Both stream_id and stream_identifier are used in this file,
and have different meanings. The latter comes from the
stream_identifier_descriptor.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Without this some operations might overflow (undefined behavior)
even though the index adding loop would never execute
No testcase known
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This massively reduces the detection of random data as low score mp3
It may improve security by making it harder to read non multimedia data
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This new optional flag makes it easier to deal with mpegts
samples where the PMT is updated and elementary streams move
to different PIDs in the middle of playback.
Previously, new AVStreams were created per PID, and it was up
to the user to figure out which streams had migrated to a new PID
(by iterating over the list of AVProgram and making guesses), and
switch seamlessly to the new AVStream during playback.
Transcoding or remuxing these streams with ffmpeg on the CLI was
also quite painful, and the user would need to extract each set
of PIDs into a separate file and then stitch them back together.
With this new option, the mpegts demuxer will automatically detect
PMT changes and feed data from the new PID to the original AVStream
that was created for the orignal PID. For mpegts samples with
stream_identifier_descriptor available, the unique ID is used to
merge PIDs together. If the stream id is not available, the demuxer
attempts to map PIDs based on their position within the PMT.
With this change, I am able to playback and transcode/remux these
two samples which previously caused issues:
https://tmm1.s3.amazonaws.com/pmt-version-change.tshttps://kuroko.fushizen.eu/videos/pid_switch_sample.ts
I also have another longer sample in which the PMT changes
repeatedly and ES streams move to different pids three times
during playback:
https://tmm1.s3.amazonaws.com/multiple-pmt-change.ts
Demuxing this sample with the new option shows several new log
messages as the PMT changes are handled:
[mpegts] detected PMT change (program=1, version=3/6, pcr_pid=0xf98/0xfb7)
[mpegts] re-using existing video stream 0 (pid=0xf98) for new pid=0xfb7
[mpegts] re-using existing audio stream 1 (pid=0xf99) for new pid=0xfb8
[mpegts] re-using existing audio stream 2 (pid=0xf9a) for new pid=0xfb9
[mpegts] detected PMT change (program=1, version=6/3, pcr_pid=0xfb7/0xf98)
[mpegts] detected PMT change (program=1, version=3/4, pcr_pid=0xf98/0xf9b)
[mpegts] re-using existing video stream 0 (pid=0xf98) for new pid=0xf9b
[mpegts] re-using existing audio stream 1 (pid=0xf99) for new pid=0xf9c
[mpegts] re-using existing audio stream 2 (pid=0xf9a) for new pid=0xf9d
[mpegts] detected PMT change (program=1, version=4/5, pcr_pid=0xf9b/0xfa9)
[mpegts] re-using existing video stream 0 (pid=0xf98) for new pid=0xfa9
[mpegts] re-using existing audio stream 1 (pid=0xf99) for new pid=0xfaa
[mpegts] re-using existing audio stream 2 (pid=0xf9a) for new pid=0xfab
[mpegts] detected PMT change (program=1, version=5/6, pcr_pid=0xfa9/0xfb7)
Signed-off-by: Aman Gupta <aman@tmm1.net>
With these fields, the user has enough information to
detect PMT changes and switch to new streams when the PMT
is updated with new ES pids.
To do so, the user would monitor the AVProgram they're interested
in for changes to pmt_version. If the version changes, they would
iterate over the program's streams to find new streams added with
the updated version number.
If new versions of streams are found, then the user would first try
to replace existing streams where stream_identifier matched.
If stream_identifier is not available, then the user would compare
pmt_stream_idx instead to replace the stream that was previously
at the same position within the PMT.
Signed-off-by: Aman Gupta <aman@tmm1.net>
These fields will allow the mpegts demuxer to expose details about
the PMT/program which created the AVProgram and its AVStreams.
In mpegts, a PMT which advertises streams has a version number
which can be incremented at any time. When the version changes,
the pids which correspond to each of it's streams can also change.
Since ffmpeg creates a new AVStream per pid by default, an API user
needs the ability to (a) detect when the PMT changed, and (b) tell
which AVStream were added to replace earlier streams.
This has been a long-standing issue with ffmpeg's handling of mpegts
streams with PMT changes, and I found two related patches in the wild
that attempt to solve the same problem:
The first is in MythTV's ffmpeg fork, where they added a
void (*streams_changed)(void*); to AVFormatContext and call it from
their fork of the mpegts demuxer whenever the PMT changes.
