| 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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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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Use -16dB as the default DRC reference level when decoding streams
with DRC metadata.
Bug 7370764
Change-Id: I900cee22f32384a5657fb041b69d42657bcddf09
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Configure the AAC decoder to use the modile DRC settings when
decoding streams with DRC metadata.
Settings are:
- target reference level of -12dB
- DRC compression factor (attenuation) at max
These settings can be overridden by platform properties if the
default values are to be modified.
Change-Id: If329fe566497685b735abe4e4b9a5e01697c063b
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output frame"
* commit 'a923dd4e6c14f5cba2fc5f6f152619ec6bc70c5d':
avcenc: Set OMX_BUFFERFLAG_EOS on the last output frame
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Without this, the caller might hang waiting for more output buffers.
Change-Id: Ief6be00b7600be8498a953e8e88f0582ea8fce85
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* commit 'f755a506335df3bfb32c062096deceaf045044ed':
aacenc: Mark some global arrays const
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This gets rid of initialized writeable data in the library.
Change-Id: I7c1cbeb3df10c302b645b7311591ead87a289ca1
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Decoding of HE-AAC fails due to output buffer size being too small.
Size output buffer size according to maximum number of channels.
Change-Id: I6b10842cf7ba3da1f4b4075b1f7f5404bf76be41
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before reading info from it"
* commit 'bf30787f2694057e775f3b527b649a9f8074237b':
avcenc: Don't erase the InputBufferInfo before reading info from it
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Previously, the timestamp and flags were read from the InputBufferInfo
struct after it had been erased.
This fixes cases where the output timestamp from the avc encoder
could spuriously be garbage every now and then.
Change-Id: I21acf23cfac93bcc201a603feace6499ec6d8e76
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assembly"
* commit '8d4d590dcbbd2b178976d0bd024625e9d6216365':
avcenc: Update a comment about the inline assembly
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Since the inline assembly was fixed in commit 3fdb405597f the
comment was no longer accurate.
Change-Id: Id4f8a111d1fa6e0c2e97f7db64262e8d1604c519
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constraints"
* commit '41ee747fc019eb5856454c698adf23f55073c69e':
avcenc: Fix the gcc inline assembly constraints
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Use +r for registers whose value is used as input to the block.
Also switch input registers to output registers with +r, for
registers that are modified by the assembly block (SUB_SAD in
sad_inline.h).
Without this, the inline assembly fails on modern gcc versions
(tested on gcc 4.6), where the intended input value never is
loaded into the registers, causing segfaults at runtime.
Currently this assembly isn't used, but this change makes it
work properly if enabled.
Change-Id: I37c247cd5c8aff213445e31ab3bd33d8f0f37e05
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* commit '894a7e450704b0c7858a2187a5432b479ac69c1d':
avcenc: Properly indent assembly blocks
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Also line break multiline assembly blocks - previously they were
virtually unreadable.
Change-Id: Icb269909b78746e26b28ab7dcb6979c4655a0b0c
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h264"
* commit 'add7c4d329586119fb8b5501b049905004400233':
hardening: eliminate text relocations from h264
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Change-Id: I8bdf9746f28f2f7bed6c1109c0f4badbab765a7c
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* commit '3b939a5e893ccfe87f5b7b3fe229439a9b2fc87a':
avcenc: Remove a leftover define
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This should have been removed as part of f5af6314db, but
the second __arm__ define was introduced after the original commit
was written, and not removing the second one was an oversight in
rebasing the patch on top of the new code.
Change-Id: Iccf08ffdef130446dd6e97dc1e15e6a1d4f9386f
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source files"
* commit '523c09d56c78933eea7fb4f028e0bfa595ca82b1':
stagefright amrnb: Remove unused leftover source files
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This allows getting rid of adding the plain common directory
(not common/include) to the include paths in projects that use
this code, too.
