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* Squashed commit of A/V changes from CodeAuroraKrishnankutty Kolathappilly2013-06-181-0/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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
* Merge tag 'android-4.2.2_r1' of ↵Steve Kondik2013-02-121-3/+24
|\ | | | | | | | | | | https://android.googlesource.com/platform/frameworks/av into 1.1 Android 4.2.2 release 1
| * New VHQ resamplerGlenn Kasten2012-11-271-3/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Squashed commit of the following: commit 12b6952da9f25e94d06dd7185bce255924e7e791 Author: Mathias Agopian <mathias@google.com> Date: Mon Nov 19 15:27:26 2012 -0800 fix a typo in SINC resampler that prevented tracks to be mixed we were always erasing the current mix instead of mixing into it. Change-Id: Ib229245f9e5a0d384f1727640a59e9f0469211a2 commit 0019ce082df430278f14ab922e900ce33b64897d Author: Dave Bort <dbort@google.com> Date: Tue Nov 13 01:30:32 2007 -0800 Rename "TARGET" to "MODULE" in the build system. Part one of the grand renaming. API_CHANGE: Third parties may need to update their makefiles. Any variables with "LOCAL" and "TARGET" in their names should now use "MODULE" instead of "TARGET"; e.g., LOCAL_MODULE, LOCAL_MODULE_TAGS. PRESUBMIT=passed OCL=39840 Change-Id: Ica9a7937d3d9552ab84db46ac6eea8a290e404fe Signed-off-by: Glenn Kasten <gkasten@google.com> commit f01adc0cef0e39e75c76d9195ac26a94cac0a100 Author: Glenn Kasten <gkasten@google.com> Date: Wed Nov 14 08:32:08 2012 -0800 Fix build warnings Change-Id: Ic43bcca166a529a6431711b05a7fa21849b6a38b commit 9bb031a565c753a03d9c9397edea318947d80528 Author: Mathias Agopian <mathias@google.com> Date: Sat Nov 10 04:44:30 2012 -0800 more optimizations... calculate the offsets from the phase differently, this happens to reduce the register pressure in the main loop, which in turns allows the compiler to generate much better code (doesn't need to spill a lot of stuff on the stack). this gives another 15% performance increase Change-Id: I2ce3479dd48b9e6941adb80e6d443d6e14d64d96 commit 5a951598f31217b8cd2babd0720c9608ee17291a Author: Mathias Agopian <mathias@google.com> Date: Sat Nov 10 03:26:39 2012 -0800 refactor code to improve neon code we want to make sure we don't transfer data from the neon unit to the arm register file, as this can be quite slow. instead we do all the calculation on the neon side and write the result directly to main memory. Change-Id: Ibb56664d3ab03098ae2798b75e2b6927ac900187 commit b381ee9e83bc9fd18986e79c7809841514ed590e Author: Mathias Agopian <mathias@google.com> Date: Sun Nov 4 15:16:13 2012 -0800 NEON optimized SINC resampler this currently gives us a 60% to 80% boost depending on the quality level selected. Change-Id: I7db385007e811ed7bffe5fd3403b44e300894f5b commit bea077354210242ea193a50b0dbab0fedab25df3 Author: Mathias Agopian <mathias@google.com> Date: Mon Nov 5 01:51:37 2012 -0800 minor cleanups Change-Id: Ia12ee4fb59e90221761bec85e6450db29197591f commit 8f4ed7decbe161a5ff38200b218f5216d80aba46 Author: Mathias Agopian <mathias@google.com> Date: Sun Nov 4 18:49:14 2012 -0800 improve resample test - handle stereo input - input file can now be ommited, in this case a linear chirp will be used automatically - better usage information Change-Id: I5d62a6c26a9054a1c1a517a065b4df5a2cdcda22 commit 5fcd634ea6cb4df27c495abe20f5f9b8ff55d128 Author: Mathias Agopian <mathias@google.com> Date: Sun Nov 4 02:03:49 2012 -0800 change how we store the FIR coefficients The coefficient table is now transposed and shows much better its polyphase nature: we now have a FIR per line, each line corresponding to a phase. This doesn't change at all the results produced by the filter, but allows us to make slightly better use of the data cache and improves performance a bit (although not as much as I thought it would). The main benefit is that it is the first step before we can make much larger optimizations (like using NEON). Change-Id: Iebf7695825dcbd41f25861efcaefbaa3365ecb43 commit