp3 atom as well.
Port of 8fa3774adf9259b33ee721cfaeff26da42c29928
Change-Id: I1a04022f30a9f6516575440aba7652986ab7dc58
CRs-Fixed: 439897
audiomixer: Use High Quality resampler
Use very high quality resampler to upsample to 48KHz sample
rate.
Change-Id: I1ba5b839f1e74ae71b405538d970e6a966bd1d47
CRs-fixed: 416730
audioflinger: Fix a deadlock
- A deadlock will happen if the obit recipient
registered by the DirectAudioTrack is called.
- Fix this by moving the lock acquisition in DirectAudioTrack::clearPowerManager()
to after DirectAudioTrack::releaseWakeLock() is called.
- Also synchronize use of mPowerManager in the DirectAudioTrack
destructor with DirectAudioTrack::clearPowerManager()
Change-Id: Ib127db1406c4a61a4054ca0cf30f4c7347a5c92a
CRs-Fixed: 444093
libstagefright: TunnelPlayer: update condition to send SEEK_COMPLETE
- If the client tries to seek to 0 (e.g as a result of LOOPING)
without ever calling getPosition(), we will always sent an immediate
seek notification without seeking.
Change-Id: Id2b9d00c611278d0521cb6fd402710f0ec37bbdd
CRs-Fixed: 441411
libstagefright: Remove unnecessary code from TunnelPlayer
- TunnelPlayer tries to mimick AudioPlayer when trying
to delete the extractor source.
- It is needed for AudioPlayer as the OMXCodec object
is referenced by the CallbackDispatcher as well as AudioPlayer.
- This condition is not true for TunnelPlayer, so why do it.
Change-Id: I79c4e17d01910e73ad01c5640ef374626313a18e
CRs-Fixed: 442365
Add MediaDebug header from CAF
Change-Id: I68dbe72f86a49685b82b64927d1aa80231647a7a
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Change-Id: I96ec5b79c08e37c9bca59470addb5a9f7869eaea
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libstagefright: Add QC specific media format
- Add QC specific media extensions
- Add QC specific media definitions
Change-Id: I7dca90be3b977701d9537f5e017117790a030f1f
audio: Compile AudioParameter as shared library
- AudioParameter as shared lib is needed by BT
support in WFD source.
Change-Id: I464b428ace0cbb57ce6bf7bf3b57d51a7d56f032
libstagefright: Send flush on both i/p and o/p ports together
- ANR occurs in music due to race condition in OMX component if
flush is issued separately for i/p and o/p ports as DSP only
handles simultaneous flush on i/p and o/p ports.
Change-Id: I5b16cd5a9b57c857dc8bed489d2663b8f54769e3
libstagefright: Enable extended A\V format
- Add new files to support extended A\V format
Change-Id: I1e61d78d35b868d55fd8e99f95de8cab9c465db4
libstagefright: Framework to plug-in propritory parser
- Extend the current framework to plug-in propritory
parser
Change-Id: Ia586a3048420ddf1515261f20035589447263b7b
audio: add support for QCOM audio formats
- Add support for EVRC, QCELP, and WMA formats.
Change-Id: Iaf80f982fc8b08617132dbd7d524a1748866745c
frameworks/av: Support Tunnel Playback
- Implement DirectTrack and DirectTrackClient
- DirectTrack exposes API to client so it can create a direct
output.
- DirectTrackClient allows notifications to be sent to the
client from DirectTrack
- DirectTrack is being used for Tunnel Audio
Change-Id: I2fbb18a781d8e44b8d65da9a357f6e39375f063a
frameworks/av: Support LPA Playback
Add support to enable Playback in LPA mode
Change-Id: I1b8ac4904f4735017d62f3757ede7bbb56e62fd3
audio: Send correct channel mask in voice call recording.
-Using popCount function to get channel count gives incorrect value on
voice call recording.
-Only STEREO and MONO bits to be considered to count
channels on input
Change-Id: I04c2c802422e868bdba0538ff8623dbf9eb659fe
libstagefright: Thumbnail mode initial commit
- use sync frame decoding mode when kClientNeedsFrameBuffer
is set for hardware decoders
- hardware decoder will only expect I frames, OMXCodec will
set EOS on first ETB to stop more frames from being pulled
- skip EOS check on FTB so that the first frame will be
handled
Change-Id: I0e8974e088fdcc468e27764861c128cfe291499f
audio: Add support for QCOM's VOIP solution
Change-Id: I1150f536fa204b535ca4019fdaa84f33f4695d93
audio: define QCOM audio parameters
- Define QCOM audio paramters for FM, VOIP,
fluence, SSR, and A2DP
Change-Id: I29d02e37685846f6d4f00dee02e2726b015eaae7
Add ifdefs for QCOM enhanced features
Change-Id: Ic8e5fe6ecc058466ced71030883b1af6c2bc055c
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Initial members in ToneGenerator so destructor can free
allocation properly.
