* qatar/master: (22 commits)
wma: Clip WMA1 and WMA2 frame length to 11 bits.
movenc: Don't require frame_size to be set for modes other than mov
doc: Update APIchanges with info on muxer flushing
movenc: Reindent a block
tools: Remove some unnecessary #undefs.
rv20: prevent calling ff_h263_decode_mba() with unset height/width
tools: K&R reformatting cosmetics
Ignore generated aviocat and ismindex tools.
build: Automatically include architecture-specific library Makefile snippets.
indeo5: prevent null pointer dereference on broken files
pktdumper: Use usleep instead of sleep
cosmetics: Remove some unnecessary block braces.
Drop unnecessary prefix from *sink* variable and struct names.
Add a tool for creating smooth streaming manifests
movdec: Calculate an average bit rate for fragmented streams, too
movenc: Write the sample rate instead of time scale in the stsd atom
movenc: Add a separate ismv/isma (smooth streaming) muxer
movenc: Allow the caller to decide on fragmentation
libavformat: Add a flag for muxers that support write_packet(NULL) for flushing
movenc: Add support for writing fragmented mov files
...
Conflicts:
Changelog
cmdutils.c
cmdutils.h
doc/APIchanges
ffmpeg.c
ffplay.c
libavfilter/Makefile
libavformat/Makefile
libavformat/avformat.h
libavformat/movenc.c
libavformat/movenc.h
libavformat/version.h
tools/graph2dot.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (23 commits)
aacenc: Fix identification padding when the bitstream is already aligned.
aacenc: Write correct length for long identification strings.
aud: remove unneeded field, audio_stream_index from context
aud: fix time stamp calculation for ADPCM IMA WS
aud: simplify header parsing
aud: set pts_wrap_bits to 64.
cosmetics: indentation
aud: support Westwood SND1 audio in AUD files.
adpcm_ima_ws: fix stereo decoding
avcodec: add a new codec_id for CRYO APC IMA ADPCM.
vqa: remove unused context fields, audio_samplerate and audio_bits
vqa: clean up audio header parsing
vqa: set time base to frame rate as coded in the header.
vqa: set packet duration.
vqa: use 1/sample_rate as the audio stream time base
vqa: set stream start_time to 0.
lavc: postpone the removal of AVCodecContext.request_channels.
lavf: postpone removing av_close_input_file().
lavc: postpone removing old audio encoding and decoding API
avplay: remove the -er option.
...
Conflicts:
Changelog
libavcodec/version.h
libavdevice/v4l.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtpdec: Use our own SSRC in the SDES field when sending RRs
Finalize changelog for 0.8 Release
Prepare for 0.8 Release
threads: change the default for threads back to 1
threads: update slice_count and slice_offset from user context
aviocat: Remove useless includes
doc/APIChanges: fill in missing dates and hashes
Revert "avserver: fix build after the next bump."
mpegaudiodec: switch error detection check to AV_EF_BUFFER
lavf: rename fer option and document resulting (f_)err_detect options
lavc: rename err_filter option to err_detect and document it
mpegvideo: fix invalid memory access for small video dimensions
movenc: Reorder entries in the MOVIentry struct, for tigheter packing
rtsp: Remove extern declarations for variables that don't exist
aviocat: Flush the output before closing
Conflicts:
Changelog
RELEASE
libavcodec/mpegaudiodec.c
libavcodec/pthread.c
libavformat/options.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also change synchronize_audio to only calculate the wanted number of samples
instead of doing the actual synchronization, and make swr_convert handle the
sample addition or deletion.
This new method replaces the old buggy way which simply deleted or
multiplied samples.
Signed-off-by: Marton Balint <cus@passwd.hu>
This allows to get out of ffplay if it or SDL got stuck.
This for example happens when the audio driver is playing something
else and doesnt support mixing multiple sources.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Marton Balint <cus@passwd.hu>
* qatar/master:
avconv: make -frames work for all types of streams, not just video.
bfi: K&R cosmetics
bgmc: K&R cleanup
rawdec: Set start_time to 0 for raw audio files.
Detect 'yuv2' as rawvideo also in avi.
rawdec: propagate pict_type information to the output frame
rawdec: Support more QT 1bpp rawvideo files.
avconv: free bitstream filters
threads: limit the number of automatic threads to MAX_AUTO_THREADS
avplay: K&R cleanup
fate: use rgb24 as output format for dfa tests
threads: set thread_count to 1 when thread support is disabled
threads: introduce CODEC_CAP_AUTO_THREADS and add it to libx264
Conflicts:
ffplay.c
libavcodec/avcodec.h
libavcodec/pthread.c
libavcodec/version.h
tests/ref/fate/dfa1
tests/ref/fate/dfa10
tests/ref/fate/dfa11
tests/ref/fate/dfa2
tests/ref/fate/dfa3
tests/ref/fate/dfa4
tests/ref/fate/dfa5
tests/ref/fate/dfa6
tests/ref/fate/dfa7
tests/ref/fate/dfa8
tests/ref/fate/dfa9
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
APIchanges: fill in revision for AVFrame.age deprecation
avcodec: deprecate AVFrame.age
4xm: remove unneeded check for remaining unused data.
