Commit 6c7f1155 authored by wm4's avatar wm4 Committed by Michael Niedermayer

avformat/mp3dec: avoid early EOF with concatenated gapless mp3s

Consider a file created with something like:

    cat file1.mp3 file2.mp3 > result.mp3

Then if file2.mp3 has gapless information, result.mp3 would stop playing
something in the middle. This happens because the gapless info directs
the decoder to discard all samples after a certain position. To make
matters worse, the gapless info of file2.mp3 will be used when playing
the file1.mp3 part, because the gapless info is located at the end of
the file.

While handling concatenated gapless files correctly would be insane and
a lot of effort (especially without scanning the whole file on opening),
it's easy to prevent at least early EOF. Playback will happen to work,
even if it's slightly broken.
Signed-off-by: 's avatarMichael Niedermayer <michaelni@gmx.at>
parent 36bf549b
...@@ -1036,7 +1036,14 @@ typedef struct AVStream { ...@@ -1036,7 +1036,14 @@ typedef struct AVStream {
* avoided for broken by design formats such as mp3 with ad-hoc gapless * avoided for broken by design formats such as mp3 with ad-hoc gapless
* audio support. * audio support.
*/ */
int64_t end_discard_sample; int64_t first_discard_sample;
/**
* The sample after last sample that is intended to be discarded after
* first_discard_sample. Works on frame boundaries only. Used to prevent
* early EOF if the gapless info is broken (considered concatenated mp3s).
*/
int64_t last_discard_sample;
/** /**
* Number of internally decoded frames, used internally in libavformat, do not access * Number of internally decoded frames, used internally in libavformat, do not access
......
...@@ -219,8 +219,10 @@ static void mp3_parse_info_tag(AVFormatContext *s, AVStream *st, ...@@ -219,8 +219,10 @@ static void mp3_parse_info_tag(AVFormatContext *s, AVStream *st,
mp3->start_pad = v>>12; mp3->start_pad = v>>12;
mp3-> end_pad = v&4095; mp3-> end_pad = v&4095;
st->skip_samples = mp3->start_pad + 528 + 1; st->skip_samples = mp3->start_pad + 528 + 1;
if (mp3->frames) if (mp3->frames) {
st->end_discard_sample = -mp3->end_pad + 528 + 1 + mp3->frames * (int64_t)spf; st->first_discard_sample = -mp3->end_pad + 528 + 1 + mp3->frames * (int64_t)spf;
st->last_discard_sample = mp3->frames * (int64_t)spf;
}
if (!st->start_time) if (!st->start_time)
st->start_time = av_rescale_q(st->skip_samples, st->start_time = av_rescale_q(st->skip_samples,
(AVRational){1, c->sample_rate}, (AVRational){1, c->sample_rate},
......
...@@ -1358,13 +1358,14 @@ static int read_frame_internal(AVFormatContext *s, AVPacket *pkt) ...@@ -1358,13 +1358,14 @@ static int read_frame_internal(AVFormatContext *s, AVPacket *pkt)
if (ret >= 0) { if (ret >= 0) {
AVStream *st = s->streams[pkt->stream_index]; AVStream *st = s->streams[pkt->stream_index];
int discard_padding = 0; int discard_padding = 0;
if (st->end_discard_sample && pkt->pts != AV_NOPTS_VALUE) { if (st->first_discard_sample && pkt->pts != AV_NOPTS_VALUE) {
int64_t pts = pkt->pts - (is_relative(pkt->pts) ? RELATIVE_TS_BASE : 0); int64_t pts = pkt->pts - (is_relative(pkt->pts) ? RELATIVE_TS_BASE : 0);
int64_t sample = ts_to_samples(st, pts); int64_t sample = ts_to_samples(st, pts);
int duration = ts_to_samples(st, pkt->duration); int duration = ts_to_samples(st, pkt->duration);
int64_t end_sample = sample + duration; int64_t end_sample = sample + duration;
if (duration > 0 && end_sample >= st->end_discard_sample) if (duration > 0 && end_sample >= st->first_discard_sample &&
discard_padding = FFMIN(end_sample - st->end_discard_sample, duration); sample < st->last_discard_sample)
discard_padding = FFMIN(end_sample - st->first_discard_sample, duration);
} }
if (st->skip_samples || discard_padding) { if (st->skip_samples || discard_padding) {
uint8_t *p = av_packet_new_side_data(pkt, AV_PKT_DATA_SKIP_SAMPLES, 10); uint8_t *p = av_packet_new_side_data(pkt, AV_PKT_DATA_SKIP_SAMPLES, 10);
......
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