d652231abf4c7e2ea1fc89caae730cec1f7259a1 Author: Mathias Agopian <mathias@google.com> Date: Sat Nov 3 23:37:53 2012 -0700 improve SINC resampler performance The improvement is about 60% by just tweaking a few things to help the compiler generate better code. It turns out that inlining too much stuff manually was hurting us. Change-Id: I8068f0f75051f95ac600e50ce552572dd1e8c304 commit 9dc68ef5b94c700c4ee68790e8cbb334c90a538d Author: Mathias Agopian <mathias@google.com> Date: Thu Nov 1 21:03:46 2012 -0700 new coefficients for the vhq resampler previous coefficients were provided by a 3rd party and didn't have a way to re-generate them. we're now using the 'fir' utility. the performance of the filter is virtually identical, except for the down-sampling case which seems slightly better now: It looks like both the previous and new coefficients are generating some sort of clipping for full-scale signals in the down-sampling case (although the new ones seem better), the reason for that is unknown (see bug: 7453062) Also updated the HQ coefficients for the down-samplers, previous ones were a little bit too conservative -- the new ones push the cut-off frequency up by about 1 KHz. Change-Id: I54a827b5c707c7cc41268ed01283758dce1d7647 commit 38e0b8560a6fc1b7124e22e0e09a84a285182f8e Author: Mathias Agopian <mathias@google.com> Date: Tue Oct 30 13:51:44 2012 -0700 fix SINC resampler on non ARM architectures make sure the C version of the code generates the same output than the ARM assemply version. Change-Id: Ide218785c35d02598b2d7278e646b1b178148698 commit a1878128b182696ba508569b4d211d0dfae92463 Author: Mathias Agopian <mathias@google.com> Date: Tue Oct 30 12:49:07 2012 -0700 fix another issue with generating FIR coefficients the impulse response of a low-pass is 2*f*sinc(2*pi*f*k), we were missing the 2*f scale factor. This explains why we were seeing clipping and had to manually scale the filter down. Change-Id: I86d0bb82ecdd99681c8ba5a8112a8257bf6f0186 commit 1a0fb993430acc9f601e6c538305bc407c20ac5d Author: Mathias Agopian <mathias@google.com> Date: Mon Oct 29 17:13:20 2012 -0700 fir a typo that caused up-sampling coefficiens to be wrong up-sample coefficient were generated with a cut-off frequency of 24KHz intead of ~20KHz, which caused more aliasing in the audible band. also increased the attenuation to 1.3 dB on both up and down sampling coefficient to avoid clipping. Change-Id: Ie8aeecf1429190541b656810c6716b6aae5ece2e commit 9520ad6862bd682ad075a9d9e3e94ada9f6e58b6 Author: Mathias Agopian <mathias@google.com> Date: Mon Oct 29 17:13:16 2012 -0700 test-resample: clip instead of overflowing Change-Id: I550e5a59e51c11e1095ca338222b094f92b96878 commit ba36656300f250f7f1fdeb75149749344260e6cb Author: Mathias Agopian <mathias@google.com> Date: Sun Oct 21 01:01:38 2012 -0700 a test app for the resamplers Change-Id: I66852d90d384f1d9e77b51ad1a1ebdbaf61d0607 commit 056a08b9bfd33cf27228c992adc8293a56b01be8 Author: Mathias Agopian <mathias@google.com> Date: Fri Oct 26 14:11:01 2012 -0700 reenable the cubic resampler cubic resampler was disabled because it hadn't been qualified, however after I did some tests, it does improve significantly the sound quality over the order-1 resampler, even if it is still quite bad. also HIGH_QUALITY resampler was partially disabled, it's now fully enabled. It's a big improvement over the cubic resampler in terms of aliasing noise (it's not as good in the pass-band). Change-Id: I70e3658c255896588642697be9eb594ff4ec0f8b commit 8c0241d3ff50ae85167f69b3bd369244894cfa44 Author: Mathias Agopian <mathias@google.com> Date: Fri Oct 26 13:48:42 2012 -0700 improve SINC resampler coefficients - we increase the interpolation precision from 4 to 7 bits this doesn't increase CPU power required, it only increases the size of the filter table but significantly reduces the noise introduced by the quantization of the impulse response. - the parameters of the filter are set such that aliasing is rejected at 80 dB below 20 KHz. Because we don't use a lot of coefficient (to