| Commit message (Collapse) | Author | Age | Files | Lines |
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* Original work by Michael Chen - https://github.com/omxcodec
* Kitkat port by Chih-Wei Huang / Android X86 project
* Additional fixes by Arcee and Cyanogen
* High resolution support by Cyanogen
* Lollipop port and refactoring by Cyanogen
* Marshmallow port and refactoring by Cyanogen
------
libstagefright: add null checking to addPlugin
libstagefright: fix error handling of dlsym()
According 'man dlsym', a NULL return from dlsym() doesn't indicate an error.
The correct way to test for an error is to call dlerror() to clear any old
error conditions, then call dlsym(), and then call dlerror() again to check
whether its return value is not NULL.
libstagefright: add more media mimetypes
This is the first step to add the ffmpeg plugins.
libstagefright: add ExtractorPlugin
The patch allows to load an ExtractorPlugin to extend the functions of
DataSource::sniff and MediaExtractor. A plugin has to implement a C
function getExtractorPlugin to fill the MediaExtractor::Plugin struct.
The filename of the plugin could be specified by the
media.stagefright.extractor-plugin property.
Change-Id: I995a37a4f1ab4bba6ca3c24c7001a27a1e3ccb90
FLACExtractor: Add more sample rates support
In FLACExtractor.cpp, it has function to check file's sample rate.
If the input sample rate is not in its list, it will return "unsupported
sample rate" issue. Modify code to make other sample rates (100,1k,42k,46k)
pass the check
Issue: AXIA-1441
Change-Id: I48f91119275560ec6d00feb0dedc70d10aa55262
Signed-off-by: Xiaobing Feng <xiaobing.feng@windriver.com>
Signed-off-by: Matt Gumbel <matthew.k.gumbel@intel.com>
libstagefright: add ffmpeg components
libstagefright: add more decoders
Add support for wma, wmv, ra, ape, dts decoders.
Change-Id: Iaf48a806aa0cef7d9bcb848383fc3d778c8bd248
libstagefright: allow to use the extended extractor in priority
If the meta contains the string "extended-extractor-use",
use the extended extractor first.
add support for rv20, rv30
add ffmpeg heuristic decoder
Change-Id: I5eed11b563ca7f15d44bacfb795d6f3da08ab883
add HEVC(H.265) decoder and cleanup
Squashed the following commits of branch cm_maguro-10.1 from
https://github.com/omxcodec/android_frameworks_av.git
by Michael Chen <omxcodec@gmail.com>
defb904 remove MEDIA_MIMETYPE_AUDIO_MP2
8036958 add fetchUriFromFd func to get file name
91bc7d5 fix videoCompressionFormatString and audioCodingTypeString funcs
f03069f reset FRAME_DROP_FREQ to 0
718b99a add HEVC(H.265) decoder
84f8bf6 cleanup
f026c93 cleanup
440614a add debug info
Change-Id: Ie75db0778f633357e2280aef6d47a0fa3beb823e
AwesomePlayer: use AwesomeLocalRenderer for OMX.ffmpeg.* components
stagefright: Remove duplicate types from QC media defs
Change-Id: I50ecafe79a2985d0868a1ac82464d6ca448aa2c5
Conflicts:
media/libstagefright/ExtendedMediaDefs.cpp
media/libstagefright/OMXCodec.cpp
libstagefright: Re-introduce a QCOM_HARDWARE ifdef after the FFMPEG commit
Fixes a build error on non QCOM hardware.
Change-Id: I4a4268b351d0d8bf748dd03ccea0fbab20ed4314
DataSource: Split off ffmpeg sniffer to a second pass, and only if necessary
ffmpeg's sniffer is intended as a grab-all-that-doesn't-work-elsewhere
extractor. Unfortunately, this causes two issues:
- As written, it requires ffmpeg to whitelist any extractors supported
by stagefright, or else it will blindly override them. This has codebase
sync issues, as shown by the VP9 and WAV accidental overrides
- It imposes an in-depth analysis of _all_ media, even that which we
want to be processed quickly by shipping stagefright plugins (hardware
or not). This is mostly noticeable in network streams and thumbnail
generation.
This patch moves FFmpegExtractor to an independent sniffer queue, and
it only invokes it when the regular MediaExtractor hits 1 of 3 conditions:
1 - The confidence in the identified container type is low
2 - No container is identified at all
3 - A video container was found by other extractors, but only 1 stream
(audio or video) was identified.
