| Commit message (Collapse) | Author | Age | Files | Lines |
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portFormat.nIndex is being incremented which is not
trustworthy since the nIndex value could be overriden
by the OMX Component, which causes an indefinate loop
which inturn causes a memory leak and crashes the system.
OMX Component on encore and p970 exhibits this behaviour
(OMX.TI.720P.Decoder). This patch prevents stagefright
freezes when QueryCodec is called during Gallery Thumbnail
generation for videos and Adobe Flash playback.
Change-Id: I825c99ddecacbb927e22ac7d1a53facb26d95ff2
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This reverts commit 9a814ad626233ff02dd2d393929f32225bc94b68.
This is wrong. kPortIndexInput is defined as 0, the original value was correct.
Additionally, it breaks android.media.cts.MediaCodecListTest
Change-Id: Ib273cde69a4c622daf239bab5d12c5e7d568af2f
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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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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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This reverts commit ae57fbc021cfc8b018cfb23b90112b1b17173d1b.
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-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
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Use BOARD_HTC_3D_SUPPORT to enable.
Change-Id: I28fa3f1586071bcc78b8e887bbbf699d338a0ceb
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Needed for Samsung legacy camera libs.
Change-Id: If03d8525b55181ea20dc934dbcbfef85402c42c7
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Support is already there, but is not in the codec quirk reading list.
Re-implement it as required by Broadcom's OMX
Change-Id: I1beac06af8118dcf0c248b631bc8e6dbbab2c1d5
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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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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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Change-Id: Id6319774806e152333d468b9ff62d148e476555a
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The query index is wrong, it will make a death loop in
my ME722 when get resource thumbnail for MPEG4 video.
Change-Id: I64532156e762b847a8eae59560fb828549c29519
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libstagefright: Exceptions in using Tunnel mode decode
- Accumulate all known exceptions to a separate function
Change-Id: I61bbc288c9a087559db210e76141b8c57e67fff0
CRs-Fixed: 432080
libstagefright : Stability fixes for Tunnel Player (part 2)
- Synchronize b/w reset() and onPauseTimeout
- Synchronize b/w seekTo() and onPauseTimeout
Change-Id: Ia5cfc6b4dcc326ead440fba35d809d4f3f1b5a81
CRs-Fixed: 449122
Revert "Revert "libstagefright: Convert mono to stereo for LPA clips""
This reverts commit 0db8a19fb3216a8a83d5d6cbd5f1ccbf997a20d8.
libstagefright: Port Tunnel mode fixes to LPA
- Miscellaneous fixes for seek, pause/resume, EOS handling
- Miscellaneous fixes for synchronization between the decoder thread,
TimedEventQueue and the player thread.
- This change is a port of a similar set of changes made for
TunnelPlayer
Change-Id: I82c2904f7aedfb9c4f03200419fcba8b038e3d54
libstagefright: Avoid use of extra bytes to signal seek processed
- A few bytes were reserved in the buffer sent by Tunnel/LPA
player to audio HAL to indicate a seek has been processed and
there is no need to skip it.
- We won't need this method anymore as this can be fixed instead
by synchronizing seekTo() and the extractor/decoder threads.
Change-Id: Ic02ae1699bb59e2f6b8d9fb599d0fa43fd3f19e3
libstagefright: LPAPlayer synchronization fixes
- synchronize b/w seekTo() and onPauseTimeout()
- synchronize b/w reset() and onPauseTimeout()
Change-Id: I29a4ccf02e28fe7b7c00e35a679ff2b5271ffb6f
libstagefright: TunnelPlayer performance tweaks
Some tweaks when TunnelPlayer is used for audio/video playback
- Keep the extractor thread at ANDROID_PRIORITY_NORMAL
- sched_yield() after reading a frame to give the video thread(s)
(CallbackDispatcher and/or TimedEventQueue) to be scheduled
Change-Id: If0d86d629fd0e15aff917af8589472578cd28bf4
CRs-Fixed: 444041
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- duration = 0 can cause divide by zero and for this clip duration
is indiacted as 0.
