This commit allows to benefit from implementing child_next
callback for muxers' AVClasses.
Without that, options cannot be set in nested structs.
Signed-off-by: Lukasz Marek <lukasz.m.luki2@gmail.com>
Files produced by windows media center contain meaningless mpeg1 sequence
header. The mpeg2 decoder detects the presence mpeg1 sequence header start
codes and attempts to decode the stream as mpeg1. (This problem introduced
in 73a2d16b.)
Fixes ticket #3601.
Signed-off-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'ef9732162cd4b593c6db28fdd352ebef21b5c1ca':
rmdec: do not export anything to AVCodecContext.codec_name
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '6072184e702b4b631ac72f1b66b75e5f21e0ce2d':
asfenc: use codec descriptors instead of AVCodecs to write codec info
Conflicts:
tests/ref/lavf/asf
tests/ref/seek/lavf-asf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '92e4b643dfdafdb6528f1baffdbea2b9a028d7c0':
oggparseskeleton: do not use AVCodecContext.codec_name
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'a1aa37dd0b96710d4a17718198a3f56aea2040c1':
matroskaenc: write CodecDelay
Conflicts:
libavformat/matroskaenc.c
tests/ref/lavf/mkv
tests/ref/seek/lavf-mkv
This is largely not merged as it causes assertion failures and av sync errors
Further investigation of this is warranted if the changes are found to
fix/improve something in relation to d92b1b1bab
See: d92b1b1bab
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also, stop using AVCodecContext.codec_name as fallback, since it will be
deprecated.
Changes the result of the lavf-asf test (and its associated seektest),
since 'msmpeg4v3' gets written instead of just 'msmpeg4'.
* commit '7d027b9d6d6290557cc5d4fc56f4b9ed630a7feb':
librtmp: Map native options to librtmp ones when possible
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '7266e24f176389d2e81bfc7c829934f7c8ae361c':
hls: Sync the file number with the start sequence
Conflicts:
libavformat/hlsenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c7603b3c243331057300337a61464e6ac4a605cb':
hls: Print start_number as first sequence value
Conflicts:
libavformat/hlsenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '87a3ea3192bf5e4aafa08bca8686a2b577eae818':
segment: Report the current media sequence
Conflicts:
libavformat/segment.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Partially undoes commit 2c4e08d893:
riff: always generate a proper WAVEFORMATEX structure in
ff_put_wav_header
A new flag, FF_PUT_WAV_HEADER_FORCE_WAVEFORMATEX, is added to force the
use of WAVEFORMATEX rather than PCMWAVEFORMAT even for PCM codecs.
This flag is used in the Matroska muxer (the cause of the original
change) and in the ASF muxer, because the specifications for
these formats indicate explicitly that WAVEFORMATEX should be used.
Muxers for other formats will return to the original behavior of writing
PCMWAVEFORMAT when writing a header for raw PCM.
In particular, this causes raw PCM in WAV to generate the canonical
44-byte header expected by some tools.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* cigaes/master:
lavfi/drawtext: allow to format pts as HH:MM:SS.mmm.
lavf/concatdec: implement automatic conversions.
lavf/concatdec: reindent after last commit.
lavf/concatdec: always do stream matching.
lavf/concatdec: check match_streams() return value.
lavf/concatdec: use a structure for each stream.
ffprobe: use the codec descriptor if no decoder was found.
lavf/matroska: add "binary" pseudo-MIME type.
lavc: minor bump and APIchanges for AVCodecDescriptor.mime_types.
lavc: add a mime_types field to codec descriptors.
lavc: add AV_CODEC_ID_BIN_DATA.
lavc: add codec descriptors for TTF and OTF.
lavc: add codec descriptors for deprecated ids.
lavc/codec_desc: add separation comment.
tools/ffhash: implement base64 output.
tools/ffhash: use av_hash_final_hex().
lavu/hash: add hash_final helpers.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '5a70a783f04919514efec7751d710b64d8975fd7':
hls: Add an option to prepend a baseurl to the playlist entries
Conflicts:
doc/muxers.texi
libavformat/hlsenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '5c08ae4f37281441188447cd04dcaf7cd7ce031f':
segment: Add an option to prepend a string to the list entries
Conflicts:
doc/muxers.texi
libavformat/segment.c
See: 5e278c19c7
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This doesn't allow encoding of DTS or TrueHD. It just sets the correct
stream ID in the TS output file when a DTS or TrueHD audio stream is copied.
Fixes ticket #1398
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* cehoyos/master:
Enable muxing ac-3 in caf.
Use correct msvc type specifiers for ptrdiff_t and size_t.
Fix vf_eq.c and vf_eq2.c compilation with !HAVE_6REGS.
Fix libpostproc compilation with !HAVE_6REGS.
Never write 0 as maximum bitrate for asf files.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The rational for this is another issue that plex has exposed. When it is
conducting a transcode of video to HLS for streaming, my father noticed
artifacts when played on his GoogleTV (NSZ-GT1). He sent me a test file
and I reproduced it on my device of the same model. It is important to
note that the artifacts were not present when streaming to VLC or QuickTime
Player. I copied the command-line that plex used, and conducted all of the
following tests using FFmpeg git.
