HLS demuxer calls the subdemuxer avformat_find_stream_info() while
overriding the subdemuxer AVFMTCTX_NOHEADER flag by clearing it.
However, this prevents some streams in some MPEG TS streams from being
detected properly.
Simply removing the clearing of the flag would cause the inner
avformat_find_stream_info() call to take longer in some cases, without
a way to control it.
To fix the issue, do not clear the flag but propagate it to HLS demuxer.
To avoid the above-mentioned mandatory delay, the call to
avformat_find_stream_info() is dropped except in the HLS ID3 timestamped
case. The HLS demuxer user should be calling avformat_find_stream_info()
on the HLS demuxer if it wants to find the stream info.
The main streams are now created dynamically after read_header time if
the subdemuxer uses AVFMTCTX_NOHEADER (mpegts).
Subdemuxer avformat_find_stream_info() is still called for the HLS ID3
timestamped case as the HLS demuxer needs to know the packet durations
to properly interleave ID3 timestamped streams with MPEG TS streams on
sub-segment level.
Fixes ticket #4930.
Creation of main demuxer streams from subdemuxer streams is moved to
update_streams_from_subdemuxer() which can be called repeatedly.
There should be no functional changes.
Commit 81306fd4bdf ("hls: eliminate ffurl_* usage", merged in d0fc5de3a6)
changed the hls demuxer to use AVIOContext instead of URLContext for its
HTTP requests.
HLS demuxer uses the "offset" option of the http demuxer, requesting
the initial file offset for the I/O (http URLProtocol uses the "Range:"
HTTP header to try to accommodate that).
However, the code in libavformat/aviobuf.c seems to be doing its own
accounting for the current file offset (AVIOContext.pos), with the
assumption that the initial offset is always zero.
HLS demuxer does an explicit seek after open_url to account for cases
where the "offset" was not effective (due to the URL being a local file
or the HTTP server not obeying it), which should be a no-op in case the
file offset is already at that position.
However, since aviobuf.c code thinks the starting offset is 0, this
doesn't work properly.
This breaks retrieval of ranged media segments.
To fix the regression, just drop the seek call from the HLS demuxer when
the HTTP(S) protocol is used.
Commit 9200514ad8 ("lavf: replace AVStream.codec with
AVStream.codecpar") merged in commit 6f69f7a8bf changed
avformat_find_stream_info() to put the extradata it got from
st->parser->parser->split() to st->internal->avctx instead of st->codec
(extradata in st->internal->avctx will be later copied to st->codecpar).
However, in the same function, the "is stream ready?" check was changed
to check for extradata in st->codecpar instead of st->codec, even
though st->codecpar is not yet updated at that point.
Extradata retrieved from split() is therefore not considered anymore,
and avformat_find_stream_info() will therefore needlessly continue
probing in some cases.
Fix that by checking for the extradata at st->internal->avctx where it
is actually put.
Without EXT-X-MAP support we miss the first bytes of some streams.
These streams worked by luck before byte-ranged segment support was added in
da7759b357
Fixes ticket #4797.
Commit ad701326b4 ("avformat/hls: open playlists immediately when
AVDISCARD_ALL is dropped") inadvertently caused first_packet to never be
cleared, causing select_cur_seq_no() to not use the specific code for
live streams.
In practice this means that when the user selects a different audio
track during live stream (i.e. non-VOD) playback, there may be some
additional delay as the code might select an incorrect segment at first,
and we have to wait for video to catch audio (if too late segment was
selected) or to download more following audio segments (if too early
segment was selected).
Fix that by restoring the zeroing of first_packet.
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>
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>
Not all "sub-playlists" are variant playlists (containing the same
content with a different bitrate, etc) in the current version of the HLS
specification. They can now also be alternative renditions, containing
e.g. alternative audio tracks etc.
Decouple playlists from variants to prepare for handling the new
features.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Add ff_id3v2_read_dict() for parsing without AVFormatContext, but
instead with AVIOContext and AVDictionary.
AVFormatContext is still used for logging, if available.
Chapter parsing is the only non-logging functionality that actually
needs AVFormatContext, and AFAICS it should be modified to write the
data to ID3v2ExtraMeta first, from where it can be implanted to
AVFormatContext by a separate function (like it is done with
read_apic() and ff_id3v2_parse_apic()). That is outside the scope of
this patch, though.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Add AVOptions for setting the initial offset and the ending offset, so
they can be used for setting an appropriate Range header.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Used in some HLS streams ("Timed Metadata for HTTP Live Streaming").
This is just ID3 tags at arbitrary stream positions, but I still added
"timed" to the codec name to avoid confusion with regular non-stream ID3
tags.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
AAC specification has 7.1(wide) as a default layout for 8-channel
streams (channel config 7). However, at least Nero AAC encoder encodes
non-wide 7.1 streams using the default channel config 7, mapping the
side channels of the original audio stream to the second
AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD decodes
the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
the incorrect streams as if they were correct (and as the encoder
intended).
FFmpeg currently decodes such files by-the-spec, i.e. after decoding the
original front pair will be in AV_CH_FRONT_x_OF_CENTER and the original
side pair will be in AV_CH_FRONT_x.