The second was proposed by XBMC in
https://ffmpeg.org/pipermail/ffmpeg-devel/2012-December/135036.html,
where they created a new AVMEDIA_TYPE_DATA stream with id=0 and
attempted to send packets to it whenever the PMT changed.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Some filtered mpegts streams may erroneously include PMTs for
programs that are not advertised in the PAT. This confuses ffmpeg
and most players because multiple audio/video streams are created
and it is unclear which ones actually contain data.
See for example https://tmm1.s3.amazonaws.com/unknown-pmts.ts
In this sample, the PAT advertises exactly one program. But the
pid it points to for the program's PMT contains PMTs for other
programs as well. This is because the broadcaster decided to
re-use the same pid for multiple program PMTs.
The hardware that filtered the original multi-program stream
into a single-program stream did so by rewriting the PAT to
contain only the program that was requested. But since it just
passed through the PMT pid referenced in the PAT, multiple PMTs
are still present for the other programs.
Before:
Input #0, mpegts, from 'unknown-pmts.ts':
Duration: 00:00:10.11, start: 80741.189700, bitrate: 9655 kb/s
Program 4
Stream #0:2[0x41]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv, bt709, progressive), 1280x720 [SAR 1:1 DAR 16:9], Closed Captions, 11063 kb/s, 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc
Stream #0:3[0x44](eng): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 384 kb/s
Stream #0:4[0x45](spa): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 128 kb/s
No Program
Stream #0:0[0x31]: Video: mpeg2video ([2][0][0][0] / 0x0002), none(tv), 90k tbr, 90k tbn, 90k tbc
Stream #0:1[0x34](eng): Audio: ac3 (AC-3 / 0x332D4341), 0 channels, fltp
Stream #0:5[0x51]: Video: mpeg2video ([2][0][0][0] / 0x0002), none, 90k tbr, 90k tbn
Stream #0:6[0x54](eng): Audio: ac3 (AC-3 / 0x332D4341), 0 channels
With skip_unknown_pmt=1:
Input #0, mpegts, from 'unknown-pmts.ts':
Duration: 00:00:10.11, start: 80741.189700, bitrate: 9655 kb/s
Program 4
Stream #0:0[0x41]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv, bt709, progressive), 1280x720 [SAR 1:1 DAR 16:9], Closed Captions, 11063 kb/s, 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc
Stream #0:1[0x44](eng): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 384 kb/s
Stream #0:2[0x45](spa): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 128 kb/s
Signed-off-by: Aman Gupta <aman@tmm1.net>
Parses the video_stream_descriptor (H.222 2.6.2) to look
for the still_picture_flag. This is exposed to the user
via a new AV_DISPOSITION_STILL_IMAGE.
See for example https://tmm1.s3.amazonaws.com/music-choice.ts,
whose video stream only updates every ~6 seconds.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
For seekable mpegts streams, duration is calculated from
pts by seeking to the end of the file for a pts and subtracting
the initial pts to compute a duration.
This can be expensive in terms of added latency during
probe, especially when streaming over a network. This new
option lets you skip the duration calculation, which is useful
when you don't care about the value and want to save some overhead.
This patch is particularly useful when dealing with live mpegts
streams. Normally such streams are not seekable, so durations
are not calculated. However in my case I am dealing with a seekable
live mpegts stream (networked access to a .ts file which is still
being appended to).
Signed-off-by: Aman Gupta <aman@tmm1.net>
My conversation from AVFormatContext->filename to AVFormatContext->url was
wrong in this case because get_chunk_filename uses filename as an output
buffer, and not as an input buffer.
Fixes ticket #7188.
Signed-off-by: Marton Balint <cus@passwd.hu>
Fixes PMT parsing in some mpegts streams which contain
multiple tables within the PMT pid. Previously, the parser
assumed only one table was present in each packet, and discarded
the rest of the section data after attempting to parse the first
table.
A similar issue was documented in the BeyondTV software[1], which
helped me diagnose the same bug in the ffmpeg mpegts demuxer. I also
tried DVBInspector, libdvbpsi's dvbinfo, and tstools' tsinfo to
help debug. The former two properly read PMTs with multiple tables,
whereas the last has the same bug as ffmpeg.
I've created a minimal sample[2] which contains the combined PMT.
