| Commit message (Collapse) | Author | Age | Files | Lines |
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&pDwmModule->config may equal to pConfig that triggers
a building error using gcc 4.7:
In file included from bionic/libc/include/signal.h:33:0,
from bionic/libc/include/sys/select.h:34,
from bionic/libc/include/unistd.h:34,
from system/core/include/cutils/log.h:34,
from frameworks/av/media/libeffects/downmix/EffectDownmix.c:19:
In function 'memcpy',
inlined from 'Downmix_Configure' at frameworks/av/media/libeffects/downmix/EffectDownmix.c:651:11,
inlined from 'Downmix_Init' at frameworks/av/media/libeffects/downmix/EffectDownmix.c:605:9,
inlined from 'DownmixLib_Create' at frameworks/av/media/libeffects/downmix/EffectDownmix.c:233:9:
bionic/libc/include/string.h:114:31: error: call to '__memcpy_overlap_error' declared with attribute error: memcpy called with overlapping regions
Change-Id: Ia6c4aab72e38d00d8d526f3b56fc03b6530b4e44
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MPEG4 audio packets may be multiplexed using the so called
LATM (Low Overhead Audio Transport Multiplex) scheme.
LATM parsing was recently introduced in Stagefright and it
has caused issues in cases when the LATM config element
cannot be parsed correctly. The main problem occurrs when
the AudioSpecificConfig part of the config element contains
more information than what is expected, causing the
frameLengthType parameter to get the wrong value. This fix
introduces default values of some config parameters that are
used in case config parsing fails.
Change-Id: I3cb35df76826f95ca0831dc08c2a1e7c6c2c586d
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FLACExtractor had artificial limits which did not allow multichannel
audio, 88.2 or 96 kHz samplerate, these were removed since it is up
to the audio subsystem to handle.
Change-Id: Iadc20cdf4724814b5c980827a528a1e1b4bd6a82
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support stereo/multichannel 24-bit format
kMaxFrameSize is fixed to 32768. When converting 24-bit to 16-bit,
number of samlpes is maxBytesToRead /3.
In this case, if the maxBytesToRead is not multiple of 3,
pcm data is messed when converting.
Bug:7630939
Change-Id: I0ea1b53eb1272a8d83b63815fc0a05b73cef75f1
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See I2c0a87b0fc049e855e21f455b806287b578091de
Change-Id: Iec8566bce0a4763a9a17b4a46f15aa39b5bcd141
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Add destructor to call fclose.
Change-Id: I765c56236f8df58af998439492b0b955ab57606f
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Remove checks that causes crash for rtsp streamed h.263 content
with certain values in the RTP payload header:
Remove zero check for the five reserved bits in the payload header.
According to RFC 4629 these bits MUST be ignored by receivers.
Remove zero-check for the VRC (Video Redundancy Coding) bit,
skip packet instead.
Remove zero-check for the PLEN bits (extra picture header),
skip packet instead.
Remove zero-check for the PEBIT bits (extra picture header),
skip packet instead.
Remove corresponding zero check for the four resreved bits in the
AMR payload header. According to RFC 4867 these bits MUST be
ignored by receivers.
Change-Id: I7fc21d69a19d23da24f9267623c338d415ef1387
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Repeated leaks detected in mediaserver context. Most chunks small but
some bigger. Adding up it was not uncommon to see 200Mb footprint.
libc memdebug allocation call stack showed the root cause in "covr"
parsing - that explains why content often was empty but sometimes big.
Change-Id: I0de6c1eeef174d2529973ff9c6a020ec3dd44c75
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For an MP3 source, within the prepare command, ID3 tags are checked in search of
gapless playback info. This causes problems for streamed sources. If ID3v2 tags
aren't present, then a check is done for ID3v1 tags. This results in a read
command that asks the cache to jump to the end of the file, and subsequently
make an extra http call to request those bytes. For a streamed source, this
causes the file to not be downloaded automatically when MediaPlayer::prepare()
is called, and causes stuttering and extra buffering time to be needed when
start() is finally called.
The solution is to ignore the ID3v1 tags as the gapless info would never exist
there, and only check for ID3v2 tags.