lavf: force threads to 1 in avformat_find_stream_info()
swscale: fix overflows in vertical scaling at top/bottom edges.
lavf: add OpenMG audio muxer.
omadec: split data that will be used in the muxer to a separate file.
lavf: rename oma.c -> omadec.c
tmv decoder: set correct pix_fmt
Conflicts:
Changelog
doc/APIchanges
libavcodec/mpegvideo.c
libavcodec/version.h
libavformat/oma.c
libavformat/version.h
libswscale/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
x86: cabac: replace explicit memory references with "m" operands
avplay: don't request a stereo downmix
wmapro: use av_float2int()
lavc: avoid invalid memcpy() in avcodec_default_release_buffer()
lavu: replace int/float punning functions
lavfi: install libavfilter/vsrc_buffer.h
Remove extraneous semicolons
sdp: Restore the original mp4 format h264 extradata if converted
rtpenc: Add support for mp4 format h264
rtpenc: Simplify code by introducing a separate end pointer
movenc: Use the actual converted sample for RTP hinting
Fix a bunch of common typos.
Conflicts:
doc/developer.texi
doc/eval.texi
doc/filters.texi
doc/protocols.texi
ffmpeg.c
ffplay.c
libavcodec/mpegvideo.h
libavcodec/x86/cabac.h
libavfilter/Makefile
libavformat/avformat.h
libavformat/cafdec.c
libavformat/flvdec.c
libavformat/flvenc.c
libavformat/gxfenc.c
libavformat/img2.c
libavformat/movenc.c
libavformat/mpegts.c
libavformat/rtpenc_h264.c
libavformat/utils.c
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mov: Don't av_malloc(0).
avconv: only allocate 1 AVFrame per input stream
avconv: fix memleaks due to not freeing the AVFrame for audio
h264-fate: remove -strict 1 except where necessary (mr4/5-tandberg).
misc Doxygen markup improvements
doxygen: eliminate Qt-style doxygen syntax
g722: Add a regression test for muxing/demuxing in wav
g722: Change bits per sample to 4
g722dec: Signal skipping the lower bits via AVOptions instead of bits_per_coded_sample
api-example: update to use avcodec_decode_audio4()
avplay: use avcodec_decode_audio4()
avplay: use a separate buffer for playing silence
avformat: use avcodec_decode_audio4() in avformat_find_stream_info()
avconv: use avcodec_decode_audio4() instead of avcodec_decode_audio3()
mov: Allow empty stts atom.
doc: document preferred Doxygen syntax and make patcheck detect it
Conflicts:
avconv.c
ffplay.c
libavcodec/mlpdec.c
libavcodec/version.h
libavformat/mov.c
tests/codec-regression.sh
tests/fate/h264.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
configure: add check for w32threads to enable it automatically
rtmp: do not hardcode invoke numbers
cinepack: return non-generic errors
fate-lavf-ts: use -mpegts_transport_stream_id option.
Add an APIchanges entry and a minor bump for avio changes.
avio: Mark the old interrupt callback mechanism as deprecated
avplay: Set the new interrupt callback
avconv: Set new interrupt callbacks for all AVFormatContexts, use avio_open2() everywhere
cinepak: remove redundant coordinate checks
cinepak: check strip_size
cinepak, simplify, use AV_RB24()
cinepak: simplify, use FFMIN()
cinepak: Fix division by zero, ask for sample if encoded_buf_size is 0
applehttp: Fix seeking in streams not starting at DTS=0
http: Don't use the normal http proxy mechanism for https
tls: Handle connection via a http proxy
http: Reorder two code blocks
http: Add a new protocol for opening connections via http proxies
http: Split out the non-chunked buffer reading part from http_read
segafilm: add support for raw videos
...
Conflicts:
avconv.c
configure
doc/APIchanges
libavcodec/cinepak.c
libavformat/applehttp.c
libavformat/version.h
tests/lavf-regression.sh
tests/ref/lavf/ts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
tls: Use ERR_get_error() in do_tls_poll
indeo3: Fix a fencepost error.
mxfdec: Fix comparison of unsigned expression < 0.
mpegts: set stream id on just created stream, not an unrelated variable
ra288: return error if input buffer is too small
ra288: utilize DSPContext.vector_fmul()
ra288: use memcpy() to copy decoded samples to output
mace: only calculate output buffer size once
Remove redundant filename self-references inside files.
indeo3data: add missing config.h #include for HAVE_BIGENDIAN
x86: drop pointless ARCH_X86 #ifdef from files in x86 subdirectory
avplay: reset rdft when closing stream.
doc/git-howto: expand format-patch and send-email notes.
lavf: expand doxy for some AVFormatContext fields.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (23 commits)
x86inc: use sse versions of common macros instead of sse2 when applicable
doc/APIchanges: add missing dates and hashes
lavf: don't return from void av_update_cur_dts()
Changelog: add more entries.