Change-Id: Ib96ff4f6bc06223fe0e819a57560d3c872a79ddd
stagefright: OMX.ffmpeg.* are software decoders, ensure they're treated as such
Wherever the component name for OMX.google soft decoders is used to identify
a software-based component, do the same for ffmpeg. Things like memory management
and window buffers care about this.
Change-Id: Ib83561936c7383e8726edb073cea9d78f7d1312f
libstagefright: Don't invoke FFMPEG for MP3
Change-Id: Ia30d25d1a994328827f14a286661cd2e1eaa1181
stagefright: Fix audio codec fallthru
* We shouldn't return an error from setAudioFormat unless it's
really an error since fallthru is necessary.
* We don't even need to do this, since the component name is
checked before calling into mm-parser or FFMPEG.
* Fixes Vorbis decoding after FFMPEG patch.
Change-Id: I4871c62044c6693e5698119dee3a10c20c26e2c7
stagefright: Fix codec lookup bugs on NuPlayer
* Fix use of WMA/WMV software codecs
* Fix mpeg2 software codec name
* Don't override the component name in ACodec. This actually breaks
stuff because the format isn't available in the kInit message.
Change-Id: I93c292e039de5f24c2ccbd6ae2242b06d28fe518
stagefright: Cleanup and improve format parsing
* Move FFMPEG-specific exceptions to FFMPEGSoftCodec
* Add handling for AAC MAIN profile
* Use the new OMX_AUDIO_CodingAndroidAC3 to handle AC3
Change-Id: Ibb806cd2b9dd23dc1e1b2c862fcde40605023a49
stagefright: Keep track of the bit width in the RAW codec
* We need this to push 24-bit PCM around Stagefright and OMX
Change-Id: Ic94ec972162a01545d5dd0ad0bf3eb6c6731f42e
stagefright: Adjust confidence threshold for extended sniffers
* Some sniffers return 0.2 for cases where they only find an audio
track in some containers.
* Change the comparison to also examine files right on this threshold.
* This allows us to score ONE FUCKING HUNDRED PERCENT on the Antutu
Video Test \o/
Change-Id: I78b6ab8a634771e0e290f669801f5b08d6a32a51
stagefright: Fix FFMPEG catchall decoders
* Get this metadata properly flowing
* Allows us to play tracks such as Apple Lossless :)
Change-Id: I2990b30eef5b672da339d24444424c61a43b85c2
stagefright: Fix metadata/message conversion issues
* Remove duplication of code between ExtendedCodec and FFMPEGSoftCodec,
just call into ExtendedCodec and properly ifdef QCOM-only parts.
* Fix CSD not being converted when AV_ENHANCEMENTS wasn't set- this
was breaking the software video codecs on Hammerhead.
Change-Id: I9cd4316ce19b15baabf12b78b992498ce48e2697
Fix compile error after I9cd4316ce19b15baabf12b78b992498ce48e2697
frameworks/av/media/libstagefright/ExtendedCodec.cpp:1187:1: error:
expected '}' at end of input
Change-Id: I7d75e69160f794b177f4235f4a6bb5a188dc0d08
stagefright: Fix AC3 playback
* Skip setupAC3Codec in ACodec for non-Google components.
Change-Id: I5090485ba020f7ad1c0962fc977e38675b4c8314
stagefright: Guard against crash with mismatched codecs
* Return unsupported error if WMV file can't be scanned.
Change-Id: Ia4a1ac7a299990f8b9c05a93736cb2fa9d0ee965
stagefright: Correct ifdeffage of some QC codecs
Change-Id: Ie8cc7287967b84e09941283559ca542efd928d91
stagefright: Create native window for FFMPEG software codecs
Change-Id: I178f334f1fa1ea9edc6898fb61e72902c2cb2651
stagefright: Don't ever try to use extended sniffers on DRM
* This can cause long retry intervals during key exchange. Don't do it!
Change-Id: Id9a87dcbe43cd0cc9919fe07f0a963e087baccad
stagefright: Be more tolerant of missing metadata for FFMPEG codecs
* If these codecs are instantiated programatically and required
metadata isn't sent, just set some defaults instead of crashing on
an assert.
* This fixes testAllNonTunneledVideoCodecsSupportFlexibleYUV in MR1 CTS
Change-Id: I69bf6105a1be529298de574bd5d3b6813e7a4e8f
stagefright: Fix issues with software decoders
* Fix MKV thumbnails
* Fix VC1 thumbnails
* Fix FFMPEG thumbnails
* Fix trial decoder
* Fix edge cases with WMV3/VC1 playback
* Fix a state issue which caused some codecs to get wrong configuration
Change-Id: I09599166aa24bcff53f91e43de096c4fad8ca7ad
stagefright: ffmpeg: Slightly raise the threshold for the ffmpeg scanner
0.2 is the success value for the OMX.google soft audio sniffers, which
was making ffmpeg own the unpacking of those streams needlessly.