Change-Id: I4a686084af48f99d2eefc33e56e3239846d3d34f
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AudioSystem::getDeviceConnectionState" into mr1-staging
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With jb-4.2.1 the type audio_devices_t found in system/core/include/system/audio.h
was redefined from a typedef enum to uint32_t. This causes the signature
of AudioSystem::getDeviceConnectionState to change in libmedia.so.
Any older than 4.2.1 prebuilt audio.primary.___.so binaries (such as mine from ICS)
may refer to the old signature. This patch adds back in that reference.
Change-Id: Ie4f92eaec20d581c9bebc805cfd25f8558406e30
Signed-off-by: Hashcode <hashcode0f@gmail.com>
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Add customized parameters for qcom camera features.
Squashed commit of the following submissions to CAF:
camera: add custom camera parameters to hardware java file.
Change-Id: Ic2b363df6d25a0c3a49aed30e717d10874f533ae
Camera: Add support for wavelet denoise
Change-Id: I97be72615fe5909987ac430034bbdc16f2d20509
Camera : Add support for histogram in frameworks
Change-Id: Ibf751c3b2d488f4be0afaaa3e994372bafeffbe7
Camera: Add ZSL support in framework.
Change-Id: I4e863020d7d407d9c3936beb83dc7abf385c4654
Camera: Add touch AF/AEC support in ICS framework
In, ICS the old touch AF/AEC functions getTouchIndexAf(),
setTouchIndexAf() etc are no longer valid. Added new functions to get
the center of touched area.
Change-Id: I37d00040f895cd8a4de2e2a438971aa2d9c1a045
Camera: Add support for Red eye reduction
Change-Id: Iada280ac33aee8b1114dc40d8d79e0e47ad5c2fd
Conflicts:
api/current.txt
Camera: Enable streaming texture and YV12 feature
Change-Id: I814c92b469e0f25baef0ce80338761d1e6443085
Camera: Add frameworks support for high frame rate
Change-Id: I9722e9b7e61433651ab5f76f76208853eb1fd849
Camera: Simplify ASD menu and best mode menu.
Change-Id: Ia7a1eca9268eec021a6e2c3f7ce2a932e8819d50
Camera : Macro definition change. PIXEL_FORMAT_YV12 to "yuv420p"
Why: CTS testcase is expecting "yuv420p" as the format string for YV12 mode.
CRs-Fixed: 328252
(cherry picked from commit 0a263f67b75497e4b95b9de8c370ed9496ab59ef)
Change-Id: I66f418a52775a969d81bcb332bbe500370e3ab7c
Camera: Changes for face detection
Change-Id: I7b2172e8690d6e23762e42f5888fecbfda15ba61
Camera: Add support for burst count
Change-Id: I4488a5eac1eebb0b5c8e2cc7cbbe7be0c7301be7
Camera: enable HDR feature
(cherry picked from commit e40770bbdadd925fc1e07a08249a15981b961570)
Change-Id: Ie9ca3066e8e53710bf292d047a3724aefc029854
Camera : Add API to find fullsize liveshot support
Change-Id: I3c2fc3578decbe5cf87fded5362f317df0b40b5e
(cherry picked from commit 65d80f04892776eec1c19c844eacb8c454b4bd1a)
Camera: Set default camera area string
The default invalid value for KEY_FOCUS_AREA and
KEY_METERING_AREA is changed to "(0,0,0,0,0)"
CRs-fixed: 334934
Change-Id: I354802c8611ee5c00a591b96b1a1ca04926969cb
Camera: Change the scenemode names in framework
To comply with ICS API Specification,
SCENE_MODE_AUTO changed to SCENE_MODE_ASD
SCENE_MODE_OFF changed to SCENE_MODE_AUTO
Change-Id: I8e744f10c728f469923b09f69d6d64bc5c7c6850
Camera : Framework changes for camcorder UI menu
Change-Id: Iff9e9021da44b67f81a24c828b960113ee249248
Camera : Remove unwanted video resolution in Framework
- FWVGA, WQVGA resolution UI options are removed for camcorder.