save compute power), there are quite a bit of attenuation in the pass-band: starting at 9KHz for the down-sampler (48 to 44.1), and starting at 13 KHz for the up-sampler (44.1 to 48) -- the transition band is about 15 KHz. Change-Id: I855548d2aab8a0fb0d2a2da3a364b6842d7d3838 commit 69e7dab2192adc1f780464146810629ebd01b145 Author: Pixelflinger <mathias.agopian@gmail.com> Date: Thu Oct 25 19:43:49 2012 -0700 improve fir tool: cleanup, better default, bug fixes - all parameters can be changed on the command-line - added float output - added debug option - added an option to generate a polyphase filter coefficients - added an attenuation option in dBFS - added a lot of comments and references - fixed kaiser window parameter also the default should generate a filter with 80 dB rejection (of the 24 KHz aliasing) above 20 KHz and a 15 KHz transition band around ~20 KHz (for 48 KHz sampling rate). It's not very good but corresponds to the current code. commit 8347499d105a50257c18e9dac652e750b06428b1 Author: Glenn Kasten <gkasten@google.com> Date: Mon Oct 22 17:09:27 2012 -0700 Increase allowed number of VHQ resamplers to 3 Bug: 7378660 Change-Id: I69e33ca2eb4bb9bd38e2c63df62cd1130d68baf6 commit f91cf3cad7f5c4d52614271c89ab468741c5d24c Author: Mathias Agopian <mathias@google.com> Date: Sun Oct 21 03:04:05 2012 -0700 Fix a typo that caused the high quality resampler to produce garbage the problem is that if libaudio_resampler is present, it is those coefficients that will always be selected, but the correct meta-data. Bug: 7385994 Change-Id: Ieebeb37b4dfb62a1a051bc29fae2ce056dbc6621 commit e158a9e4262a174c59469a205658bc3ca4078234 Author: Dan Bornstein <danfuzz@google.com> Date: Fri Oct 3 10:34:57 2008 -0700 Manually merge change #111620 from tc3 to mainline, to keep the automerger from choking on it. p4 sync p4 integrate -r -b android_to_tc3 //...@111620,111620 p4 resolve -a p4 resolve # resolve a couple merge travesties PRESUBMIT=passed BUG=1399648 TBR=edheyl OCL=111902 Change-Id: I854b01553dd92bbf9c864f5a9bd51a3d665f0ac2 Signed-off-by: Glenn Kasten <gkasten@google.com> commit b9f3c26032be7a6ea01a10d93d94826f449e68ab Author: Dave Bort <dbort@google.com> Date: Fri Jan 18 14:51:05 2008 -0800 Rename "Makefile" to "Android.mk" throughout the tree. For <http://b/issue?id=960416>. I've tested this as much as I can, but 1500 open files = easy to mess things up. Please let me know if there's a problem rather than rolling back this change. PRESUBMIT=passed BUG=960416 TBR=joeo OCL=46537 Change-Id: I5a404caf0f398a7afa7ae7abaf2f2a1c6ab490eb Signed-off-by: Glenn Kasten <gkasten@google.com> commit 0c22a9a44c4103483fba1d944acf1354c5eb1617 Author: Mathias Agopian <mathias@google.com> Date: Mon Oct 29 23:44:25 2007 -0700 Tweak the SINC resampler parameters and double the performance. It's using about 10% CPU in the worse case now. Change-Id: I50ac7e6c6702a427fa36ab6d976c507155057507 Signed-off-by: Glenn Kasten <gkasten@google.com> commit b85e41487983ad085b859acf8251e7e54480308a Author: Mathias Agopian <mathias@google.com> Date: Mon Oct 29 04:34:36 2007 -0700 A sinc resampler for Audioflinger. It's not enabled yet, but fully functional and apparently working. It need more "quality" tests. In the 48->44 KHz, it takes about 25% of the CPU time. Change-Id: I80eb5185e13ebdb907e0b85c49ba1272c23d60ec Signed-off-by: Glenn Kasten <gkasten@google.com> commit ba3949ef17cac2ba71cc3096c413782a49c922e5 Author: Mathias Agopian <mathias@google.com> Date: Thu Aug 23 21:01:28 2007 -0700 fix a few small typos in the FIR computation Change-Id: I6e56b514fe520f30f7487f85c64ea5d2a7c19b40 Signed-off-by: Glenn Kasten <gkasten@google.com> commit 7474bfa7de2604021963794dddfe44985648db6a Author: Mathias Agopian <mathias@google.com> Date: Thu Aug 23 03:16:02 2007 -0700 This is a tool to compute the the reconstruction filter coefficients for a sinc audio resampler. Change-Id: I99be2505139b8e0e7647200e1647509d4f7e6067 Signed-off-by: Glenn Kasten <gkasten@google.com> Bug: 7577965 Change-Id: I2c84a9283a1668723bad83e1a119c849c88c3e6b