Fixes CYNGNOS-282
Change-Id: I75f50ed838cb8af9acdf99aa284b80a070555284
stagefright: Add support for loading a custom OMXPlugin
* To facilitate moving the stagefright-plugins glue out of the
framework, support is added to OMXMaster to load multiple
external plugins besides internal/vendor versions.
* This is currently limited to one plugin, defined by the
"mm.sf.omx-plugin" system property. The code will allow any
number of libraries to be loaded, though.
* Should also be useful for nonstandard vendor implementations too.
Change-Id: I27d7e16ad56baa17754d8ea47a8c608a0d73d6f1
stagefright: Move a bunch of FFMPEG stuff out of here
* Get rid of some of the glue code for stagefright-plugins
and use the new extension header and plugin.
* Still a bunch of TODOs on this, but it works.
Change-Id: If07d3213952b624d48035e5f58ad883b2a4049b0
stagefright: Remove deprecated FFMPEG config
Change-Id: I1fcdb4eeba72e2420493b89ddd6fc718d170ced7
stagefright: Support for 24-bit audio in StageFright
* Plumb bit depth thru ACodec and OMX
* Add support for 24-bit PCM offload in NuPlayer on QC devices
* Use new AudioFlinger features for mixing multi formats without offload
* Clean up a bunch of code
Change-Id: I018d3a995b63450a38c6c43eaa37c86be30fd893
nuplayer: Fix PCM offload turning on all the time
* Remove the extra condition, since this will be set even if
PCM offload was denied.
Change-Id: I8f33ef68562d8e057e7a86c5ae6187d0049bf3aa
stagefright: Cleanup of PCM offload checks
* Put the checks in a single place.
Change-Id: I2d0d5b542593896e78bf989296de1a1d1e3a4963
stagefright: Add bit-depth plumbing for new formats
Change-Id: I13cfd75e4b4819543b64babf20cc9af57ea2978f
nuplayer: Fix bitrate propagation
* We use "bitrate" rather than "bit-rate".
Change-Id: I4699194e3e3f7ef55b4eb554f5de7a6b5f6b80ce
libstagefright: Implement fallback mechanism to SW decoder
Implement fallback mechanism to software decoder
when hardware decoder configuration fails in ACodec
Change-Id: Idf4c445942b03e28b264c91a20e69d52224727bd
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Use this struct to handle the parameters for TimestretchBufferProvider all
across the system.
Add stretch mode and fallback mode to TimestretchBuffer Provider.
Change-Id: I19099924a7003c62e48bb6ead56c785cb129fba2
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Bug: 19196501
Change-Id: I42d1f90e6297cf3f1304860d1691a5dfedd4c37d
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In preparation for playback rate support and timestretching.
Bug: 19196501
Change-Id: I435accb852d32110dd0b3a9917488522c567ba80
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Change-Id: I163f9d3d216c283ae1160ce4802e5247cf44fba7
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Previously conversion to AUDIO_FORMAT_PCM_16_BIT was required.
Client shared memory for static tracks and AudioFlinger track creation
and buffer delivery now use native 8 bit PCM data.
Change-Id: I485c07a4971fde9a25442bd43fed95019d39abcc
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* commit 'd0e12a161471823d1ab269cb608be3f3b0a71ed1':
C++11 compatibility.
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* Fix string literal concatenation to not be interpreted as UD
literals.
* Add constexpr compatibility for non-integral static members.
* Use __typeof__ instead of typeof (should become decltype once this
actually becomes C++11).
* Add an appropriate cast for atomic_uintptr_t, since moving to C++11
means moving from <stdatomic.h> to <atomic>, which has better
typechecking (hooray for not macros!).
Bug: 18466763
Change-Id: I9561dcb2526578687819ff85421ba80d8e1a9694
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Previously _if_ the full floating point mixer is enabled, a downmixer
would force the mixer input for a session submix to integer, breaking
other mixer inputs to the same submix that were in float.
Use another ReformatBufferProvider after the downmixer to solve
this issue.
Update the test-mixer app and the mixer_to_wave_tests shell
script to detect this issue.
Bug: 17363939
Change-Id: I74a56333f9ee75ddde39a75392c021c5eebddbef
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Bug: 17363939
Change-Id: I1f0f597b008f7369df87ba369fabaa93dec08ea1
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Legacy code that has not been hooked up.