Here's what ffmpeg probe shows before and after this patch:
Before:
Input #0, mpegts, from 'combined-pmt-tids.ts':
Duration: 00:00:01.08, start: 4932.966167, bitrate: 741 kb/s
Program 1
No Program
Stream #0:0[0xf9d]: Audio: ac3, 48000 Hz, mono, fltp, 96 kb/s
Stream #0:1[0xf9b]: Audio: mp3, 0 channels, fltp
Stream #0:2[0xf9c]: Unknown: none
After:
Input #0, mpegts, from 'combined-pmt-tids.ts':
Duration: 00:00:01.11, start: 4932.966167, bitrate: 718 kb/s
Program 1
Stream #0:0[0xf9b]: Video: mpeg2video ([2][0][0][0] / 0x0002), none(tv, top first), 29.97 fps, 29.97 tbr, 90k tbn, 90k tbc
Stream #0:1[0xf9c](eng): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 384 kb/s
Stream #0:2[0xf9d](spa): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, mono, fltp, 96 kb/s
With the patch, the PMT is parsed correctly so the streams are
created in the correct order, are associated with "Program 1",
and their codecs are set correctly.
[1] http://forums.snapstream.com/vb/showpost.php?p=343816&postcount=201
[2] https://s3.amazonaws.com/tmm1/combined-pmt-tids.ts
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This allows remuxing streams from one mpegts container to another,
without requiring avformat_find_stream_info() (or using `ffmpeg
-probesize 32` on the cli).
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
SMPTE 386M (D-10) lists 4 as value to be used
SMPTE 377-1-2009 says
"The definitions of 00h (coSiting) and 04h (Rec 601) are equivalent. The value of 04h is deprecated. New
MXF encoders shall use the value of 00h instead."
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Thomas Mundt <tmundt75@gmail.com>
Reviewed-by: Tomas Härdin <tjoppen@acc.umu.se>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This mimics the logic flow in all the other callbacks
(pat_cb, sdt_cb, m4sl_cb), and avoids calling skip_identical()
for non PMT_TID packets.
Since skip_identical modifies internal state like
MpegTSSectionFilter.last_ver, this change prevents unnecessary
reprocessing on some streams which contain multiple tables in
the PMT pid. This can be observed with streams from certain US
cable providers, which include both tid=0x2 and another unspecified
tid=0xc0.
Signed-off-by: Aman Gupta <aman@tmm1.net>
This can "demux" .vpy files. Autodetection of .vpy scripts is
intentionally not done, because it would be a major security issue. You
need to force the format, for example with "-f vapoursynth" for the
FFmpeg CLI tools.
Some minor code copied from other LGPL parts of FFmpeg.
I did not find a good way to test a few of the more obscure VS features,
like VFR nodes, compat pixel formats, or nodes with dynamic size/format
changes. These can be easily implemented on demand.
This code will print a warning if any user agent is set - even if the
API user used the proper non-deprecated "user_agent" option.
This change should not even break anything, because even if the user
sets the deprecated "user-agent" option, http.c copies it to the
"user_agent" option anyway.
If the API user doesn't set avg_frame_rate, matroskaenc will write the
current timebase as "default duration" for the video track. This makes
no sense, because the "default duration" implies the framerate of the
video. Since the timebase is forced to 1/1000, this will make the
resulting file claim 1000fps.
Drop it and don't write the element. It's optional, so it's better not
to write it if the framerate is unknown.
Strangely does not require FATE changes.
Enables one to test possibly nonstandard formats such as Opus or
FLAC in ISOBMFF, among other things.
This becomes much more useful if output segment format becomes an
option, or if the WebM segment feature gets removed.
It has not ever been working and has not been validated,
Additionally, mention that the segment file names should be changed
to end with webm instead of m4s, which is utilized for ISOBMFF
fragments.
The specs says that the the first color component in the color array is
not alpha, but simply 0.
Fixes 0 alpha of fate-suite/cvid/catfight-cvid-pal8-partial.mov
Signed-off-by: Marton Balint <cus@passwd.hu>
1. an audio component with an ISO_639_language_descriptor in the PMT with the
audio_type field set to 0x03
2. a supplementary_audio_descriptor with the editorial_classification field set
to 0x01
3. an ac-3_descriptor or an enhanced_ac-3_descriptor with a component_type field
with the service_type flags set to Visually Impaired
Tested-by: Łukasz Krzciuk <lkrzciuk@vewd.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The track's media duration from the mdhd atom takes precedence
over both the stts and elst atom for calculating and setting
the track's total duraion.
Technically, we shouldn't be using the stts atom at all for
calculating stream durations.
This fixes incorrect stream and final packet durations on files
with edit lists that are longer than the media duration.
The FATE changes are expected, and output is more correct (the
AAC frame is not 1028 samples).