Change-Id: If621140966b01dbc8c30a4d161e6771528b8adda
Camera : Enable FWVGA and WQVGA resolutions in camcorder media profiles
Change-Id: Iad9cc33f7b6d8eb7de21df372c016edeef0b99ba
Unlock camera AWB lock only when white-balance is changed.
bug:5633546
Change-Id: I90bb4b5b445daedf38a64e2f8fb980fc0f0aaa81
Change-Id: Ie3cc15132489502e64321776d49d96846b5705d6
Conflicts:
camera/Android.mk
camera/CameraParameters.cpp
include/camera/CameraParameters.h
services/camera/libcameraservice/CameraService.cpp
services/camera/libcameraservice/CameraService.h
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Change-Id: I5c1a3d1093cbf16111638a37601093c5eb977dd4
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Used by the Tegra ICS OMX libraries
Change-Id: I2f6e7f10f11b53853626d1be86e7b2be870720f4
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The new camcorder start sound is longer than previous one and we
must discard more audio when capture starts.
Ideally, camcorder should use synchronous record start.
Bug 7394330.
Change-Id: I219b4e231aba706776dc7ccc4f1c996eaf22f61a
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Add support for querying whether there is currently a recording
underway from the specified audio source.
Bug 7314859
Change-Id: I986b231a10ffd368b08ec2f9c7f348d28eaeb892
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Bug: 6635041
Change-Id: I3386a4a6c226bc4eceaf65556119e4fb15f73224
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Change-Id: Ia010e7a00534f9356b3247369d0ffd65591d91aa
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Change-Id: I866768b1e3f3b232f1934a35b65f66befc12f3f6
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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
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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
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expat header files from the approved location."
* commit 'e7c992c675e0075cd60b0b11c06d5956e3ddd7a3':
Fix libmedia and libstagefright to get expat header files from the approved location.
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files from the approved location."
* commit '5abc91d61718e334e05c6d4858bd868bd5a5769c':
Fix libmedia and libstagefright to get expat header files from the approved location.
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location.
Found during a libexpat upgrade.
Change-Id: I9e5abd502a884975d898486edde374fb2519164e
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AudioRecord::set() was calling getMinFrameCount() with
a channel count instead of a channel mask.
Change-Id: Iabace7686426430fd53deac0c71b0c36aa64171c
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Renamed disconnect() to dispose() to emphasize the fact that
this method is intended to clean up the IRemoteDisplay
completely, not just temporarily disconnect the current client
(which might be useful someday).
Other minor tweaks.
Change-Id: I1209639eb0cd8af09c724206642d7e52aab48257
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Change-Id: I582ed000026bba6d116db8304e15a3c52f8a9a01
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jb-mr1-dev
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Change-Id: I89e62d84328e12f08c8f73cbfd53cfb42569de97
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Change-Id: Ie7504d0ddb252f7e4d4f99ed0b44cfc7b1049816
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interface.
Change-Id: I7d3c44cb79cd40e73499f2d7ccf35c69b628e6d7
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jb-mr1-dev
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Change-Id: I08f17efa0c7d007e17408feb7d4fbef0a19f531a
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Change-Id: Ie0de1c086c629a668d1c7863992eef56fb94157c
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jb-mr1-dev
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This reverts commit 5d464eb0b8cffb994a754ff108795e858a882414.
It caused the wrong thread ID -1 to be passed to IAudioFlinger::createTrack().
Change-Id: Ic221d2bb4af572d3d2d752af19238c52f6728e3a
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Add the ability to dynamically register low level MediaPlayer
factories which will be probed at setDataSource time to determine the
proper MediaPlayerBase to instantiate.
This change is in preparation for moving libaah_rtp out of
frameworks/base and into phantasm platform directory.
Change-Id: Icf8904db3ab9e3c85df6e780d5546d9988cb9076
Signed-off-by: John Grossman <johngro@google.com>
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Simplifies the error recovery in case IAudioTrack fails.
Change-Id: I6aee41a2ac747a5689fb4836b04174e6107bf32f
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Document where int is used instead of AudioSystem::sync_event_t
(probably because of a header file dependency).
TrackBase::start() and RecordTrack::start() don't need default parameters.
Change-Id: I82f4a4d078be900f3aa4bd926697e32f5ed68ec8
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Change-Id: Iad008f20d35a18acf500f773900164552fd0c19e
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Fix race conditions for EVENT_MARKER and EVENT_NEW_POS callbacks.
Marker and new position update fields are protected by lock.
getSampleRate() doesn't need a lock because it reads from shared memory
control block.
Enforce that the parameter passed with EVENT_MARKER and EVENT_NEW_POS
cannot not be changed by the callback handler, and will not change during
the call by another thread.
Session ID should never change; log if it does.
Change-Id: Ia2c63cf1a71b10bb06c37981bd76437f83fffa91
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for POD structs effect_descriptor_t and effect_config_t
Change-Id: Ib2fc47f85fb65ed91b0abb1f87217c49b5eb571d
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Change-Id: I1c1896da48983aa9f1462a4b471f910498816f60
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like AudioTrack::stop()
Change-Id: Iab62f4665151345f1ad5874c97a21d1a331f0154
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