- check for duration > 0 rather than duration >= 0
Change-Id: I58ccacbf7ede892dff9626715162ea7b1f2ddbc6
CRs-Fixed: 451855
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- Set decoder in frame-by-frame mode always, except for interlaced
content, for which arbitary mode should be set
Change-Id: I8195a40549898b43a0e03d65663c7148f458c448
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- On port settings changed first flush output port
- Move ACodec to new state called FlushingOutputState
- Flush all output buffers, wait till all decoded buffers
are rendered
- Then disable output port, and allocate output buffers
with new resolution, and reset native window
Change-Id: Iafa266371ed2a87b909fbcb4eeae6b64208df617
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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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- Make necessary adjustments to align the buffer size
for TunnelPlayer to satisfy the following conditions
- Buffer size is a multiple of LCM(1,2,4,6,8)
- Buffer size is aligned to 4k.
CRs-Fixed: 447274, 442365
Change-Id: I1b7f9ac3cf8ff86f972a8b6798bfcff3a4ba7c64
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Change-Id: I9e0fefa3f2dabe991c6be63ab13a18ca38c37f71
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https://android.googlesource.com/platform/frameworks/av into 1.1
Android 4.2.2 release 1
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For an MP3 source, within the prepare command, ID3 tags are checked in search of
gapless playback info. This causes problems for streamed sources. If ID3v2 tags
aren't present, then a check is done for ID3v1 tags. This results in a read
command that asks the cache to jump to the end of the file, and subsequently
make an extra http call to request those bytes. For a streamed source, this
causes the file to not be downloaded automatically when MediaPlayer::prepare()
is called, and causes stuttering and extra buffering time to be needed when
start() is finally called.
The solution is to ignore the ID3v1 tags as the gapless info would never exist
there, and only check for ID3v2 tags.
Cherrypicked from external contribution ffd6ffc5429c45577fd8e9f8fa90e79bb91b8a84
b/7638165
Change-Id: I7d1b94cffbfe7c38ca094834dedbc92a58855e20
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jb-mr1.1-dev
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Change-Id: Ibe42bfa73816bbfeb7e652d435254d0171b89727
related-to-bug: 7638150
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in our upcoming wfd _sink_ implementation.
Change-Id: I4509c30d5a7b992bc841b73d63db902bbcf8f76a
related-to-bug: 7638155
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for WiFi display" into jb-mr1.1-dev
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display
The time interval between periodic neighboring IDR frames is increased from 1 second to 15 seconds.
o related-to-bug: 7524791
Change-Id: Ic32f37448f952f329549eda5e73637ee3b02f046
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o related-to-bug: 7524791
Change-Id: I95ac4ee925e2dbeb00b3cfb2e29c611698c5cc9f
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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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jb-mr1.1-dev
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previously any error signaled by setupXXX inside ACodec::configureCodec
would be overwritten with the result of setMinBufferSize at the end
of the function.
Change-Id: Id4beb571ca52ea4646239d0af006e09ce1130268
related-to-bug: 7542181
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- manually prepend SPS/PPS if encoder doesn't support it
- latency improvements
- support for "our" method of optional RTP retransmission
- improvements to the wfd commandline tool for testing
- make it easier to turn on/off suspension of the video pipeline on idle
- fixes an issue where an error during encryption would cause a SEGV
- add HDCP descriptor if necessary
Squashed commit of the following:
commit 1115be0ebb3b885b4f1b7dba56761ca013d0ec4a
Author: Andreas Huber <andih@google.com>
Date: Fri Nov 9 11:32:23 2012 -0800
Better shutdown of wfd -l sessions.
Change-Id: Id898a14ae21efd3b065b00a729830063d39195a7
commit 0e7d106dfe4eb6e2640b0b66c65deaba265f7ff0
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 8 16:38:55 2012 -0800
No more sending delay, create rtp packets upfront.
Change-Id: I809a225f664fdb485c7d9a49a27886601a6a26b2
commit d399e8571b77353d59afb57508dfd2a82c1ef93a
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 8 14:19:43 2012 -0800
Restore AudioSource buffer size, factor out TimeSeries, make
suspending video optional.