As actual intended 7.1(wide) streams are very rare while misencoded 7.1
files actually exist in the wild, default to assuming a 7.1 layout was
intended unless in strict mode.
Fixes playback of e.g. 8_Channel_ID.m4a in samples.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Return value of avpriv_aac_parse_header() is not checked correctly. Fix
it.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Align IEC 61937 length_code for DTS-HD so that
(length_code & 0xf) == 0x8. This is reportedly needed with some
receivers.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Align IEC 61937 length_code for DTS-HD so that
(length_code & 0xf) == 0x8. This is reportedly needed with some
receivers.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
As per issue2629, most 23.976fps matroska H.264 files are incorrectly
detected as 24fps, as the matroska timestamps usually have only
millisecond precision.
Fix that by doubling the amount of timestamps inspected for frame rate
for streams that have coarse time base. This also fixes 29.970 detection
in matroska.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 78431098f9)
Tested with mplayer based on this report
http://thread.gmane.org/gmane.comp.video.mplayer.user/66043/focus=66063
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
As per issue2629, most 23.976fps matroska H.264 files are incorrectly
detected as 24fps, as the matroska timestamps usually have only
millisecond precision.
Fix that by doubling the amount of timestamps inspected for frame rate
for streams that have coarse time base. This also fixes 29.970 detection
in matroska.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is used at least on some older DVB broadcasts for dual-mono audio
tracks.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 6a7e074eb9)
This avoids the core substream extensions scan when the EXT_AUDIO_ID
field indicates no extensions or only unsupported extensions. The scan
is done only if the value of EXT_AUDIO_ID is unknown or indicates a
present XCh extension which we can decode.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 7e06e0ede3)
This avoids the core substream extensions scan when the EXT_AUDIO_ID
field indicates no extensions or only unsupported extensions. The scan
is done only if the value of EXT_AUDIO_ID is unknown or indicates a
present XCh extension which we can decode.
Signed-off-by: Mans Rullgard <mans@mansr.com>
There is a check for HAVE_BIGENDIAN when outputting the IEC 61937
stream. On big-endian systems the payload data is not byteswapped,
causing in effect the outputted payload data to be in a different byte
order on big-endian than on little-endian systems.
However, the IEC 61937 preamble (and the final odd byte if present) is
always outputted in the same byte order. This means that on big-endian
systems the headers have a different byte order than the payload,
preventing useful use of the output.
Fix that by outputting the data in a format suitable for sending to an
audio device in S16LE format by default. Output as big-endian (S16BE)
is added as an AVOption. This makes the muxer output the same on all
archs by default.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 57f2c9aed9)
There is a check for HAVE_BIGENDIAN when outputting the IEC 61937
stream. On big-endian systems the payload data is not byteswapped,
causing in effect the outputted payload data to be in a different byte
order on big-endian than on little-endian systems.
However, the IEC 61937 preamble (and the final odd byte if present) is
always outputted in the same byte order. This means that on big-endian
systems the headers have a different byte order than the payload,
preventing useful use of the output.
Fix that by outputting the data in a format suitable for sending to an
audio device in S16LE format by default. Output as big-endian (S16BE)
is added as an AVOption. This makes the muxer output the same on all
archs by default.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
It is useful for applications that hand input data directly to lavf via
a ByteIOContext.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 3940caad02)
There is no need to pass the ByteIOContext via a pointer to a pointer
anymore.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit aad216fd7e)
NASM versions older than 2.08 fail to build ffmpeg with several
"error: operation size not specified" errors but this is not caught in
configure.
Fix that by checking if "pextrd [eax], xmm0, 1" works in configure.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 48545a8f72)
NASM versions older than 2.08 fail to build ffmpeg with several
"error: operation size not specified" errors but this is not caught in
configure.
Fix that by checking if "pextrd [eax], xmm0, 1" works in configure.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
DTS-HD HRA streams do not always have an XBR extension in the extension
substream. Instead they can have only XXCh and X96 extensions in
there and still be considered DTS-HD HRA.
This is also confirmed with Onkyo TX-SR607 receiver which recognizes
such a stream as HiRes Audio.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 8f4a5d225c)
DTS-HD HRA streams do not always have an XBR extension in the extension
substream. Instead they can have only XXCh and X96 extensions in
there and still be considered DTS-HD HRA.
This is also confirmed with Onkyo TX-SR607 receiver which recognizes
such a stream as HiRes Audio.
Signed-off-by: Mans Rullgard <mans@mansr.com>
wav. In that case, DTS can be transmitted through S/PDIF without
the IEC 61937 headers.
Patch by Anssi Hannula, anssi d hannula a iki d fi
Originally committed as revision 26160 to svn://svn.ffmpeg.org/ffmpeg/trunk
Noticed by CrystalP from XBMC.
Patch by Anssi Hannula, anssi d hannula a iki d fi
Originally committed as revision 26130 to svn://svn.ffmpeg.org/ffmpeg/trunk
This works at least for some people testing it.
Patch by Anssi Hannula, anssi d hannula a iki fi
Originally committed as revision 25834 to svn://svn.ffmpeg.org/ffmpeg/trunk
