| Commit message (Collapse) | Author | Age | Files | Lines |
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Change-Id: I5d6dedbb9ba03fc313ab27f91bf83376fa20a7a9
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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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Change-Id: I6a7a91d930f7789ca78370f0c0e0e306dad87028
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- Metada mode video recording is enabled by default.
- use setprop debug.camcorder.disablemeta 1 to disable metadata mode recording.
Change-Id: I422c49c0ace0c3a3e1f4459c7e4bf29e70af763a
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libstagefright: Return seek position until seek has been processed
If it so happens that the client to TunnelPlayer (e.g. AwesomePlayer)
queries for the current time before data from the new seek position
is given to the compressed driver, we need to return the seek position
Change-Id: If709e61f67cc8e81d34c14d19145dc61ecd82c2b
CRs-Fixed: 454825
libstagefright: Use 64 bit offsets only when needed.
For enabling >2GB recording, 64 bit offsets are
needed for file writing. So, this feature was turned
on by default. This in turn increased the file size.
With this change, by default this feature will be
off and turned on only when required.
- Use 64 bit offsets for resolutions >= 720p.
- Limit maximum file size for recording to 4GB.
- Set max file size only if no value is set from the client.
- Fix MPEG4Extractor to use 64 bit offsets
CRs-Fixed: 273144, 285785, 288319
(cherry picked from commit 04476a3fb89dfbb025f7852dd4d62cae72385f1a)
Change-Id: I00af2c7cddbbf86c566fe4bb989fe728ca06dd19
libstagefright: TunnelPlayer sync fix
- Allow close on the AudioSink to be called from
the extractor thread and the application thread.
- This fixes a race condition where an onPauseTimeout
event scheduled from the main thread closes the
audio sink while the extractor thread was about
to issue write() on audio HAL. (note: on HAL, not audio sink)
Change-Id: I22a5c655dfcb40f3cbda3765dc23ad8e6f99c9bb
CRs-Fixed: 443205
Frameworks/av: Fix to prevent deadlock in AudioEffects
-Write is blocked waiting for effect chain lock and this causes
decoder thread to wait indefintely.
-Sometimes it is observed that effectschain is locked before
mLPAEffectChain is initialized and but unlocking is skipped if
mLPAEffectChain is initialized in between.Due to this LPA
silence and framework reboot issues are observed as
applyEffectsOn() cannot acquire lock to progress further.
-Use flag to check if all effects have been locked and unlock
accordingly to prevent the deadlock scenario.
(cherry picked from commit 011db22abf565dfbe7f9d0a5c7af7564587b3b48)
Change-Id: I82cfdab045ecf077f0ba0185fc693fc623fa10db
CRs-Fixed: 435661, 435664, 435680, 430309
audio: Use tunnel player only for music stream
- Check stream type before creating tunnel player to
use tunnel player only for STREAM_MUSIC
Change-Id: I6e4b58524e61441ad2e09499bd9187c6dd56cd3d
framework/av: Fix for audio recording test through CTS
- Issue: Failure in stop is observed with the audio recording test
through CTS.
TestScenario: When the audio record test is initiated in the CTS
console, the recording session is force closed with a notification
File Size limit exceeded. Further, the stop fails with the same
message(notification of the File size exceeded error).
- Cause: The calculation of nTotalBytesEstimate for the recording
session exceeds the limit 95 percent of mMaxFileSizeLimitBytes.
As a result of size deficit, the recording is stopped at the
beginning of the recording session notifying
MEDIA_RECORDER_INFO_MAX_FILESIZE_REACHED.
- Fix: The factor size used in the calculation of nTotalBytesEstimate
has been updated properly for 64bit file offset setting. The
setParam64BitFileOffset in StagefrightRecorder::prepare() is executed
based on two additional validations so that the factor size is updated
appropriately.
Change-Id: I4749ce8f9735ccc9e1d9e49718c36470837ab27f
CRs-Fixed: 396057
audioflinger: apply volume on direct track when track is active
During back to back tunnel playback, we encounter a race condition
where setVolume can be called when the track is not updated to
active state. Fix to apply the volume on direct track only when the
track is in active state.
Change-Id: I70c289fbf8a9266bae0bd01b04be9f43ad32c70d
CRs-Fixed: 464148
LPAPlayer: Update condition to ignore seek
- Reject seek if the new seek time is greater than the current
position and within an empirical limit (default 60ms).
- This limit must be measured for each target.
Change-Id: I86b44679fb5fe442bb5adb510c62514f6be3d304
CRs-Fixed: 453067
audioflinger: for DirectAudioTrak, call startOutput before stream is active
For LPA and Tunnel playback, when resume is done in paused state, before
starting actual playback, volume should be set through AudioPolicy command
thread.
Change-Id: I7ee1098058c01a35a3e7181d3b291304abf3cac1
CRs-Fixed: 464348
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Change-Id: I9e5c22af1a5cf916b0efaec7ca1c5f48f6d0c82a
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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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- Add new player factory to support dash playback.