Change-Id: Ifb1fe4fc1f5c6f1264cb2b64f7b78e42b1e7833a
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bitstream_format enum into frame_type_3gpp.h"
* commit 'f20e4ad9f23fe0267f8f3434a6eb961eb69bde3a':
stagefright amrnb: Move the bitstream_format enum into frame_type_3gpp.h
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Change-Id: I6d6f9dc5799750394a853325466ffa2ebec44625
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* commit '0df213babc4fb41b7daa05cb3c08071f2b60f6a5':
hardening: eliminate text relocations from mp3
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Change-Id: I671ed4c8c7bb15172a8b64d513e2844319495102
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inline assembly"
* commit 'a109e9d5d5dbdbbd63a3f32fce131c718f0f8c99':
avcenc: Clarify the ifdefs surrounding inline assembly
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Previously, the code was structured like this:
#if defined(__GNUC__) && defined(__arm__)
/* Generic C implementation */
#elif defined(__CC_ARM)
/* RVCT version */
#elif defined(__GNUC__) && defined(__arm__)
/* GCC arm assembly version */
#endif
This had two implications - the gcc arm version never was used
in practice, and the code required -D__arm__ to build at all for
other architectures.
The inline assembly constraints are buggy (requires changes so as
to not crash), and if fixed, the optimizations still aren't faster
than the generic C code on modern gcc versions. Therefore, just
ignore them and use the generic C code.
Due to the peculiar code block ordering and preprocessor conditions,
the generic C code is what has been used before anyway, but it
did require -D__arm__ to build, which can cause problems if building
for other architectures, if the system headers have ifdefs for this
define. This change makes it build without that define.
Change-Id: Ib3fcf5651ee720310b7dff10fd04279b5f90759c
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amrwb"
* commit 'fba9d50d1b2e487bc08b59ec93d7ed3183c76dbe':
hardening: eliminate text relocations from amrwb
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Change-Id: I47235fc4afe6b916da1e999401fb3c8b12c94e62
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LINUX define"
* commit '1191287bc438c164a89cf86704936aeddbefb136':
aacenc/amrwbenc: Remove the now unused LINUX define
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This is no longer used by the codec code (except for the
amrwbenc sample).
Change-Id: I78c7f62f257c490bf44477217975a93abcc7779e
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typedefs"
* commit 'eeb9fde0cb8e0ace44eb720bcac272ee70282d4d':
aacenc/amrwbenc: Remove unused and dubious typedefs
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These typedefs aren't used anywhere, and their description
doesn't really match their actual definition. The VO_TCHAR
type is even defined as pointer for one configuration and
as non-pointer for another configuration.
This gets rid of the last use of the LINUX define in the
VO codecs (except for the amrwbenc sample application).
Within the codecs, the use of this define is unclear - does it
indicate code specific to android but not normal linux, linux
but not other unixes, unix but not windows, or gcc but not
other compilers, etc.
Originally, prior to cleanup, the define has meant many
different of these - these uses has now been cleaned up to
use the proper feature detection macros (e.g. looking for
defines automatically set by the compiler, instead of
relying on manually setting this define).
Change-Id: I1a675d653d6d305f558aa7f74c7255858dc62758
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The decoder will bypass all DRC processing during decode unless
any of the DRC parameters is set.
The fix consists in setting the DRC reference level to its
default value when after the decoder is initialized.
Change-Id: Ibb8355ab8c3c65020c5eef0752957040f7316c2a
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retraversing the array on overflow"
* commit '979df012eb0588ac7f87d5f6e8bff77e2049ee46':
amrnb: Restart the p_xn pointer when retraversing the array on overflow
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This is a safeguard to avoid issues in a rare codepath.
This was committed on the master branch of the opencore
repository in commit f532d145194e474cb30d8644b8eee83873ad24cc,
based on AOSP contribution 10905.
Change-Id: I9688b009bcbe63d88ee5040e883a021aefa30664
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* commit '10004fe612bc38bedc258fd1c380ebccacfa0c1e':
amrnb: Fix audio distortion in an AMR-NB clip
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This fix had the PacketVideo internal tracking number
RIO-9190.
This was committed on the master branch of the opencore
repository in commit 7d6ccc907d3a15ca2ff74b9c76f122ab87d0ffbe.
Change-Id: Icbd7d6fe94e02cae360946cfb6f3b8dc90c8b518
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