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
The av_rc4_crypt() documentation allows src == dst.
Silences the following warning:
libavformat/rtmpcrypt.c:304:36: warning: passing argument 2 of 'av_rc4_crypt' discards 'const' qualifier from pointer target type
Reported-by: Reino Wijnsma
There is a separate muxer(webmdashenc.c) for supporting VP9+webm output in DASH.
Hence in this muxer we will focus on supporting VP9 in MP4
Have verified playout support of VP9+MP4 in Chrome and Firefox.
Should be useful for muxers that require values as defined in the
vpcc atom but don't need to write the atom itself.
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes a warning:
libavformat/dashdec.c:1900:65: warning: argument to 'sizeof' in 'memcpy' call is the same pointer type 'struct fragment *' as the destination; expected 'struct fragment' or an explicit length
This doesn't support saio atoms with more than one offset.
Signed-off-by: Jacob Trimble <modmaker@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Similar to 4c9c4fe8b2, but for durations. This fixes#7151, where
the report duration and bitrate on a mpegts stream is wildly off
due to the dvb_teletext stream's timings.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In write only mode, the TCP receive buffer's data keeps growing with
http response messages and the buffer eventually becomes full.
This results in zero tcp window size, which in turn causes unwanted
issues, like, terminated tcp connection. The issue is apparent when
http persistent connection is enabled in hls/dash live streaming use
cases. To overcome this issue, the logic here reads the buffer data
when a file transfer is completed, so that any accumulated data in
the recieve buffer gets flushed out.
reference hls support fmp4 file from draft-pantos-http-live-streaming-20
the spec describes version 7 of hls protocol
Suggested-by: Ronak <ronak2121@yahoo.com>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
This refactors get_cookies to simplify some code paths, specifically for
skipping logic in the while loop or exiting it. It also simplifies the logic
for appending additional values to *cookies by replacing strlen/malloc/snprintf
with one call av_asnprintf.
This refactor fixes a bug where the cookie_params AVDictionary would get leaked
if we failed to allocate a new buffer for writing to *cookies.
- Parse schm atom to get different encryption schemes.
- Allow senc atom to appear in track fragments.
- Allow 16-byte IVs.
- Allow constant IVs (specified in tenc).
- Allow only tenc to specify encryption (i.e. no senc/saiz/saio).
- Use sample descriptor to detect clear fragments.
This doesn't support:
- Different sample descriptor holding different encryption info.
- Only first sample descriptor can be encrypted.
- Encrypted sample groups (i.e. seig).
- Non-'cenc' encryption scheme when using -decryption_key.
Signed-off-by: Jacob Trimble <modmaker@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes https://trac.ffmpeg.org/ticket/2798
This makes movenc handle AV_DISPOSITION_ATTACHED_PIC and write
the associated pictures in iTunes cover atom. This corresponds
to how 'mov' demuxer parses and exposes the cover images when
reading.
Most of the existing track handling loops properly ignore
these 'virtual streams' as MOVTrack->entry is never incremented
for them. However, additional tests are added as needed to ignore
them.
Tested to produce valid output with:
ffmpeg -i movie.mp4 -i thumb.jpg -disposition✌️1 attached_pic \
-map 0 -map 1 -c copy movie-with-cover.mp4
The cover image is also copied correctly with:
ffmpeg -i movie-with-cover.mp4 -map 0 -c copy out.mp4
AtomicParseley says that the attached_pic stream is properly
not visible in the main tracks of the file.
Signed-off-by: Timo Teräs <timo.teras@iki.fi>
The logic is applicable only when use_template is enabled and use_timeline
is disabled. The logic monitors the flow of segment indexes. If a streams's
segment index value is not at the expected real time position, then
the logic corrects that index value.
Typically this logic is needed in live streaming use cases. The network
bandwidth fluctuations are common during long run streaming. Each
fluctuation can cause the segment indexes fall behind the expected real
time position. Without this logic, players will not be able to consume
the content, even after encoder's network condition comes back to
normal state.
availability time of Nth segment = availabilityStartTime + (N*segment duration) - availabilityTimeOffset.
This field helps to reduce the latency by about a segment duration in streaming mode.
@availabilityStartTime specifies the anchor for the computation of the earliest
availability time (in UTC) for any Segment in the Media Presentation.
As per this requirement, the @AvailabilityStartTime should be set to the
wallclock time at which the first frame of the first segment begins encoding.