Change-Id: Ifdfe4d447b901e714abf52856b4641d1d55a5d41
commit f8b649f0b8f917d59f4b8a2e8e6d7db61a684a78
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 8 09:34:06 2012 -0800
Pull 480 frames at a time from AudioSource/AudioRecord
Change-Id: I1e215abd329faec3da026631122c0f4c800c0ac4
commit 1bc13452eb35eebbba00f5da93fa86535be5db59
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 8 08:50:30 2012 -0800
fixed bitrate traffic simulation
Change-Id: Ic5efb7cbb0b5d3b4917bc77b8ba73d447249e695
commit 016cdff18e74bdc631a5679e97192645ed095aa2
Author: Andreas Huber <andih@google.com>
Date: Wed Nov 7 14:00:03 2012 -0800
resurrected "our" style of retransmission.
Change-Id: I34d757aba67428437cb39b8293a9651750ad20d9
commit 384cf1a3c8fb4ec410bdf8fa5722c298e6028f3e
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 6 09:38:55 2012 -0800
Changes to make wfd work on manta.
Change-Id: I7a4e00cf16581fe2146edd1b359af195774090e4
commit 9628f24b22b35f28630d99dda3614babf51bc07e
Author: Andreas Huber <andih@google.com>
Date: Wed Nov 7 09:15:44 2012 -0800
Patch up rtp timestamps to more accurately measure network jitter.
Change-Id: I9502a4615575f97f98a215a13131a89a6ae93c6d
commit 7c891a1a24f08bbd50f55be13f7d05f43e807eb8
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 6 09:37:24 2012 -0800
Additions to the "wfd" tool to create a local wfd source.
Change-Id: I99558653a70fdc703f9d13990b3ce1c4d3ae227a
Change-Id: Ia94c63fc390f597014531073485f0cfc53b3418a
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This happens occasionally when taking a bugreport.
Bug: 6447319
Change-Id: Ia6531a4a3658461f8fd3f7106e7996da7cc5933a
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* Was breaking on 8660 due to previous commit.
Change-Id: Ia9f5c45552cc933db336a66b6d1214b65e810488
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Change-Id: Ic73cb2f4152ac77a83fcda7e89b4a8202df440c6
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Change-Id: Ib6cc3401c889a8ec52dc83e3bea367f94a582a40
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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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Creating interface and framework for using FM Radio
RX and TX from different vendors.
Change-Id: I1a71aed01bfffdddfabf1cdfbfa3707cb1ed016b
Signed-off-by: Benn Porscke <benn.porscke@stericsson.com>
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This patch enables the TI ducati H264 encoder profile
via BoardConfig setting:
BOARD_USE_TI_DUCATI_H264_PROFILE := true
Allows correct video decoding on Motorola OMAP4 / Kindle Fires
and other devices using newer DOMX libs.
Effectively a cherry-pick of omapzoom commit (with creative
board setting name):
http://www.omapzoom.org/?p=platform/frameworks/av.git;a=commit;h=e28784d5c68c8699cfd9ebe0231e7132d8b13dad
Change-Id: Idc49b00030558a22a9e50e8798e5814ad54fe841
Signed-off-by: Hashcode <hashcode0f@gmail.com>
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DOMX default
Part 3 of 3 patches: To allow omap4 devices to use custom "domx" source
via a new BoardConfig.mk item:
TI_CUSTOM_DOMX_PATH := device/<manufacturer>/<device-name>/domx
This setting provides for 3 changes during the build:
1. In hardware/ti/omap4xxx this settings stops standard Google domx
source from being built and changes the domx reference for tiutils.