Transcode to HLS: artifacts on playback
Transcode to TS: playback is fine
Cat HLS segments into a single TS: playback is fine
Segment single TS file to segments: artifacts on playback
Segment single TS file to segments using Apple's HLS segmenter: playback is
fine
At this point I carefully examined the differences between Apple's HLS
segmenter output and FFmpeg's. Among the considerable differences, I
noticed that the video PES packets always had a 0 length. So I continued:
Transcode to HLS using FFmpeg with 0 length PES packets: playback is fine.
Segment single TS to segments with 0 length PES packets: playback is fine.
All failures mentioned are only on the GTV since it is the only player on
which I could reproduce artifacts. I only tested the GTV, VLC, and
QuickTime Player though, so my test case is limited. I do not know if
other players exhibit this issue.
Since it was useful last time, I have uploaded the test file as
hls_pes_packet_length.m4v along with its associated txt file which contains
the transcode command-line that was used.
Reviewed-by: Kieran Kunhya <kierank@obe.tv>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Several chunked formats (AIFF, IFF,DSF) store ID3 metadata within an 'ID3 '
chunk tag. If such chunks are stored sequentially, it is possible for the
ID3v2 parser to confuse the chunk tag for the ID3 magic number. e.g.
[1st chunk tag ('ID3 ') | chunk size] [ID3 magic number | metadata ...]
[2nd chunk tag ('ID3 ') | chunk size] [ID3 magic number | metadata ...]
Fixes ticket #3530.
Signed-off-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This ensures stream[0] is always the audio stream (an assumption made
in dsf_read_packet).
Signed-off-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '32d05934abc7427bb90380a4c1ab20a15fd7d821':
mp3dec: decode more data from Info header
Conflicts:
libavformat/mp3dec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'fbd8e042107ec63e0ddf155588c59dcb76007641':
mp3dec: move XING/Info and VBRI parsing into their own functions
Conflicts:
libavformat/mp3dec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This adds a function to export raw replaygain values (i.e. in the (u)int32_t
form). It first checks whether AV_PKT_DATA_REPLAYGAIN side data is present, in
which case it does nothing.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
After this commit applications needs to call av_format_inject_global_side_data()
or handle AVStream side data by some other means if they want it not to be lost.
This fixes a API incompatibility with libav.
libav API does not allow the data to be passed through AVPackets
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '6477139721f559b26eafd415e23e13ea2b0c27e1':
rtmpproto: Make sure to pass on the error code if read_connect failed
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In all other cases where ff_rtmp_packet_read is used, the packet returned
is passed to rtmp_parse_result more or less immediately. In this single
case, the content of the packet was required to be a connect packet.
Some clients, e.g. Open Broadcaster Software, send a chunk size packet
before the connect packet. If the first packet is a chunk size packet,
handle it and read another one, requiring this to be a connect packet
instead.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously, if read_connect failed, the ret variable was unmodified
and had the value 0, indicating success, which then was returned from
the rtmp_open function, even though it actually failed.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
bits_per_coded_sample and block_align are calculated again at end of if() block
Signed-off-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Instead of using a fixed bitrate_idx, calculate a matching bitrate for
the XING header.
Using a fixed bitrate_idx causes tools such as file(1) and mediainfo(1)
to report wrong bitrate and bitrate mode when using CBR.
Bug-Id: https://bugs.debian.org/736088
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Provides API to query device capabilities.
Each device must implement callbacks to benefit from this API.
Signed-off-by: Lukasz Marek <lukasz.m.luki@gmail.com>
ffurl_seek() will not work even when it should be a no-op, so do not
call it on crypto protocol.
Also replace use of ffurl_size() for the same reason.
Reported-by: Michael Schenk <Michael.Schenk@albistechnologies.com>
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Calling avformat_free_context() right after avformat_alloc_output_context2()
leaved option's default values not freed.
Options were freed only in av_write_trailer().
Signed-off-by: Lukasz Marek <lukasz.m.luki2@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* cehoyos/master:
Fix a typo in amr.c.
Remove an unneeded include of avassert.h from amr.c.
Do not allow writing invalid wav channel layouts by default.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Avoids a warning with gcc 4.7 and -Wtype-limits. Albeit
superfluous (At least gcc 4.8 didnt consider this been
a problem).
Signed-off-by: Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
Also add reason phrases from http://www.ietf.org/rfc/rfc2326.txt
and macro to translate.
Signed-off-by: Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
Since we are basically seeking the AVIOContext under the subdemuxer, we
need to flush the subdemuxer to avoid old packets from being read from
the packet queue after the seek.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Properly take stream_index into account so that a keyframe will be
looked for in the specified stream_index only.
Similarly, only check timestamp validity against the specified
stream_index.
Also remove code for stream_index == -1 case which does not actually
happen as it is handled by generic code.