But, it was getting set only when the first segment was completely ready. Making
the required correction in this patch. This correction is mainly needed to reduce
the latency in live streaming use cases.
Calling 'write_manifest' from 'write_header' was causing creation of
first MPD with invalid values. Ex: zero @duration param value. Also,
the manifest files (MPD or M3U8s) should be created when at-least
one media frame is ready for consumption.
When use_template is enabled and use_timeline is disabled, typically
it is required to generate the segments at the configured segment duration
rate on an average. This commit is particularly needed to handle the
segmentation when video frame rates are fractional like 29.97 or 59.94 fps.
There are use cases where average segment duration needs to be configured
and muxer is expected to maintain the average segment duration. So, using
the name 'min_seg_duration' will be misleading. So, changing the parameter
name to 'seg_duration', where it can be minimum segment duration or average
segment duration based on the use-case. The additional updates needed for
this functinality are made the sub-sequent patches of this patch series.
The HLSContext struct contains fields which duplicate the data stored in the
avio_opts field. This change removes those fields in favor of avio_opts, and
updates the code accordingly.
The original patch caused the buffer pointed to by new_cookies in open_url to be
leaked. The only thing that buffer is used for is to store the value until it
can be passed to av_dict_set. To fix the leak, v2 of the patch simply calls
av_dict_set with the AV_DICT_DONT_STRDUP_VAL flag, so that the dictionary takes
ownership of the memory instead of copying it again.
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Richard Shaffer <rshaffer@tunein.com>
The rw_timeout option is currently not applied when opening media playlist,
segment, or encryption key URLs. This can cause the HLS demuxer to block
indefinitely, even when the rw_timeout option has been specified. This change
simply enables carrying over the rw_timeout option when the demuxer opens these
URLs.
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Richard Shaffer <rshaffer@tunein.com>
The rw_timeout option is currently not applied when opening media playlist,
segment, or encryption key URLs. This can cause the HLS demuxer to block
indefinitely, even when the rw_timeout option has been specified. This change
simply enables carrying over the rw_timeout option when the demuxer opens these
URLs.
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Richard Shaffer <rshaffer@tunein.com>
This prevents creating potentially broken files, as both the AV1 and
the AV1 in ISOMBFF specs are unfinished.
Signed-off-by: James Almer <jamrial@gmail.com>
and refine the warning message when use both hls_key_info_file and hls_enc
Reviewed-by: Gyan Doshi <ffmpeg@gyani.pro>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
If the parser returns full frames, then the output pointer retured by
av_parser_parse2() is guaranteed to point to data contained in the
input packet's buffer.
Create a new reference to said buffer in that case, to avoid
unnecessary data copy when queueing the packet later in the function.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes: runtime error: signed integer overflow: 18133149658382192 - -9223090561878065151 cannot be represented in type 'long long'
Fixes: crbug 831552
Reported-by: Matt Wolenetz <wolenetz@google.com>
Reviewed-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: crbug 829153
Reported-by: Matt Wolenetz <wolenetz@google.com>
Reviewed-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In some cases, mainly working with multiprogram mpeg-ts containers as
input, it would be handy to select sub stream of a specific program by
their metadata.
This patch makes it possible to narrow the stream selection among
streams of the specified program by stream metadata.
Examples:
p:601:m:language:hun will select all sub streams of program with id 601
where sub streams have metadata key named 'language' with value 'hun'.
p:602:m:guide will select all sub streams of program with id 602 where
sub streams have metadata key named 'guide'.
Signed-off-by: Bela Bodecs <bodecsb@vivanet.hu>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When using hls_list_size with hls_flags delete_segments, currently
hls_list_size * 2 +- segments remain on disk. With this new option,
the amount of disk space used can be controlled by the user.
fix ticket: #7131
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Aman Gupta <aman@tmm1.net>
Fixes: crbug 822705
Reported-by: Matt Wolenetz <wolenetz@google.com>
Reviewed-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Currently when specifying the program id you can only decide to select
all stream of the specified program (e.g. p:103 will select all streams
of program 103) or narrow the selection to a specific stream sub index
(e.g. p:145:1 will select 2nd stream of program 145.) But you can not
specify like all audio streams of program 145 or 3rd video stream of
program 311.
In some case, mainly working with multiprogram mpeg-ts containers as
input, this feature would be handy.
This patch makes it possible to narrow the stream selection among
streams of the specified program by stream type and optionally its
index. Handled types: a, v, s, d.