* | Squashed commit of updates from CodeAuroraMingming Yin2013-02-061-0/+4
|/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* audioflinger/resampler: add build source for libaudio-resamplerty.lee2012-10-071-0/+2
| | | | | | | Bug: 7229644 Change-Id: I93bde36be1c3ec84174a4c98423e28f8b3d8782f Signed-off-by: ty.lee <ty.lee@lge.com> Signed-off-by: Iliyan Malchev <malchev@google.com>
* Integrate improved coefficient sinc resampler: VHQGlenn Kasten2012-10-041-1/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | Summary: Very high quality is enabled only for 44.1 -> 48 or 48 -> 44.1, and uses low quality for all other use cases. Track estimated CPU load and throttles the quality based on load; as currently configured it should allow up to 2 instances of very high quality. Medium quality and high quality are currently disabled unless explicitly requested. Details: Only load .so the first time it is needed. Cleanup code style: formatting, indentation, whitespace. Restore medium quality resampler, but it is not used (see next line). Fix memory leak for sinc resampler. Check sample rate in resampler constructor. Add logs for debugging. Rename DEFAULT to DEFAULT_QUALITY for consistency with other quality levels. Renumber VERY_HIGH_QUALITY from 255 to 4. Use enum src_quality consistently. Improve parsing of property af.resampler.quality. Fix reentrancy bug - allow an instance of high quality and an instance of very high quality to both be active concurrently. Bug: 7229644 Change-Id: I0ce6b913b05038889f50462a38830b61a602a9f7
* Disable audio watchdogGlenn Kasten2012-09-301-2/+2
| | | | | | | It's not critical, and is wasting power Bug: 7241714 Change-Id: I6ad4375f0000c92529688723dbe0ff0caa809c5d
* audioflinger: use resample coefficients from audio-resampler library.SathishKumar Mani2012-09-261-2/+2
| | | | | | | | | | | -Add a separate quality VERY_HIGH_QUALITY in resampler -Use resample coefficients audio-resampler library for quality VERY_HIGH_QUALITY. -This improves the quality of resampled output. Bug: 7024293 Change-Id: Ia44142413bed5f5963d7eab7846eec877a2415e4 Signed-off-by: Iliyan Malchev <malchev@google.com>
* Move libnbaio out of AudioFlingerGlenn Kasten2012-08-301-27/+1
| | | | | | | | | | libnbaio is now a separate shared library from AudioFlinger, rather than a static library used only by AudioFlinger. AudioBufferProvider interface is now also independent of AudioFlinger, moved to include/media/ Change-Id: I9bb62ffbc38d42a38b0af76e66da5e9ab1e0e21b
* Remove debug code HAVE_REQUEST_PRIORITY and SOAKERGlenn Kasten2012-07-031-1/+1
| | | | Change-Id: I73a2afe72d8acb53e57e6b4e6fb5133e22b7875a
* Make CPU frequency statistics optionalGlenn Kasten2012-06-141-0/+3
| | | | | | | | | | Certain CPUs with dynamic cluster swapping and hotplug don't report CPU frequency accurately. The file descriptors used to read the frequency become stale and report bogus data. So make this feature a build time option for debugging only. This will also improve performance of the fast mixer loop. Change-Id: I602f81ec3281a37992769208be08084ed1469e8c
* Add audio watchdog threadGlenn Kasten2012-06-111-0/+4
| | | | Change-Id: I4ed62087bd6554179abb8258d2da606050e762c0
* Reduce underruns in screen off, esp. with EQGlenn Kasten2012-06-081-0/+3
| | | | | | | | | | | | Add MonoPipe APIs to specify setpoint. Use screen state to configure pipe setpoint. Fix a long-standing bug where pipe sleep time was excessive, which interacted poorly with governor and low clock frequencies. Now it deducts the elapsed time since last write(), which was significant when there was EQ and low clock frequency. Bug: 6618373 Change-Id: I6f3b0072c2244aeb033ef0795ad164491a164ff5
* State queue dumpGlenn Kasten2012-06-041-0/+3
| | | | | Bug: 6591648 Change-Id: Iac75e5ea64e86640b3d890c46a636641b9733c6d
* Keep a copy of most recent audio playedGlenn Kasten2012-05-211-0/+4
| | | | Change-Id: I6b2f97881c39998a2fae9ab79d669af6c0a37e94
* systrace for audioGlenn Kasten2012-05-171-0/+3