- DASH urls end with .mpd. When media player receives
an url with .mpd, it will use new factory to instantiate the player
to be used for supporting DASH playback.
Change-Id: I69e5a08fb2baf89d97b1e0711dbe52a8b1c39c29
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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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libstagefright: Add support for VC1 clips
- configure decoder in frame-by-frame or arbitrary mode
based on the codec type.
Change-Id: I6404e5b7ee217045e6456f51f914dbd8a651d98a
CRs-Fixed: 432847
stagefright: Miscellaneous fixes for LPA and Tunnel playback
1. Fix for AV sync issue with Tunnel playback
APIs for returning correct timestamps were implemented
2. Crash while exiting TunnelPlayer
Check whether sink is open before flushing or closing it.
Check for mIsAudioRouted is good to know if we are closing it
3. Seekbar freezes after seek and pause
Check pause status before writing
Change-Id: Id8ab7b258e9c05b20e121bdf3c4dc30d519f6c15
frameworks/av: Add support for surround sound recording
- Add 5.1 channel as supported input channel
Change-Id: I50fcd87245c5c855ede8f09ea1a7c5be2e684640
stagefright: Add tunnel decode for amr-wb/wb+
When tunnel.decode is true, playback of AMR-WB
and AMR-WB+ uses tunnel decode.
Change-Id: Ic06cb3faaac18f605402c98a27cb4dfa4b7faee2
stagefright: Skip LPA if #channels > 2
Skip LPA for Multi channel playback
Change-Id: I5b97471ef55aea260d1c02db672eb4bd64c3edef
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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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o related-to-bug: 6971073
Change-Id: Ia6d926663231f9a9ef31c82c85c70595c4a30ebe
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Change-Id: Ie54ee3edd672ec629360b4ecc5df2f85ecbaa45f
related-to-bug: 7247584
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Change-Id: Ia010e7a00534f9356b3247369d0ffd65591d91aa
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Change-Id: I866768b1e3f3b232f1934a35b65f66befc12f3f6
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We only checked the mNativeWindow against NULL, but did not check the native window object itself against NULL.
o related-to-bug: 7153683
Change-Id: I430d344dc09a2fc337bf51001551627056713773
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Using a MediaPuller now, audio and video on their separate threads.
No more flushing the RTP stream on audio tracks, since it wastes too much
of a full size UDP packet.
Change-Id: I53346b4aea739c3142da13bd179428503a3c98b0
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Change-Id: I321b6c1a0223e0c3d5df04fde7a13ab91c08c7bc
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Change-Id: Ie07eca6b45142bdd83412ee0e38d732a4c355630
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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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Bug: 7082658
Change-Id: I21269d72d1d21627c6adb2e6fbcb078d0eda6020
Tested: checked MediaScanner scans .mxmf and Music app plays .mxmf file.
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Still experimental. Set property "media.stagefright.use-fragmp4" to true to
enable.
Change-Id: I210b9c5b5164b5c5eefc31309845ee881ac7db8e
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Change-Id: I7602cda0e38073c9f0e63fa4de238ca0d4b4ce8e
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Change-Id: I582ed000026bba6d116db8304e15a3c52f8a9a01
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o related-to-bug: 7089195
Change-Id: Ic30d9312673f2d5837c779e023ac64468ecd4951
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Change-Id: I0beac87025b93c60164daa865c89f16b72197a47
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Verifies the continuity counter
Parses and associates PCR time with streams
Allows for a absolute time anchor to be signalled via discontinuity.
Change-Id: I4bc88c78382c9cc6380f28df584cc6c254e0a8f9
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and rename it from Parser to FragmentedMP4Parser
Change-Id: I986f50d0c5c93648aac675d6160e18623b031541
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Change-Id: I409d7133a53a71e62523b1acc2b03302fcf824a5
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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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method
o The mClient could be disconnected before the notify message is sent out. Use
a local reference could help resolve this issue.
o related-to-bug: 6782035
Change-Id: Ia2047bd7f2666100fed83c6e5417a3bf2e20cd2c
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Don't crash if its time to start the next media player in a media
player chain, but the current player's client has a NULL mAudioOutput,
as is the case when the low level player implements
MediaPlayerHWInterface.
Change-Id: If555da3a46960cbee91ff26e5b1fd05e52c8ef28
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container.
Change-Id: I2f0f92dabe31341a6a91d435943e4a65e34884ba
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* changes:
Move the AAH RTP code out of framework and into vendor.
Add the ability to dynamicaly register MediaPlayer types.
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Relocate the AAH RTP code from framework/av into
vendor/google_devices/phantasm. This change is the deletion, there
will be a separate CL which re-introduces on the vendor side of
things.
Change-Id: Ibe7e6d4b633a3886b87a615691a2692f2382af6c
Signed-off-by: John Grossman <johngro@google.com>
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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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Change-Id: I16696740d3551aa8a5ba904ec757c64b0ecd1307
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Change-Id: I5628d3437b5e6f8836d78557fd07ab87e5a914e3
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Change-Id: I867e2499fac3045075654aaaecf44327753327fd
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