Examples: p:601:a will select all audio streams of program 601,
p:603:a:1 will select 2nd audio streams of program 603,
p:604:v:0 will select first video stream of program 604.
Signed-off-by: Bela Bodecs <bodecsb@vivanet.hu>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Some ADTS streams can have multiple ID3 tags between frames. This
change parses all of them, rather than just the first one.
Signed-off-by: Mattias Amnefelt <mattiasa@avm.se>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Such streams are found on Blu-ray, and identified as EAC3 type in
avformat, while the bitstream of the core stream is actually a pure AC3
frame.
Adjust the parsing accordingly, since AC3 frames always hold 6 blocks
and the numblkscod syntax element is not present.
When var_stream_map option is used, %v must appear either in segment
name template or in the directory path. This latter case currently is
not handled and delete_segments flag of hls_flags is broken now. This
patch fix this. The root cause of the bug was that HLSSegment struct
only stores the final filename part, but not the final directory path.
Most of the cases, final path info is unneded, It only necessary when
you want to delete old segments (e.g in case of live streaming).
Without variant streams it was unnecessary to store the final directory
path, because all segment were stored into the same directory. But
introducing %v in directory names either require to store the final
directory path into HLSSegment or associate segments with their variant
streams to be able deleting them later. I have choosen the second
solution and introduced a variant index data member into the segment
struct.
Signed-off-by: Bela Bodecs <bodecsb@vivanet.hu>
Signed-off-by: Steven Liu <lq@onvideo.cn>
Support for writing copyright metadata was added in commit bed4fc54c9
for 3GP, MOV and iTunes metadata. 3GP and MOV cases are formally
specified. However, iTunes format does not have specification, and
it seems to have been assumed that it would use the same atom as
MOV (both being Apple formats).
However, Apple uses 'cprt' atom for iTunes metadata (do note that
the iTunes 'cprt' encoding is generic iTunes ItemList atom, not
the 3GP 'cprt' encoding. These are also inside different parent
atoms).
Most references trying to document iTunes atoms mention only
the 'cprt' tag. See:
- http://atomicparsley.sourceforge.net/mpeg-4files.html
- http://mutagen.readthedocs.io/en/latest/api/mp4.html
Same applies to other software supporting this tag. Most of them
encode and decode only the 'cprt' atom.
ffmpeg mov demuxer supports both atoms in this context. There are
few pieces of other software that support similarly both 'cprt' and
the incorrect '\251cpy' atom in this context. I believe they do it in
order to read the ffmpeg encoded incorrect copyright atom.
In light of the above this changes the copyright atom to 'cprt' as
it seems to be supported univerally and is the correct atom to use.
Signed-off-by: Timo Teräs <timo.teras@iki.fi>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Newly allocated data buffers (wavpack, prores, compressed buffers)
are padded to meet the requirements of AVPacket.
About 10x speed up in matroska_parse_frame().
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Simplifies code in matroska_parse_frame(). This is in preparation for
the following patch.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Data in EbmlBin objects is never changed after being read from the
input file (save for two specific cases with encoded CodePrivate), so
using AVBufferRef we can prevent unnecessary copy of data by instead
creating new references to said constant data.
Signed-off-by: James Almer <jamrial@gmail.com>
This is shared test and this simplifies code a bit. Follow up
commit will have additional tests for this function.
Signed-off-by: Timo Teräs <timo.teras@iki.fi>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Change the seek2any location in avformat_options to make code more
readable.
Signed-off-by: Jun Zhao <mypopydev@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
avdevice_register_all() is still required to register devices into
lavf (this is required due to lavd being somewhat of a hack).
Signed-off-by: Josh de Kock <josh@itanimul.li>
This reverts commit 909e00ae81.
There is no need to use the old API anymore as the new API now
behaves in the same way (treating devices as formats when loaded).
Signed-off-by: Josh de Kock <josh@itanimul.li>
Multiple cover images are supported by having multiple data atoms
inside the covr atom. AtomicParsley and mutagen amongst others
support and document this construct.
Signed-off-by: Timo Teräs <timo.teras@iki.fi>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This reverts commit 0fd475704e.
Revert "lavd: fix iterating of input and output devices"
This reverts commit ce1d77a5e7.
Signed-off-by: Josh de Kock <josh@itanimul.li>
The protocol requires libsrt (https://github.com/Haivision/srt) to be
installed
Signed-off-by: Sven Dueking <sven.dueking@nablet.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
av_strtok calls strspn on a non-NULL *saveptr, so not NULL initializing
it is an issue.
