Previously, we returned any error code except AVERROR_EOF to the
caller - only if AVERROR_EOF or 0 was returned, we proceeded to
the next segment.
With some setups of web servers, using Connection: close in https
and GnuTLS, we don't get a clean error code at the end of segments.
In those cases, just proceed to the next segment.
Tested-by: Antti Seppälä <a.seppala@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
OpenSSL returns 0 when the peer has closed the connection. GnuTLS
doesn't return that though, but returns
GNUTLS_E_UNEXPECTED_PACKET_LENGTH if the connection simply is closed
without a clean close notify packet.
Tested-by: Antti Seppälä <a.seppala@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
proresdsp: port x86 assembly to cpuflags.
lavr: x86: improve non-SSE4 version of S16_TO_S32_SX macro
lavfi: better channel layout negotiation
alac: check for truncated packets
alac: reverse lpc coeff order, simplify filter
lavr: add x86-optimized mixing functions
x86: add support for fmaddps fma4 instruction with abstraction to avx/sse
tscc2: fix typo in array index
build: use COMPILE template for HOSTOBJS
build: do full flag handling for all compiler-type tools
eval: fix printing of NaN in eval fate test.
build: Rename aandct component to more descriptive aandcttables
mpegaudio: bury inline asm under HAVE_INLINE_ASM.
x86inc: automatically insert vzeroupper for YMM functions.
rtmp: Check the buffer length of ping packets
rtmp: Allow having more unknown data at the end of a chunk size packet without failing
rtmp: Prevent reading outside of an allocate buffer when receiving server bandwidth packets
Conflicts:
Makefile
configure
libavcodec/x86/proresdsp.asm
libavutil/eval.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Without this fix the last sample was missing from the packet.
Signed-off-by: Marton Balint <cus@passwd.hu>
Reviewed-by: Matthieu Bouron <matthieu.bouron@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
libopenjpeg: support YUV and deep RGB pixel formats
Fix typo in v410 decoder.
vf_yadif: unset cur_buf on the input link.
vf_overlay: ensure the overlay frame does not get leaked.
vf_overlay: prevent premature freeing of cur_buf
Support urlencoded http authentication credentials
rtmp: Return an error when the client bandwidth is incorrect
rtmp: Return proper error code in handle_server_bw
rtmp: Return proper error code in handle_client_bw
rtmp: Return proper error codes in handle_chunk_size
lavr: x86: add missing vzeroupper in ff_mix_1_to_2_fltp_flt()
vp8: Replace x*155/100 by x*101581>>16.
vp3: don't use calls to inline asm in yasm code.
x86/dsputil: put inline asm under HAVE_INLINE_ASM.
dsputil_mmx: fix incorrect assembly code
rtmp: Factorize the code by adding handle_invoke
rtmp: Factorize the code by adding handle_chunk_size
rtmp: Factorize the code by adding handle_ping
rtmp: Factorize the code by adding handle_client_bw
rtmp: Factorize the code by adding handle_server_bw
Conflicts:
libavcodec/libopenjpegdec.c
libavcodec/x86/dsputil_mmx.c
libavfilter/vf_overlay.c
libavformat/Makefile
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
By moving it to a later point relative and unknown timestamps
are more likely to have been corrected
similar patch reviewed-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Conflicts:
libavformat/utils.c
It should be possible to specify usernames in http requests containing
urlencoded characters. This patch adds support for decoding the auth
strings.
Signed-off-by: Antti Seppälä <a.seppala@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
rtmp: Add a new option 'rtmp_pageurl'
doc: Update the description of the rtmp_tcurl option
rtmp: Make the description of the rtmp_tcurl option more generic
libfdk-aacenc: add LATM/LOAS encapsulation support
sctp: add port missing error message
tcp: add port missing error message
avfilter: Fix printf format string conversion specifier
Conflicts:
libavcodec/version.h
libavfilter/avfilter.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Without this patch a user a bit absent-minded may not notice that
the connection doesn't work because the port is missing.
Signed-off-by: Martin Storsjö <martin@martin.st>
Without this patch a user a bit absent-minded may not notice that
the connection doesn't work because the port is missing.
Signed-off-by: Martin Storsjö <martin@martin.st>
The native decoder and MPlayer's binary decoder only need the
APRG atom, QuickTime at least requires also the ARES atom and
four additional 0 bytes padding at the end of stsd.
Attached patch (together with demuxing patch) allows WMP/msacm G723.1 codec decode files encoded by FFmpeg.
Tested with both 6400 and 5333 mode
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Add credit/copyright to librtmp authors for parts of the RTMPE code
rtmp: Move the CONFIG_ condition into the if conditions
aac: Mention abbreviation as well in long_name
build: Skip compiling rtmpdh.h if ffrtmpcrypt protocol is not enabled
doc: Add Git configuration section
configure: Add a dependency on https for rtmpts
rtp: Only choose static payload types if the sample rate and channels are right
Conflicts:
doc/git-howto.texi
libavformat/rtmpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Attached patch fixes remuxing of G723.1 in wav, so the output is playable by WMP.
(It's still not enough for encoding - probably some extradata should be added to the output file
to make it playable by WMP/win codec)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This makes sure these calls are removed by dead code elimination
even if optimization is disabled. This fixes building without
crypto libraries without optimization.