| | | | | | | | | | Trace fast track buffer fill status for underruns etc. Move the definition of macro to Android.mk. No overhead if disabled. Change-Id: If0e83e21b61b059ca38f543f8a6ffb58e08c79ee
* Use scheduling policy serviceGlenn Kasten2012-04-241-1/+13
| | | | Change-Id: I3c09da1dc0de5039d0c15ce7fb2bc373fa398712
* AudioFlinger normal mixer uses FastMixerGlenn Kasten2012-04-231-1/+1
| | | | Change-Id: I3131bb22d2d057e9197a2ebfa6aa1cfaab9e5321
* Configure policy of mediaserver threadsGlenn Kasten2012-04-221-0/+2
| | | | Change-Id: Ifd825590ba36996064a458f64453a94b84722cb0
* FastMixer updateGlenn Kasten2012-04-201-1/+1
| | | | | | | | | | Updates: - Add support for mono fast tracks - Add support for optional sample rate conversion on fast tracks - Log sample rate and frame count - Enable statistics Change-Id: Ife014edf4f452da361f3eaaae19209ef6ff6958b
* Fast mixerGlenn Kasten2012-04-181-0/+6
| | | | Change-Id: I61552f83507e08e4c706076b9fb15362869e6265
* Add template class StateQueueGlenn Kasten2012-04-181-0/+2
| | | | Change-Id: Iccc5eb42bc295a22b2e429a4551f083cd7b6831a
* Non-blocking audio I/O interface, WIPGlenn Kasten2012-04-021-0/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Yet another abstraction similar to AudioTrack::Buffer and AudioBufferProvider, but with support for streaming, non-blocking, and eventually PTS. This is intended to be used as follows: - primary HAL output stream implements a Sink - primary HAL input stream implements a Source - Pipe implements a Sink - PipeReader implements a Source or TimedSource (not shown yet), which supports "read at PTS" - fast AudioTrack on server side will implement a Source using cblk - normal AudioTrack on server side will not be changed initially - fast AudioRecord on server side will implement a Sink using cblk - normal AudioRecord on server side will not be changed initially - fast mixer thread will read from Sources and write to a Sink, or (unlikely) implement a Source and multiple Sinks - Visualization and PCM logger will read from Source or TimedSource - A2DP normal mixer will be connected directly to its output stream and there will be a kind of OutputTrack for duplication that will read from a Sink with non-blocking write fed by the fast mixer. Patch set 3 changes: - Add more implementations of NBAIO interfaces: added SourceAudioBufferProvider, MonoPipe, MonoPipeReader. - Added Format_sampleRate and Format_channelCount. - Extract out the roundUp() method. - Respond to most comments from previous code review. - The new classes are untested. Patch set 4 changes: - Fix bugs in MonoPipe::write() and MonoPipeReader::read() - Fix bug initializing mFrameBitShift too early - renamed roundUp() to roundup() - Fix Android.mk - Add LOG_TAG an LOG_NDEBUG, use ALOG_ASSERT and utils/Log.h instead of assert - Fix build warnings - Move constructor and destructor bodies from .h to .cpp - Line length 100 - Following naming conventions for #include double-include protector macros - Include what you use - More NBAIO logging - MonoPipe write can be blocking Patch set 5 changes: - Address code review comments - Use a static library so unused implementations don't take memory - Comment out libsndfile dependency - Remove debugging LOGV and LOG_NDEBUG Patch set 6 changes (would be 6 at old location, actually 2 at new location): - Address code review comments on patchset 5 - For MonoPipe, allow the full pipe to be used, no need to omit one slot - Don't do atomic releasing stores unless needed Still to do: - I'm not happy with the Pipe class names - Update build/ for new static library? Change-Id: Ie6c61f05ce06b676b033be448a8ef9025a2ffcfd
* Revert "AudioFlinger does not need libmedia any more"Glenn Kasten2012-03-231-0/+1
| | | This reverts commit c920dee060ac69684be33210ee44b99a5fc3e8b2
* AudioFlinger does not need libmedia any moreGlenn Kasten2012-03-221-1/+0
| | | | Change-Id: Ifd2c61882109ec36ca68072a2bf6506e08c8cf34
* Add libmedia_nativeGlenn Kasten2012-03-191-0/+2
| | | | Change-Id: I3ac357c78fb89f108d15c6e5b9fa317de0e9fb9a
* Remove dependency on audio_* locationGlenn Kasten2012-03-141-2/+2
