- 01 Mar, 2015 12 commits
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Michael Niedermayer authored
The code has undefined behavior and makes no difference when optimizations are enabled. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Gilles Chanteperdrix authored
When precision is fixed and volume is 0, filter_frame does not perform any operation on the output buffer. This works if the output buffer has been allocated and zeroed with ff_get_audio_buffer but not if the input buffer is used as output buffer. Fix this by not using the input buffer as output buffer if precision is fixed and volume is 0. Signed-off-by: Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit '4f6cd883': rtpenc: Don't set max_frames_per_packet based on the packet frame size or frame rate Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit 'bde2bba4': rtpenc: Restructure if statements in packetizers to simplify adding more conditions Conflicts: libavformat/rtpenc_xiph.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit 'd4c7fc02': rtpenc: Skip redundant initialization Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit 'f8c01257': rtpenc: Always do the default initialization regardless of codecs Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit '11edeaea': rtpenc_xiph: Don't exclude headers from max_payload_size Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit '7c1e2e64': rtpenc_xiph: Use AV_WB16 instead of manual bitshifts Conflicts: libavformat/rtpenc_xiph.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit 'd16c8d28': rtpenc_aac: Use AV_WB16 instead of manual bitshifts Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit '9c9b0218': rtpenc_aac: Merge a definition with a declaration Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit '1fc64e2e': rtpenc: Write conditional statements on separate lines Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit '0662440b': rtpenc_aac: Set a default value for max_frames_per_packet at init Merged-by: Michael Niedermayer <michaelni@gmx.at>
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- 28 Feb, 2015 28 commits
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Michael Niedermayer authored
* commit '12b34599': rtpenc_amr: Use s->num_frames instead of s->buf_ptr - s->buf Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit '98563953': rtpenc_aac: Fix sending fragmented frames Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit '990e4a66': Add a QSV decoding example. Conflicts: configure doc/Makefile Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit 'ea1d0b7e': avcodec/utils: use correct printf specifier in ff_set_sar See: 732c3ebfMerged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit '802987f8': x11grab: Unbreak building Conflicts: libavdevice/x11grab.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit '71f1ad37': lavc: do not compile fmtconvert unconditionally Conflicts: configure libavcodec/ppc/Makefile libavcodec/x86/Makefile Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit 'd74a8cb7': fmtconvert: drop unused functions Conflicts: libavcodec/arm/fmtconvert_vfp_armv6.S libavcodec/x86/fmtconvert.asm libavcodec/x86/fmtconvert_init.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit 'ee964145': lavc: remove unused traces of fmtconvert usage Conflicts: libavcodec/aac.h libavcodec/aacdec.c libavcodec/atrac3.c libavcodec/vorbisdec.c libavcodec/wma.c libavcodec/wma.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit 'b9d2d684': tls: Pass AVOptions dictionaries through to the chained protocol Conflicts: libavformat/tls.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit 'e14f98c6': tcp: Clarify the units for the timeout avoptions Conflicts: libavformat/tcp.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
* commit 'c86d8aed': avio: Rename avclass symbols relating to avio Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Andreas Cadhalpun authored
flags is later written with avio_w8 and if it doesn't fit in one byte it triggers an av_assert2. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Andreas Cadhalpun authored
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
These could trigger assert failures previously Found-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Martin Storsjö authored
Instead check the timestamps while muxing, to avoid buffering a too long timestamp range into one single packet. This makes the AMR and AAC packetization slightly less efficient, since we set a possibly unnecessarily high max_frames_per_packet. (These packetizers end up doing a memmove of the TOC bytes if sending a packet before max_frames_per_packet is achieved, and we end up setting max_frames_per_packet to a value that should be high enough for most uses.) All packetizers that use max_frames_per_packet now set it either to a default value, or to a value calculated based on other parameters, so none of them rely on the previous default setting. For iLBC, copy one frame at a time, to allow checking the timestamp range for each of them - basically doing potentially multiple loops to simplify the code instead of trying to calculate the number of frames to buffer while honoring s1->max_delay. This is in preparation for reducing the coupling between libavformat and libavcodec, by not having the muxers use the encoder field frame_size (which may not be available during e.g. stream copy). Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
Factorize out the s->num_frames check at the start of the if statements, simplifying adding more alternative causes for sending the buffered frames. Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
This avoids having to jump to the defaultcase in the switch. Manually override the stream time base back to 90 kHz for the few audio codecs that don't use the sample rate as time base (mp2, mp3). Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
This makes things more consistent by using the variable in the same way as in all other packetizers. Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
This makes the code slightly more readable and understandable. Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
Intentionally keeping some conditional statements on single lines in rtpenc_h263.c. Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
This avoids having to conditionally set the default within the packetizer function. Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
This doesn't fix any bug, but makes the code simpler for later patches, and more straightforward to read as is. Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
After sending a fragmented frame, len (s->buf_ptr - s->buf) isn't zero, while s->num_frames is zero as intended. Using s->num_frames makes it work as intended, and is less convoluted than keeping track of (resetting) s->buf_ptr. This avoids sending stray data after sending a fragmented aac packet. CC: libav-stable@libav.org Signed-off-by: Martin Storsjö <martin@martin.st>
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Anton Khirnov authored
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Andreas Cadhalpun authored
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com> Signed-off-by: Anton Khirnov <anton@khirnov.net>
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