}
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);
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) {
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;
}
}
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;
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;
}
typedef struct AudioInterleaveContext {
AVFifoBuffer *fifo;
unsigned fifo_size; ///< size of currently allocated FIFO
+ int64_t n; ///< number of generated packets
+ int64_t nb_samples; ///< number of generated samples
uint64_t dts; ///< current dts
int sample_size; ///< size of one sample all channels included
- const int *samples_per_frame; ///< must be 0-terminated
- const int *samples; ///< current samples per frame, pointer to samples_per_frame
+ int samples_per_frame; ///< samples per frame if fixed, 0 otherwise
AVRational time_base; ///< time base of output audio packets
} AudioInterleaveContext;
-int ff_audio_interleave_init(AVFormatContext *s, const int *samples_per_frame, AVRational time_base);
+int ff_audio_interleave_init(AVFormatContext *s, const int samples_per_frame, AVRational time_base);
void ff_audio_interleave_close(AVFormatContext *s);
/**
avio_wb32(pb, KAG_SIZE); // system item size including klv fill
} else { // audio or data track
if (!audio_frame_size) {
- audio_frame_size = sc->aic.samples[0]*sc->aic.sample_size;
+ audio_frame_size = sc->frame_size;
audio_frame_size += klv_fill_size(audio_frame_size);
}
avio_w8(pb, 1);
return AVERROR(ENOMEM);
mxf->timecode_track->index = -1;
- if (!spf)
- spf = ff_mxf_get_samples_per_frame(s, (AVRational){ 1, 25 });
-
- if (ff_audio_interleave_init(s, spf->samples_per_frame, mxf->time_base) < 0)
+ if (ff_audio_interleave_init(s, 0, av_inv_q(mxf->tc.rate)) < 0)
return -1;
return 0;