Change-Id: I7d1b94cffbfe7c38ca094834dedbc92a58855e20
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If the set up of the RTSP stream contains an incorrect or otherwise
problematic URL, some servers will send an unsolicited server response
that contains a negative number in the sequence number (CSeq) field.
This negative value is not returned from the function findPendingRequest(),
so the check in notifyResponseListener() will not work. Instead there will
be a crash when 0 is used as the index to find a matching request/response
pair that doesn’t exist.
The fix is to return the received sequence number also when it is an
unsolicited server-client message.
Change-Id: Iedaba8a63dece7b43bce007069baefbfd10970b8
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This fix handles problems with several asynchronous calls
within streaming. This case is when the phone has sent a
request to the server and while the response is being sent
back by the server the request is aborted by the user.
The fix is an if case that checks if we have aborted while
waiting for a response from the server. If we have aborted
we should ignore the late response instead of continuing.
Change-Id: I1264bb992f6abcaee1f10a89479e08b54a95e3c2
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If the codec was chosen based on mime type, the caller does
not know what component actually was chosen. This allows
getting essential information (such as supported color formats,
for a video encoder) for this component.
Change-Id: Ie471f40f8104b37d27ced3dba5a54facc6504b1b
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This fix is protecting AudioTrack::dump from SIGSEGEV
when calling dumpsys shell command.
Change-Id: I30d136e195a12d6fdad41e14f557b0ad9e59b8a2
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Don't duplicate strings when retrieveing metadata from media
files. As any requests for metadata strings would pass through
the binder, this would cause the reference to the duplicate string
to be lost, causing a memory leak as the duplicate would not be
freed.
Change-Id: I2593733472b1bb589bc502b2c11080f581015bb5
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If enabling seq_scaling_matrix_present_flag = 1 the AVCC chunk can
be larger than the original buffer size. Changed to using ABuffer
instead.
Change-Id: Idacc14b45ea2634c5e608919f3ce567f23363135
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The MPEG4Extractor was parsing 8 bits instead of 16 bits when parsing
'trkn' and 'disk'. Also added support for 16 bytes size 'disk'.
Change-Id: I22b4de2ac800881884d5759776cb380917522a87
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The *.mpga files are playable, but are not correctly scanned.
This is because they are prevented from being scanned in
StagefrightMediaScanner.cpp.
What this fix does is to add the extension to the list of valid
file extensions so that the scanner handles the filetype properly.
We have previously added the .mpga extension to the framework to
make it playable, but not added it so that the scanner scans it.
Change-Id: I02a44d770adb80d38e8bed77d0d76efa1b28a861
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* commit '1c6833afd6cfdbc6c696ed0719ceedc37ec0b534':
Ignore SIGPIPE during write() to broken pipe
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This happens occasionally when taking a bugreport.
Bug: 6447319
Change-Id: Ia6531a4a3658461f8fd3f7106e7996da7cc5933a
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configurable
* commit '96076964863ee6887f0bed9d0f11f424b48ab9b9':
Make video encoders' bitrate mode (constant, variable, ...) configurable
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through the desired output format. Configure the video encoder to use
constant bitrate mode for WFD.
Change-Id: Id7bd619598153c13448a9c5acd69d80f8a01f333
related-to-bug: 7459597
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* commit '90a92053219ae50ddf4bb54e3d54db2d309e2b8d':
Various tweaks to wfd
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related-to-bug: 7426218
Squashed commit of the following:
commit 1553f1a1c66af998674168f7f7a3be23fcb0c794
Author: Andreas Huber <andih@google.com>
Date: Tue Oct 30 15:51:27 2012 -0700
Add LPCM, AVC and AVC HRD descriptors as necessary.
Change-Id: Ibc836fced0fe37e8a25574c2295e886765b9ea6f
commit 4e74db61d2d31ebe239acbdec8f110f88016a4ea
Author: Andreas Huber <andih@google.com>
Date: Tue Oct 30 15:50:52 2012 -0700
added copyright headers to Sender.{cpp,h}
Change-Id: If615ccb8767e32bd83ed1f0f669acc39a72489f6
commit 7144bf8ae68c5cdb8faa6e219547aabbd750f04e
Author: Andreas Huber <andih@google.com>
Date: Tue Oct 30 15:50:25 2012 -0700
Reenable suspension of the RepeaterSource
Change-Id: I765338fcde89c65e4b69be45a5949eba6bcdcf6f
commit 812164bcfa0699821d7d8eefcc0dff96b2e2cd08
Author: Andreas Huber <andih@google.com>
Date: Tue Oct 30 14:03:50 2012 -0700
Add 2 stuffing bytes to the PES headers for audio tracks.