2. In frameworks/base it changes the default openmax references for
frameworks/base/media/jni/mediaeditor/Android.mk to the new location
3. In frameworks/av changes the openmax references in 5 places, and adds
new includes in ACodec.cpp, CameraSource.cpp and OMXCodec.cpp
This is a combination of cherry-picks from omapzoom (with a more descriptive
BoardConfig setting name):
http://www.omapzoom.org/?p=platform/frameworks/av.git;a=commit;h=8044105ca117c2e99b35ad9f341d630fc5a9d2e0
http://www.omapzoom.org/?p=platform/frameworks/av.git;a=commit;h=4adf712d1f3f2050fe0010652bbba7ecb8468870
http://www.omapzoom.org/?p=platform/frameworks/av.git;a=commit;h=fa37231ca59872ac491461ca3c14e019834848e5
Change-Id: I53dbf120d515eaf5ec82688dcea4c670c173ed01
Signed-off-by: Hashcode <hashcode0f@gmail.com>
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-Handling of EOS, and triggering EOS was wrong
in TunnelPlayer. Seeking when EOS was posted
to the HAL was wrong. EOS should
not be posted till seek is complete
-Also, EOS should not be posted to
the app if we are seeking
-Player should wake up when seeked,
even after EOS was posted from player
to HAL
-Fixed this issue by cleaning up the code for EOS
-Disable tunnel mode playback for streaming use cases
to avoid jittery playback
Change-Id: I21699d2d5874bde6cbfe549ce0251b252e9a4090
CRs-Fixed: 433346
CRs-Fixed: 432233
CRs-Fixed: 429868
libstagefright: Add new mime for QC TS container
- Add new mime type for TS container that is sniffed by extended
extractor. This is needed for media extractor to determine which
parser to create.
Change-Id: I18dcebbbf3b31cea7db29a4dd65385638343bec1
libstagefright: Use software decoder for ADTS content.
Use software decoder for widevine content which uses ADTS
format.
CRs-fixed: 431096
(cherry picked from commit 3edf2e703bcdc47f122864865056d5cb65b7ab43)
Change-Id: I50eba673ddd6ec2bbb737577978e61902ff68d13
audioflinger: Fix to release wakelock after closeoutput
-In DirectTrack destructor, closeOutput is called after
releaseWakelock is done. This may sometimes result in
power collapse happening even before actual close
sequence of Audio path is completed and will result
in high power consumption.
-Release wakelock only after closeOutput is done
in directtrack destructor.
CRs-fixed: 438179
Change-Id: Ibe103804daf2cb09bade998d6d34c3a34508dd09
libstagefright: Add support to change clip duration to enable LPA
Added support to change the clip duration threshold value for LPA
playback. A new system property 'lpa.min_duration' has been added
which controls the minimum clip length for enabling LPA.
The default threshold value has been retained as 60sec.
Change-Id: I6a8be6d6bf67495977d8c75e5be14723a31353b1
frameworks/av: Skip tunnel mode for playback through AudioCache
In the use case of playback using SoundPool, decoded data is
cached from player and further rendered through AudioCache.
Avoid Tunnel mode for the use case AAC format through SoundPool
Change-Id: I21005a5b39f9fb480ae0d525ecb560fec4382620
CRs-Fixed: 437539
frameworks/base: dumpsys rendering statistics for Stagefright
- this adds extra fps statisticis
- report via dumpsys
Change-Id: I7b4d4582c4eb2ccf2d11557844dade92f9e587c0
CRs-Fixed: 435013
libstagefright: Stop extractor source after start in TunnelPlayer
Issue: In the use case of frequent suspend resume during Video
Playback with HDMI Connected, we encounter a scenario where tunnel
player is created and destroyed without the extractor source
started. In such use case, stopping the source in reset during the
tunnel player destruction leads to failure during assertion check.
Fix: Check for mStarted flag to ensure that stop on source is
called only after they are started.
Change-Id: Ib18e7ee3d10b2cc706944e358046f163d156706c
CRs-Fixed: 440239
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-AAC and MP3 clips are not playing.
-AAC and MP3 are not using LPA path.
-Enable LPA decoder path and implement LPAPlayer class.
Change-Id: I76438319fc72c4898fad5910f8de874f89287687
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Change-Id: I96ec5b79c08e37c9bca59470addb5a9f7869eaea
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