Changelog: update ffmpeg/avconv incompatibility list.
avconv: remove some redundant temporary variables.
avconv: fix broken indentation
avconv: move copy_initial_nonkeyframes to the options context.
avconv: use file:stream instead of file.stream in log messages.
doc/avconv: elaborate on basic functionality.
doc/avconv: -sample_fmts, not -help sample_fmts prints the sample formats
openssl: Only use CRYPTO_set_id_callback on OpenSSL < 1.0.0
Call avformat_network_init/deinit in the programs
Remove leftover includes of strings.h
avutil: Don't allow using strcasecmp/strncasecmp
Replace all usage of strcasecmp/strncasecmp
avstring: Add locale independent implementations of strcasecmp/strncasecmp
avstring: Add locale independent implementations of toupper/tolower
cosmetics: insert some spaces in explicit enum value assignments
move 8SVX audio codecs to the audio codec list part on the next bump
...
Conflicts:
avprobe.c
doc/APIchanges
ffplay.c
ffserver.c
libavcodec/avcodec.h
libavdevice/bktr.c
libavdevice/v4l.c
libavdevice/v4l2.c
libavformat/matroskaenc.c
libavformat/wtv.c
libavutil/avstring.c
libavutil/avstring.h
libavutil/avutil.h
libswscale/x86/swscale_template.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This patch reimplements early frame drop, it is now based on the current
difference between the master clock and the video clock, and the pts of the
current and the last displayed (or skipped) frame. If the frame to be added to
the queue is late after decoding, then we drop it early because later we would
drop it anyway (unless it is the only frame in the picture queue).
The current approach has only one downside that I know of, it does not handle
well when the filters are changing significantly the pts of the frames, because
we compare pts values from filtered and unfiltered frames.
We also start using the pictq_mutex to ensure consistent video_current_pts,
video_current_pts_drift, frame_last_pts, frame_last_dropped_pts and
frame_last_dropped_pos values.
Signed-off-by: Marton Balint <cus@passwd.hu>
* qatar/master:
id3v2: fix doxy comment - 'machine byte order' makes no sense on char arrays
VC1: restore mistakenly removed code
twinvq: check output buffer size before decoding
twinvq: return an error when the packet size is too small
lavf: export some forgotten symbols with non-av prefixes.
swscale: update altivec yuv2planeX asm to new per-plane API.
swscale: make yuv2yuvX_10_sse2/avx 8/9/16-bits aware.
yuv2planeX10 SIMD
swscale: decide whether to use yuv2plane1/X on a per-plane basis.
swscale: reintroduce full precision in 16-bit output.
Split up yuv2yuvX functions
Split out yuv2yuv1 luma and chroma in order to make them generic DSP functions
lavc: replace references to deprecated AVCodecContext.error_recognition to use AVCodecContext.err_recognition
lavc: translate non-flag-based er options into flag-based ef options at codec open
add -err_filter AVOptions to access flag-based error recognition
h264_weight: initialize "height" function argument properly.
presets: spelling error in libvpx 1080p50_60
avplay: fix fullscreen behaviour with SDL 1.2.14 on Mac OS X
Conflicts:
ffplay.c
libavformat/libavformat.v
libswscale/swscale.c
libswscale/x86/swscale_template.c
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fixes missing blue channel when switching from/to fullscren on OSX and libsdl
1.2.14. Fixes issue 548. Thanks for Jean First for the original patch and
for testing.
Signed-off-by: Marton Balint <cus@passwd.hu>
If the picture queue is empty, or when the calculated delay is 0, frame_timer
is not increased but we are still displaying the old frame. When we eventually
get a frame, where the computed delay is positive, so we don't need to drop any
more frames, then it is best to update frame_timer to be as near as the current
time as it can.
This way we dont't have to wait several frames to add the necesarry delays to
frame_timer to reach current time, therefore there are no extra frame drops
after reaching a positive delay.
Signed-off-by: Marton Balint <cus@passwd.hu>
The current impementation of early frame drops (dropping frames before adding
them to the picture queue) has multiple problems:
Even after gettin A-V sync, the frame droping continues until
VideoState->skip_frames reaches 1, which can take a lot of time causing useless
additional frame drops and bad AV-sync. This issue can be easily triggered with
for example changing the audio stream.
Also video_refresh currenly does not handle early skipped frames in every case,
for example if we skip a frame, then the last frame duration calculation will
compute the duration of the sum of the skipped frame and the duration of the
frame before that, and in compute_target_delay we may multiply this unusually
big delay.
Signed-off-by: Marton Balint <cus@passwd.hu>
Since target clock is based on the current A-V delay, it is better calculate it
when we actually need it rather than when we put a picture in the picture
queue.
The patch also makes a code a bit more readable by renaming some delay
variables to duration, and converting compute_target_time to a delay
calculating function which does not modify the state. Factoring out the
iteration of the pictq to standalone function is also done in this patch.
Signed-off-by: Marton Balint <cus@passwd.hu>