Fixes CID #1428568
Reviewed-by: Karthick Jeyapal <kjeyapal@akamai.com>
Signed-off-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Streams from a Segafilm cpk file can't be streamcopied because
keyframe flag isn't correctly set in stream index and
said flag is never conveyed to the packet
Fixes#7091
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Enables having proper bit rate values being written into the master
playlist in case of hard-constrained VBR where the maximum bit
rate utilized is known before hand.
Does the same thing as movenc.c, for example.
Signed-off-by: Jan Ekström <jan.ekstrom@aminocom.com>
When recv() returns 0 in case of SOCK_STREAM type, it means EOF and with
this patch returns value accordingly.
Signed-off-by: Bela Bodecs <bodecsb@vivanet.hu>
When building with libopenmpt 0.3, use the libopenmpt file header
probing functions for probing. libopenmpt probing functions are
allocation-free and designed to be as fast as possible.
For libopenmpt 0.2, or when libopenmpt 0.3 file header probing cannot
probe successfully due to too small probe buffer, test the filename
against the file extensions supported by the libopenmpt library that
is actually linked, instead of relying on a hard-coded file extension
list. File extension testing is also allocation-free and designed to
be fast in libopenmpt. Avoiding a hard-coded file extension list is
useful because later libopenmpt versions will likely add support for
more module file formats.
libopenmpt file header probing is tested regularly against the FATE
suite and other diverse file collections by libopenmpt upstream in
order to avoid false positives.
FATE passes with './configure --enable-libopenmpt' as well as with
'./configure --enable-libopenmpt --enable-libmodplug'.
libopenmpt probing adds about 5%..10% cpu time (depending on precise
usage pattern and host CPU and compiler version used for libopenmpt)
compared to all current internal FFmpeg probing functions combined in
tools/probetest for all of its module formats combined (currently 41
modules formats in libopenmpt 0.3.4 and 234 file formats in FFmpeg).
Signed-off-by: Jörn Heusipp <osmanx@problemloesungsmaschine.de>
Reviewed-by: Josh de Kock <josh@itanimul.li>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If the string consists entirely of whitespace, this could in theory
continue to write '\0' before the start of the memory allocation. In
practice, it didn't really happen: the generic HTTP header parsing code
already skips leading whitespaces, so the string is either empty, or
consists a non-whitespace. (The generic code and the cookie code
actually have different ideas about what bytes are whitespace: the
former uses av_isspace(), the latter uses WHITESPACES. Fortunately,
av_isspace() is a super set of the http.c specific WHITESPACES, so
there's probably no case where the above assumption could have been
broken.)
It's trivial to craft a HTTP response that will make the code for
skipping trailing whitespace access and possibly overwrite bytes outside
of the memory allocation. Why this can happen is blindingly obvious: it
accesses cstr[strlen(cstr)-1] without checking whether the string is
empty.
libavformat prints a warning that the cookie couldn't be parsed (see
callers of parse_cookie()). This is obviously not true - it could be
parsed, but was simply ignored. Don't return an error to avoid the
warning.
Thanks for the discussion. Here's the next version, now with /25 and removed
ff_log2().
The blocksize of the PCM decoder is hard-coded. This creates
unnecessary delay when reading low-rate (<100Hz) streams. This creates
issues when multiplexing multiple streams, since other inputs are only
opened/read after a low-rate input block was completely read.
This patch decreases the blocksize for low-rate inputs, so
approximately a block is read every 40ms. This decreases the startup
delay when multiplexing inputs with different rates.
Signed-off-by: Philipp M. Scholl <pscholl@bawue.de>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: runtime error: signed integer overflow: -9166684017437101870 + -2495066639299164439 cannot be represented in type
Fixes: Chromium bug 791349
Reported-by: Matt Wolenetz <wolenetz@google.com>
Reviewed-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes seek for files with empty edits and files with negative ctts
(dts_shift > 0). Added fate samples and tests.
Signed-off-by: Sasi Inguva <isasi@isasi.mtv.corp.google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
No testcase, this was found during code review
Found-by: Matt Wolenetz <wolenetz@google.com>
Reviewed-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: shift exponent 48 is too large for 32-bit type 'int'
Fixes: Chromium bug 786793
Reported-by: Matt Wolenetz <wolenetz@google.com>
Reviewed-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 3775922176 * 4278190080 cannot be represented in type 'long'
Fixes: Chromium bug 791237
Reported-by: Matt Wolenetz <wolenetz@google.com>
Reviewed-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: Chromium bug 795653
Fixes: signed integer overflow: 9223372036854775807 + 1 cannot be represented in type 'long'
Reported-by: Matt Wolenetz <wolenetz@google.com>
Reviewed-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Consider edit rate when determining edit_units_per_packet and also make sure
that checks are done in edit rate time base and not in stream time base.
