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avformat/audiointerleave: disallow using a samples_per_frame array

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Only MXF used an actual sample array, and that is unneeded there because simple
rounding rules can be used instead.

Signed-off-by: Marton Balint <cus@passwd.hu>
This commit is contained in:
Marton Balint
2020-02-28 00:26:20 +01:00
parent abbb466368
commit aef2016bb0
4 changed files with 18 additions and 22 deletions
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24
libavformat/audiointerleave.c
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@@ -39,14 +39,11 @@ void ff_audio_interleave_close(AVFormatContext *s)
}
int ff_audio_interleave_init(AVFormatContext *s,
const int *samples_per_frame,
const int samples_per_frame,
AVRational time_base)
{
int i;
if (!samples_per_frame)
return AVERROR(EINVAL);
if (!time_base.num) {
av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
return AVERROR(EINVAL);
@@ -56,6 +53,8 @@ int ff_audio_interleave_init(AVFormatContext *s,
AudioInterleaveContext *aic = st->priv_data;
if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
int max_samples = samples_per_frame ? samples_per_frame :
av_rescale_rnd(st->codecpar->sample_rate, time_base.num, time_base.den, AV_ROUND_UP);
aic->sample_size = (st->codecpar->channels *
av_get_bits_per_sample(st->codecpar->codec_id)) / 8;
if (!aic->sample_size) {
@@ -63,12 +62,11 @@ int ff_audio_interleave_init(AVFormatContext *s,
return AVERROR(EINVAL);
}
aic->samples_per_frame = samples_per_frame;
aic->samples = aic->samples_per_frame;
aic->time_base = time_base;
aic->fifo_size = 100* *aic->samples;
if (!(aic->fifo= av_fifo_alloc_array(100, *aic->samples)))
if (!(aic->fifo = av_fifo_alloc_array(100, max_samples)))
return AVERROR(ENOMEM);
aic->fifo_size = 100 * max_samples;
}
}
@@ -81,7 +79,9 @@ static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
AVStream *st = s->streams[stream_index];
AudioInterleaveContext *aic = st->priv_data;
int ret;
int frame_size = *aic->samples * aic->sample_size;
int nb_samples = aic->samples_per_frame ? aic->samples_per_frame :
(av_rescale_q(aic->n + 1, av_make_q(st->codecpar->sample_rate, 1), av_inv_q(aic->time_base)) - aic->nb_samples);
int frame_size = nb_samples * aic->sample_size;
int size = FFMIN(av_fifo_size(aic->fifo), frame_size);
if (!size || (!flush && size == av_fifo_size(aic->fifo)))
return 0;
@@ -95,13 +95,11 @@ static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
memset(pkt->data + size, 0, pkt->size - size);
pkt->dts = pkt->pts = aic->dts;
pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
pkt->duration = av_rescale_q(nb_samples, st->time_base, aic->time_base);
pkt->stream_index = stream_index;
aic->dts += pkt->duration;
aic->samples++;
if (!*aic->samples)
aic->samples = aic->samples_per_frame;
aic->nb_samples += nb_samples;
aic->n++;
return pkt->size;
}