This is based on an initial patch by James Deng.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Do not try to delay opening newly required playlists until a segment
switch. Applications expect that newly selected undiscarded streams are
available immediately, especially with alternative rendition streams
(selectable audio/subtitle tracks).
One might think that delaying variant stream switch until a segment
switch would allow a "seamless" switch without us having to download a
specific segment from two different variant playlists. However, that is
not the case, since the application would have to keep the previous
stream available (undiscarded) until the first packet of the newly
selected stream arrives, but by that time the demuxer would have already
downloaded the next segment of both variants.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
As per spec 3.4.3 ("A client MUST NOT assume that segments with the same
sequence number in different Media Playlists contain matching content.")
we cannot use sequence numbers for packet ordering.
This can be seen e.g. in the subtitle streams of
bipbop_16x9_variant.m3u8 that have considerably longer segments and
therefore different numbering.
Since the code now exclusively syncs using timestamps that may wrap, add
some additional checking for that.
According to the HLS spec all the timestamps should be in 33-bit MPEG
format and synced together.
v2: cleaner wrap detection
v3: further wrap detection improvements
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
While selecting a packet to return to caller in read_packet(), the code
corrects the timestamps for starting timestamps.
However, this is wrong, since for live streams the initial timestamps
might differ just because of the time delay between the retrieval of the
various Media Playlists.
Fortunately, spec 6.2.4 mandates that all variant streams must have
matching timestamps, so we do not need to correct for initial
timestamps.
Drop the correction code.
Note that ID3 timestamps were previously ignored, so this code was
previously actually needed.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Check if the playlist is still needed just before requesting the next
segment instead of after exhausting the previous segment.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Improve selection of the segment sequence number when restarting the
reception of a playlist after it was suspended due to being unneeded
(due to discard flags).
The current code assumes that each playlist contains matching data with
the same sequence number, while spec 3.4.3 specifically says that that
is not the case. Often subtitle playlists also have longer target
durations as well, causing the selection to be completely wrong.
Instead prefer using the playlist segment duration information for
non-live playlists, and other means if that is not possible.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Seeking needs to be tracked on a per-playlist basis, since the resyncing
code in hls_read_packet() has to sync each playlist to the seek
timestamp instead of stopping after the first playlist has reached it.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
HLS provides MPEG TS timestamps via ID3 tags in the beginning of each
segment of elementary audio streams.
v2: fix issues with streams that have multiple ID3 tags
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Add support for EXT-X-BYTERANGE added in HLS protocol v4.
v2: Better comment explaining ffurl_seek call and fix cur_seg_offset not
being updated.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Even if we returned AVERROR_EOF previously due to playlist no longer
being needed, we may still be called again, and we do not want to
trigger a segment download in that case.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
HLS protocol version 4 added alternative renditions to the
specification (e.g. alternative audio tracks).
The EXT-X-MEDIA tags can also contain metadata for "renditions" (i.e.
tracks) of the main Media Playlist.
Add support for those.
Note that the same rendition (AVStream) may be associated with multiple
variants (AVPrograms).
Alternative subtitle tracks will require additional work and are
therefore not enabled yet.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
According to the ReplayGain spec, the peak amplitude may overflow and may result
in peak amplitude values greater than 1.0 with psychoacoustically coded audio,
such as MP3. Fully compliant decoders must allow peak overflows.
Additionally, having peak values in the 0<->UINT32_MAX scale makes it more
difficult for applications to actually use the peak values (e.g. when
implementing clipping prevention) since values have to be rescaled down.
This patch corrects the peak parsing by removing the rescaling of the decoded
values between 0 and UINT32_MAX and the 1.0 upper limit.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The gain sign was incorrectly decoded: since the FFSIGN() macro treats 0 as
negative, gain values starting with "0." were always decoded as negative.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Further performance improvements and security fixes by
Vittorio Giovara, Luca Barbato and Diego Biurrun.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
* qatar/master: (31 commits)
riff: Add an additional AAC TwoCC
riff: support 0xa100 TwoCC for G723_1
riff: add 0x594a TwoCC for Xan DPCM
riff: add 0x64 to g726
riff: add G723_1 wav tag
riff: map 0x0038 to amrnb
riff: Support FLIC FourCC
riff: add escape130 FourCC
riff: support 'aas4' FourCC
riff: add "YUV8" FourCC
riff: Add "S263" FourCC
riff: Support XMPG as mpeg1
riff: support BW10 as mpeg1
riff: Add SLDV FourCC for dvvideo
riff: Support NTSC forward dvcpro videos
riff: add dvis/pdvc FourCCs
riff: add "GXVE" FourCC for WMV2
riff: add PLV1 fourcc to mpeg4
riff: Support decoding ASP variant from QNAP Systems
riff: add "SM4V" FourCC
...
Conflicts:
libavformat/riff.c
See: various commits, all the merged changes where in ffmpeg already
Merged-by: Michael Niedermayer <michaelni@gmx.at>