Fixes some errors reported with the sample in ticket #5863.
Signed-off-by: Marton Balint <cus@passwd.hu>
In this loop, |i| is the "index". And the memory allocated should be at
least the current "count", which is |i + 1|.
BUG=801821
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: runtime error: signed integer overflow: 7738135736989908991 - -7898362169240453118 cannot be represented in type 'long'
Fixes: Chromium bug 796778
Reported-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: runtime error: signed integer overflow: 5 * -2147483647 cannot be represented in type 'int'
Fixes: Chromium bug 817338
Reviewed-by: Matt Wolenetz <wolenetz@google.com>
Reported-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
ff_get_extradata() frees any existing extradata before allocating now,
and using av_free() here leaves a dangling pointer that will result in
a double free.
Fixes a regression since 0ca33b1d4e.
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Numeric hosts in certificates are not very common, but supported by LibreSSL.
Forward the IP address to make verification work in this case.
Fixes ticket #7029
This prevents leaks in the rare cases the function is called when extradata
already exists.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
mov_read_ctts ignores ctts entries having count <= 0. Generally, the
aggregate of all ctts entries' count fields resulting from mov_read_ctts
can be less than the corresponding sample_count.
mov_build_index attempts to normalize any existing ctts_data counts to
be 1, to make a 1-1 mapping of a ctts_data entry to a sample.
That 1-1 mapping left a tail of uninitialized ctts_data entries when the
aggregate, normalized ctts_count < sample_count.
Even more generally, later usage of ctts_data may depend on the entire
ctts_allocated_size having been initialized.
This change memsets the entire allocation of the normalized ctts_data in
mov_build_index, to prevent use of uninitialized data later.
BUG=816787
Change-Id: I7fd7db255e3aeed076ee32c90cb2df211741c052
Reviewed-on: https://chromium-review.googlesource.com/947110
Reviewed-by: Xiaohan Wang <xhwang@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
fix ticket: #7044
Get the first video stream to reference for split segment
when there have more than one video stream
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Reviewed-by: Karthick Jeyapal <kjeyapal@akamai.com>
The allocated size of sc->ctts_data is
(st->nb_index_entries + entries) * sizeof(*sc->ctts_data).
The size to memset at offset sc->ctts_data + sc->ctts_count should be
(st->nb_index_entries + entries - sc->ctts_count) *
sizeof(*sc->ctts_data))
The current code missed |entries| I believe, which was introduced in
https://patchwork.ffmpeg.org/patch/5541/.
However, after offline discussion, it seems the original code is much
more clear to read (before https://patchwork.ffmpeg.org/patch/5541/).
Hence this CL revert the memset logic to it's previous state by
remembering the |old_ctts_allocated_size|, and only memset the newly
allocated entries.
BUG=812567
Change-Id: Ibe94c7138e5818bfaae76866bfa6619a9b8a2b6b
Reviewed-on: https://chromium-review.googlesource.com/934925
Reviewed-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Obviously this is still not perfect, but better then it was. Using the first
index table and mxf->current_edit_unit is still hardcoded in many places, so
this change has hopefully the less chance of breaking anything that works
now.
Signed-off-by: Marton Balint <cus@passwd.hu>
Use body_offset of the partitions to search for the partition with the given
offset in the essence. This makes the function find the correct partition for
non frame-wrapped essences as well, where only the essence data is part of the
the edit unit byte count, not the KLV-s.
Signed-off-by: Marton Balint <cus@passwd.hu>
In order to do that we have to parse the EssenceContainerData and assign the
proper body_sid and index_sid to the tracks from the corresponding source
packages.
This fixes packets returned in the wrong stream for some OP1-b files.
Based on a patch by Alex Mogurenko from https://github.com/da8eat/FFmpeg
Reference: http://mogurenko.com/2018/01/02/mxf-op1b-ffmpeg-part1/
Signed-off-by: Marton Balint <cus@passwd.hu>
The reference point for a KAG is the first byte of the key of a Partition Pack.
Fixes ticket #2817.
Fixes ticket #5317.
Signed-off-by: Marton Balint <cus@passwd.hu>
This only affected demuxers that didn't return reference counted packets.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
