It can read less than the requested amount, in which case buf contains
uninitialized data, causing problems like segmentation faults later on.
Also make sure that image->size is positive, so that it can't match a
negative error code.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
FLAC streams originating from the FLAC encoder send updated and more
complete STREAMINFO metadata as part of the last packet, so write that
to CodecPrivate instead of the incomplete one available in extradata
during init.
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
aac_adtstoasc makes the aac extradata available only after the first packet
is filtered, and as packet side data.
Assume extradata will be available as part of the first packet if
avpriv_mpeg4audio_get_config() fails the first time due to missing extradata
and reserve space for the OutputSampleRate element in the Tracks master.
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Add keyframe index metadata
Used to facilitate seeking; particularly for HTTP pseudo streaming.
1. read live streaming or file by sequence
2. if use add_keyframe_index option, add a mark flag at the position,
use to insert new context at the last step.
3. add the keyframes *offset* and *timestamp* into a list
4. if use add_keyframe_index option, shift the metadata data from
mark flag offset
5. insert the keyframes *offset* and *timestamp* from the list by
sequence
6. free the list
7. end.
Add FATE test case;
Reviewed-by: Lou Logan <lou@lrcd.com>
Signed-off-by: Steven Liu <liuqi@gosun.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The code assumes that s->streams[0] is valid.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
If the value is negative then it means padding at the start of the packet
instead of at the end.
Based on a patch by Hendrik Leppkes.
Reviewed-by: James Zern <jzern-at-google.com@ffmpeg.org>
Signed-off-by: James Almer <jamrial@gmail.com>
Use the hls_close function to reduce code duplication.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
This is needed for the following commit.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Commit 04964ac311 ("avformat/hls: Fix missing streams in some
cases with MPEG TS") caused a regression where subdemuxer streams that
use probing (e.g. dts/eac3/mp2 in mpegts) no longer get probed properly.
This is because the codec parameters from the subdemuxer stream, once
probed, are not passed on to the main stream.
Fix that by updating the codec parameters if the codec id changes.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
AVCodecParameters.sample_rate is a signed integer, so
XMVAudioPacket.sample_rate should be, too.
A negative sample rate doesn't make sense and triggers assertions in
av_rescale_rnd.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
A negative sample rate doesn't make sense and triggers assertions in
av_rescale_rnd.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
A negative sample rate doesn't make sense and triggers assertions in
av_rescale_rnd.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
A negative sample rate doesn't make sense and triggers assertions in
av_rescale_rnd.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
A negative sample rate doesn't make sense and triggers assertions in
av_rescale_rnd.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
A negative sample rate doesn't make sense and triggers assertions in
av_rescale_rnd.
Also check for errors from avpriv_mpeg4audio_get_config in
ff_mp4_read_dec_config_descr.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
A negative sample rate doesn't make sense and triggers assertions in
av_rescale_rnd.
fate-aac-al07_96 fails if sample_rate == 0 is rejected in
ff_mov_read_stsd_entries.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
A negative sample rate doesn't make sense and triggers assertions in
av_rescale_rnd.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
A negative sample rate doesn't make sense and triggers assertions in
av_rescale_rnd.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
A negative sample rate doesn't make sense and triggers assertions in
av_rescale_rnd.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Fixes: error: dereferencing pointer to incomplete type
Tested-by: Dave Yeo <daveryeo@telus.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This is required since some programs are not able to correctly recognize
the metadata. See H.222, 2.6.58 Metadata pointer descriptor.
putstr8() is modified in order to allow to skip writing the string
length.
This should be more useful for users since numerical values for channel
layout can be confusing and unintuitive.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Metadata streams have priv_data set to NULL.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
When ffplay is used to play from the RTSP URL that serves 24 bit audio
content, ffplay fails to recognize the audio codec format. The attached
patch adds support for playing 24 bit audio content over RTSP by
defining a dynamic payload handler for "L24".
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The bitstream filters do not work with merged in side data
This leaves the input packet split if it is being split.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
This reverts commit fba2a8a254.
The changes were right for av_write_frame() but not for av_interleaved_write_frame().
The following commit will fix this in a simpler way.
Signed-off-by: James Almer <jamrial@gmail.com>
Similarly, merge it back before returning.
Fixes ticket #5927.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
According to spec ISO_IEC_15444_12 "For any media stream for which no segment index is present, referred to as non‐indexed stream, the media stream associated with the first Segment Index box in the segment serves as a reference stream in a sense that it also describes the subsegments for any non‐indexed media stream."
Signed-off-by: Sasi Inguva <isasi@google.com>
Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The parser depends on the codec and thus must not be used with a different one.
If it is, the 'avctx->codec_id == s->parser->codec_ids[0] ...' assert in
av_parser_parse2 gets triggered.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Andreas Cadhalpun
Sasi Inguva
James Almer
Steven Liu
Michael Niedermayer
Diego Biurrun
Vittorio Giovara
Anssi Hannula
fuqiuping
Stefano Sabatini
liu jc
Timur Aydin
Matt Oliver
Carl Eugen Hoyos