Change-Id: Ic01a4eddfa1020089d5bc42f58b6a4697698c27a
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But not enabled (kEnableExtendedChannels == false).
Change-Id: I62f7e31fbd29ad703a9a02f5d1a280b6972dd423
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tests/test-mixer updated to handle multichannel.
New mixer path is still disabled (kUseNewMixer = false).
Subsequent CL will enable multichannel capability.
Change-Id: I71dafa94cc1d9af567c285e0ded7c5c94e3ff926
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This is used to convert between channel masks for input AudioTrack
data.
Change-Id: Iedffaeccb99c7053992152e7b1cb6b2d33b12eef
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Change-Id: I6b1a8755a1e6c29b916662fd376e9509cda5df55
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AudioMixer::ReformatBufferProvider now uses it as a base class.
Fix ReformatBufferProvider object leak.
Change-Id: If196f844eaaa124a173ffa27afe88098023c2ff9
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Use floating point volume in AudioMixer mixing when floating
point input is used with the new mixer engine.
AudioResampler is updated to take floating point volume to match.
Both legacy integer and floating point mixer engines work.
For now, integer volume is used when the new mixer engine
runs in integer input mode, for backward compatibility with
the legacy mixer. The new mixer engine will generally run in
floating point input mode. When the legacy path is removed,
the integer volumes will be removed.
Change-Id: I79e80c292ae7c8b8bdd0aa371a1b2c3a1b618290
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This change adds new process and thread hooks for
multi-format handling.
It is enabled by setting kUseNewMixer = true.
Change-Id: I262a3d2e4411f8cef7370a497b77a34eb55f1f86
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Change-Id: I4da1505ce852505f86f8e5b87f60e8edceeb30e0
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Change-Id: Ic040311305026f0b4c4280a5b3bef7a447ac1da3
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Change-Id: I04ac1cafd90b6ed652f8d51888ad07576678f0bc
Signed-off-by: Andy Hung <hunga@google.com>
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Change-Id: Ia152a839014e235fbfb656104c15d7c1b456d02e
Signed-off-by: Andy Hung <hunga@google.com>
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This will allow (eventually) a greater dynamic range for gains.
However there are still a few remaining places in effects and mixer
that will also need to be changed in order to get the full benefit.
Also fixes a minor bug: was not checking for NaN in AudioTrack C++.
Change-Id: I63bce9e82e0a61546d8ff475fb94bcb700d99c96
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AudioMixer::SINK_FORMAT also changes to AudioMixer::MIXER_FORMAT
Change-Id: Ic3f8be77d2c75c082c4fd140bc907e30c304d285
Signed-off-by: Andy Hung <hunga@google.com>
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Track Sink Format now allows both AUDIO_FORMAT_PCM_FLOAT and
AUDIO_FORMAT_PCM_16_BIT (default). The float case is not
enabled now.
Change-Id: Ibee70cab4725c8bc9905e49f1f9a9c2448f76e00
Signed-off-by: Andy Hung <hunga@google.com>
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Change-Id: I0fa61025c979709ad7d655bc717df5f194b6089e
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Use more standard coding convention for bit masks,
and add a FIXME about max channel count.
Change-Id: I856784016703417ee480b92ae73757c472f9cf95
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The code that initializes downmix GUID searches for an effect of
the right type. If two or more AudioMixers are created, there is a
possibility that the GUID will temporarily have the wrong value.
Now the initialization is only done once, regardless of how many
AudioMixers there are.
Other minor changes:
- use naming convention of leading 's' for static member variables
- log message for no downmix effect is now a warning instead of an error,
since it is not an error until there is actual content to downmix
without a downmix effect
Change-Id: I33ab0e51cf9d24ecaf8ae868883736b0772178c4
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Remove almost all of the specific logs, but leave the media.log
logging infrastructure in place for the next time we need it.
Re-apply a few good changes that were reverted earlier:
- check logf format vs. argument list compatibility
- distinguish potentially modified and actually modified tracks in FastMixer
- fix benign bug where sq->end() was called more than once
- fix a build warning
Bug: 6490974
Change-Id: I02d3e83646c738acaebb415bd0d6b548638b4ef5
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This reverts commit 32584a7d672864b20ab8b83a3cb23c1858e908b7
Change-Id: I9dc680578b955b1af462eeb7a49d61a0d45eb81b
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The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
- log mFastIndex when AudioMixer sees an invalid bufferProvider.
- log both potentially modified and actually modified tracks in FastMixer.
- fix benign bug where sq->end() was called more than once.
- log StateQueue push() call and return.