| | | | Change-Id: I4bc66115fcb9ba22b057bd72db3f561dcb18a0d8
* AudioBufferProvider comments and cleanupGlenn Kasten2012-02-241-1/+0
| | | | | | | | | | | | | | | | | | | | Add comments about which methods implement the AudioBufferProvider interface. Simplified the definition of kInvalidPts. <stdint.h> is very hard to work with, there seems to be no way to use it reliably to get INT64_MAX without having a separate source file, which is ugly because it means kInvalidPts is not a compile-time constant. So I just deleted AudioBufferProvider.cpp and used a hard-coded constant instead. Added a default constructor for Buffer so that the fields aren't random (especially .raw which is used to determine if the buffer is valid). Make the pts for getNextBuffer default to kInvalidPTS so code that doesn't need a pts doesn't have to specify a value. Rename the parameter to AudioMixer::setBufferProvider to make it clearer. Change-Id: I87e7290884d4ed975b019f62d1ab6ae2bc5065a5
* Upintegrate Audio Flinger changes from ICS_AAHJohn Grossman2012-02-161-0/+2
| | | | | | | | Bring in changes to audio flinger made to support timed audio tracks and HW master volume control. Change-Id: Ide52d48809bdbed13acf35fd59b24637e35064ae Signed-off-by: John Grossman <johngro@google.com>
* Factor out and speed up permission-checking codeGlenn Kasten2012-02-131-1/+2
| | | | | | | | | | | | | Use the caching permission check for dump to save IPC. Cache getpid() to save kernel call for other permission checks. The C runtime library getpid() can't cache due to a fork race condition, but we know that mediaserver doesn't fork. Don't construct String16 on the stack. Change-Id: I6be6161dae5155d39ba6ed6228e7683e67be34ed
* Disable HQ resamplers for now until qualifiedGlenn Kasten2012-02-091-2/+2
| | | | | | This saves about 6500 bytes. Change-Id: I87102fe561c95c19c9e615dea3de914f96639257
* Extract out audio DSP code to utility libraryGlenn Kasten2011-12-161-1/+3
| | | | Change-Id: Ib8ce72028a7ea30e82baa518e381370e820ebbd0
* Fix issue 4604090: notification sound interrupted.Eric Laurent2011-07-251-1/+2
| | | | | | | | | | | | | The problem is that the audio HAL fails to acquire the wake lock when playing the notification. This is because of a change that removed the mediaserver process form the system group for honeycomb. The fix consists in requesting the wake lock from PowerManagerService when AudioFlinger mixer wakes up. A consequence of this change is that audio HALs or pcm drivers do not have to hold wake locks anymore as in the past. Change-Id: I4fb3cc84816c9c408ab7fec75886baf801e1ecb5
* Merge "Log CPU usage"Glenn Kasten2011-07-151-0/+1
|\
| * Log CPU usageGlenn Kasten2011-07-151-0/+1
| | | | | | | | Change-Id: Ie447e59be139153e526b7ad467c46c659d26816f
* | Remove the simulator target from all makefiles.Jeff Brown2011-07-111-13/+2
| | | | | | | | | | | | Bug: 5010576 Change-Id: I04d722f258951a3078fe07899f5bbe8aac02a8e8
* | Moved and renamed effect API header filesEric Laurent2011-07-071-0/+3
|/ | | | | | | | Moved specific effect header files to system/media/audio_effects/include/audio_effects and renamed to lower case (effect_xxx.h). Change-Id: Icfc2264bfd013cab0395d7e310ada636b9fe3621
* audio/media: convert to using the audio HAL and new audio defsDima Zavin2011-04-271-0/+3
| | | | | Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5 Signed-off-by: Dima Zavin <dima@android.com>
* audioflinger: move legacy audio hw/policy out to libhardware_legacyDima Zavin2011-04-271-87/+0
| | | | | Change-Id: I4adcec73d3c08bcbe15bb19e1ba2ff18b195af45 Signed-off-by: Dima Zavin <dima@android.com>
* Bug 3366885 Remove LVMX switchGlenn Kasten2011-02-031-8/+0
| | | | Change-Id: I0bf98c6f85f00b3296874571e1c049dcc4e2fcca
* move native services under services/Mathias Agopian2010-07-141-0/+131
moved surfaceflinger, audioflinger, cameraservice all native services should now reside in this location. Change-Id: Iee42b83dd2a94c3bf5107ab0895fe2dfcd5337a8
generated by cgit v1.1 at 2024-06-03 21:23:10 +0000