Signed-off-by: Martin Storsjö <martin@martin.st>
If using a different sample rate or number of channels, use a dynamic
payload type instead, where the parameters are passed in the SDP.
G722 is a special case where the normal rules don't apply.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
v410dec: Implement explode mode support
zerocodec: fix direct rendering.
wav: init st to NULL to avoid a false-positive warning.
wavpack: set bits_per_raw_sample for S32 samples to properly identify 24-bit
h264: refactor NAL decode loop
RTMPTE protocol support
RTMPE protocol support
rtmp: Add ff_rtmp_calc_digest_pos()
rtmp: Rename rtmp_calc_digest to ff_rtmp_calc_digest and make it global
swscale: add missing HAVE_INLINE_ASM check.
lavfi: place x86 inline assembly under HAVE_INLINE_ASM.
vc1: Add a test for interlaced field pictures
swscale: Mark all init functions as av_cold
swscale: x86: Drop pointless _mmx suffix from filenames
lavf: use conditional notation for default codec in muxer declarations.
swscale: place inline assembly bilinear scaler under HAVE_INLINE_ASM.
dsputil: ppc: cosmetics: pretty-print
dsputil: x86: add SHUFFLE_MASK_W macro
configure: respect CC_O setting in check_cc
Conflicts:
Changelog
configure
libavcodec/v410dec.c
libavcodec/zerocodec.c
libavformat/asfenc.c
libavformat/version.h
libswscale/utils.c
libswscale/x86/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If st is NULL, it means no 'fmt ' tag is found, but 'data' tag (which
needs a previous 'fmt ' tag to be parsed correctly and st initialized)
check will make sure st is never dereferenced in that case.
Fixes warning:
libavformat/wav.c: In function ‘wav_read_header’:
libavformat/wav.c:499:44: warning: ‘st’ may be used uninitialized in this function [-Wmaybe-uninitialized]
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPE protocol implementation uses ffrtmpcrypt:// as an alternative to the
tcp:// protocol. This allows moving most of the lower level logic out
from the higher level generic rtmp code.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
FATE: fix the asyncts test
build: Drop gcc-specific warning flag from header compilation rule
FATE: add a test for the asyncts audio filter.
matroskadec: return more correct error code on read error.
buffersrc: check ff_get_audio_buffer() for errors.
lavfi: check all ff_get_video_buffer() calls for errors.
lavfi: check all avfilter_ref_buffer() calls for errors.
vf_select: avoid an unnecessary avfilter_ref_buffer().
buffersrc: avoid creating unnecessary buffer reference
lavfi: use avfilter_unref_bufferp() where appropriate.
vf_fps: add more error checks.
vf_fps: fix a memleak on malloc failure.
lavfi: check all ff_start_frame/draw_slice/end_frame calls for errors
lavfi: add error handling to end_frame().
lavfi: add error handling to draw_slice().
lavfi: add error handling to start_frame().
Conflicts:
Makefile
ffplay.c
libavfilter/buffersrc.c
libavfilter/vf_boxblur.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_frei0r.c
libavfilter/vf_hflip.c
libavfilter/vf_overlay.c
libavfilter/vf_pad.c
libavfilter/vf_scale.c
libavfilter/video.c
libavfilter/vsrc_color.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
x86: swscale: Place inline assembly code under appropriate #ifdefs
rtsp: remove terminal comma in FF_RTP_FLAG_OPTS macro.
configure: Remove redundant RTMPT/RTMPTS dependencies
configure: add filtering of host cflags/ldflags
configure: initialise all flag filters at the same place
configure: add filtering of linker flags
configure: name some variables more consistently
configure: remove filter_cppflags
configure: set icc_version where it is needed
mpegenc: remove disabled code
Conflicts:
configure
libavformat/movenc.c
libswscale/x86/swscale_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This happens with for example mplayer.
Fixing it in ffmpeg allows new ffmpeg to be compiled with older mplayer
which would not be possible if the fix was just in mplayer.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If skip_samples is set and timestamps are synthesized using durations,
make them start at -skip_samples (rescaled) instead of 0,
so that the timestamp of the first undiscarded sample is 0.
* qatar/master: (38 commits)
alac: cosmetics: general pretty-printing and comment clean up
alac: calculate buffer size outside the loop in allocate_buffers()
alac: change some data types to plain int
alac: cosmetics: rename some variables and function names
alac: multi-channel decoding support
alac: split element parsing into a separate function
alac: support a read sample size of up to 32
alac: output in planar sample format
alac: add 32-bit decoding support
alac: simplify channel interleaving
alac: use AVPacket fields directly in alac_decode_frame()
alac: fix check for valid max_samples_per_frame
alac: use get_sbits() to read LPC coefficients instead of casting
alac: move the current samples per frame to the ALACContext
alac: avoid using a double-negative when checking if the frame is compressed
alac: factor out output_size check in predictor_decompress_fir_adapt()
alac: factor out loading of next decoded sample in LPC prediction
alac: use index into buffer_out instead of incrementing the pointer
alac: simplify lpc coefficient adaptation
alac: reduce the number of local variables needed in lpc prediction
...
Conflicts:
libavcodec/alac.c
libavformat/cafdec.c
libavformat/mov.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>