- increase StateQueue size from 4 to 8 entries
- log mixer->enable(), bufferProvider, and currentTrackMask
- log buffer provider addresses
- increase fast mixer log buffer again
- check logf format vs. argument list compatibility
- add logging to AudioMixer
- add checking of magic field in AudioMixer to detect overwrites
- add bool AudioMixer::enabled()
- increase log buffer sizes yet again
- enable assertion checking without ALOGV
- improve a few log messages
- check for corruption in more places
- log in all the process hooks
- add new mixer APIs so we can check for corruption of mixer state
- fix a build warning
Bug: 6490974
Change-Id: Ib0c4a73dcf606ef9bd898313b3b40ef61ab42f51
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This reverts commit 639482c24c911b125398b31883ba6d55faebe28b
Change-Id: I11f2829072ab11e18b0663024f27bf31192f1d39
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The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
- log mFastIndex when AudioMixer sees an invalid bufferProvider.
- log both potentially modified and actually modified tracks in FastMixer.
- fix benign bug where sq->end() was called more than once.
- log StateQueue push() call and return.
- increase StateQueue size from 4 to 8 entries
- log mixer->enable(), bufferProvider, and currentTrackMask
- log buffer provider addresses
- increase fast mixer log buffer again
- check logf format vs. argument list compatibility
- add logging to AudioMixer
- add checking of magic field in AudioMixer to detect overwrites
- add bool AudioMixer::enabled()
Bug: 6490974
Change-Id: I1f3f18aa62d9fbd35bc32285b669f5ba40efe28e
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This reverts commit 0ddd56316262ac74a95e9edb595697c163136d6d
Change-Id: I180a928af6f5a38d15a5efe44cd1fe927b5d961c
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The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
- log mFastIndex when AudioMixer sees an invalid bufferProvider.
- log both potentially modified and actually modified tracks in FastMixer.
- fix benign bug where sq->end() was called more than once.
- log StateQueue push() call and return.
Bug: 6490974
Change-Id: Iee7c8f40e20b6000cd8286c0ec6a14fff4a37af1
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Change-Id: Ib28fd7b9ce951a6933f006e7f8812ba617625530
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Change-Id: I83ea8bed375f251260945db788bdb5f280dba12d
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The audio downmixer effect might need the audio session Id, pass it
from the track creation in AudioFlinger to the downmix effect
creation in AudioMixer.
Change-Id: I836873eebd6711f1048fce81cd2eb29b94f0ad0c
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jb-mr1-dev
* commit 'c9beb01c800e6326051aee52a00659288c25c6eb':
Move libnbaio out of AudioFlinger
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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
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Change-Id: I5e6f0e371f728a5225c2fa6f778c109449d33602
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Pausing a video player will cause the track audio mixer to be
disabled, which causes the downmixer to be deleted. When reenabled,
the track channel mask hasn't changed but the downmixer is there
anymore.
Fixed by:
- instanciating a downmixer when the AudioMixer track
gets initialized (in getTrackName(), now taking a channel mask
as input), and deleted when in deleteTrackName().
- when the channel changes on a track, check whether it
needs a downmixer or not. Preparing a track for downmix
automatically removes the old downmixer if there was one.
Also: initialize the track downmixerBufferProvider field
when AudioMixer is instanciated, so we can safely call
delete on it in AudioMixer's destructor, in case
deleteTrackName() wasn't called before the mixer was
destroyed.
Change-Id: I589b0781cda5b3c82f85b561c52b08546cac21f8
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Add a new command REMOVE to remove any sample rate converter
on a track, without having to delete the track name.
Add comments.
Remove some dead code.
Fix whitespace and comparison to NULL.
Change-Id: Id55a23ab5ee673189e99675b4e6fa5df7b617987
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In the AudioMixer structure associated with each track, add an object
that acts as the buffer provider when the track has more than two
channels of input in the mixer. This object, DownmixerBufferProvider,
gets audio from the actual buffer provider of the track, and applies
a downmix effect on it.
The downmix effect is created and configured when the track gets
created in AudioFlinger, which causes AudioMixer::getTrackName()
to be called with the new track's channel mask. It is released
when the track is disabled in the mixer.
Change-Id: I05281ed5f61bef663a8af7ca7d5ceac3517c82db
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Change-Id: I371b17cef071d083eecf35cd3627a3adff907a33
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Change-Id: I327663a020670d0a72ff57bd0b682e2ce0528650
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uint16_t enabled is (mostly) changed to bool in a separate CL
Change-Id: Ied9f8c034b2479cee9a8778cee7b8ff92ae75b7b
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