Change-Id: I8b9c634f6a565ab7fa7ecdb610f7d8557e0b139b
commit a084a741a63015d47c92d99fcd8b980fe615dc7d
Author: Andreas Huber <andih@google.com>
Date: Tue Oct 30 13:19:38 2012 -0700
Fix PCM audio packetization in WFD.
Change-Id: I99a435f9fe6b4397f24d6c22afae5ae2505ffc14
commit c5cb9369585f701f34bce41534940d5f9b59248f
Author: Andreas Huber <andih@google.com>
Date: Tue Oct 30 13:19:12 2012 -0700
Support extraction of PCM audio from transport streams.
Change-Id: I28a0516756ebcb5587325b6588df013ac871ffb9
commit b0a0512300ae037d6b39c2d04952d34b5fc12b2d
Author: Andreas Huber <andih@google.com>
Date: Tue Oct 30 08:54:13 2012 -0700
disable suspend of the RepeaterSource
Change-Id: Ibf42a98185b0567f817ae582a82e6580f95d3d40
commit 4330e8b7668dc92a6d882b5622c0697cf292d04c
Author: Andreas Huber <andih@google.com>
Date: Mon Oct 29 14:11:25 2012 -0700
Better handling of datagrams in ANetworkSession
reduce unnecessary copy overhead.
Change-Id: I2ed8c767274ba07764f03e8d4913041168e5755f
commit a44e73c322ba3f2c336f7cc4e1d63d3a74faa75d
Author: Andreas Huber <andih@google.com>
Date: Mon Oct 29 11:14:47 2012 -0700
Network traffic is now handled on a separate thread.
Audio and video are queued to ensure proper A/V interleaving.
Scheduled packet sends according to capture timestamps to reduce
send-jitter.
Change-Id: Ibd6357c1e663086cf87bec0a98f8e54dfdfaa0e5
related-to-bug: 7426218
Change-Id: Ia440129d656c35814abf18df06da50b73d5bb554
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* commit '2295ce2c6fcf07ffaa134440691a8cea5fe35b43':
Increase bitrate now that other issues have been resolved.
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Bug: 7426218
Change-Id: I67dfa1e4b85f326f355ad0ec5b6c699e87b45564
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* commit 'e815552c9cb17be79cb3b7eb6fbdfa9ad81c861b':
Disable silence suppression (causes issues with dongles).
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Bug: 7426218
Change-Id: Ie1517a8017bae1f9a9b6c224cd3170dfcc5fb941
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jb-mr1-dev
* commit 'ac1a450a76c2f42f66c876200cb724d1e93ef4c7':
Don't attempt to seek audio and video separately.
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The widevine extractor doesn't deal too well with that...
Change-Id: Iadfeede4fe0c086af788c5639782854e4fbb98ff
related-to-bug: 7262386
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jb-mr1-dev
* commit '5f6f5b7d6be5d7882102bf0fc3c691ef8fbe03a1':
stagefright: discard audio at start of capture
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The new camcorder start sound is longer than previous one and we
must discard more audio when capture starts.
Ideally, camcorder should use synchronous record start.
Bug 7394330.
Change-Id: I219b4e231aba706776dc7ccc4f1c996eaf22f61a
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* commit '0328ec08dc1e90caa2a9e0c4e107d8ddaa74af20':
WFD use of HDCP is now optional.
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if the sink supports it we'll use HDCP (and fail if necessary), if it doesn't
we won't. If an HDCP session is established we'll tell our observer that
the connection is secure, otherwise we don't.
Change-Id: I7cbef384f2cf0a6ac65801c581eea227b9ef4c46
related-to-bug: 7368436
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* commit 'c3ed4a1c3ea71681c71de48c206a899e702864bd':
AAC DRC reference level for mobile is -16dB
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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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* commit '3c9e53c6ea2b01506addd4fb1164e6a9bbe6e16f':
Fix valgrind issues
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