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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
View File

@ -39,14 +39,11 @@ void ff_audio_interleave_close(AVFormatContext *s)
} }
int ff_audio_interleave_init(AVFormatContext *s, int ff_audio_interleave_init(AVFormatContext *s,
const int *samples_per_frame, const int samples_per_frame,
AVRational time_base) AVRational time_base)
{ {
int i; int i;
if (!samples_per_frame)
return AVERROR(EINVAL);
if (!time_base.num) { if (!time_base.num) {
av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n"); av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
return AVERROR(EINVAL); return AVERROR(EINVAL);
@ -56,6 +53,8 @@ int ff_audio_interleave_init(AVFormatContext *s,
AudioInterleaveContext *aic = st->priv_data; AudioInterleaveContext *aic = st->priv_data;
if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { 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 * aic->sample_size = (st->codecpar->channels *
av_get_bits_per_sample(st->codecpar->codec_id)) / 8; av_get_bits_per_sample(st->codecpar->codec_id)) / 8;
if (!aic->sample_size) { if (!aic->sample_size) {
@ -63,12 +62,11 @@ int ff_audio_interleave_init(AVFormatContext *s,
return AVERROR(EINVAL); return AVERROR(EINVAL);
} }
aic->samples_per_frame = samples_per_frame; aic->samples_per_frame = samples_per_frame;
aic->samples = aic->samples_per_frame;
aic->time_base = time_base; aic->time_base = time_base;
aic->fifo_size = 100* *aic->samples; if (!(aic->fifo = av_fifo_alloc_array(100, max_samples)))
if (!(aic->fifo= av_fifo_alloc_array(100, *aic->samples)))
return AVERROR(ENOMEM); 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]; AVStream *st = s->streams[stream_index];
AudioInterleaveContext *aic = st->priv_data; AudioInterleaveContext *aic = st->priv_data;
int ret; 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); int size = FFMIN(av_fifo_size(aic->fifo), frame_size);
if (!size || (!flush && size == av_fifo_size(aic->fifo))) if (!size || (!flush && size == av_fifo_size(aic->fifo)))
return 0; return 0;
@ -95,13 +95,11 @@ static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
memset(pkt->data + size, 0, pkt->size - size); memset(pkt->data + size, 0, pkt->size - size);
pkt->dts = pkt->pts = aic->dts; 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; pkt->stream_index = stream_index;
aic->dts += pkt->duration; aic->dts += pkt->duration;
aic->nb_samples += nb_samples;
aic->samples++; aic->n++;
if (!*aic->samples)
aic->samples = aic->samples_per_frame;
return pkt->size; return pkt->size;
} }

7
libavformat/audiointerleave.h
View File

@ -29,14 +29,15 @@
typedef struct AudioInterleaveContext { typedef struct AudioInterleaveContext {
AVFifoBuffer *fifo; AVFifoBuffer *fifo;
unsigned fifo_size; ///< size of currently allocated 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 uint64_t dts; ///< current dts
int sample_size; ///< size of one sample all channels included int sample_size; ///< size of one sample all channels included
const int *samples_per_frame; ///< must be 0-terminated int samples_per_frame; ///< samples per frame if fixed, 0 otherwise
const int *samples; ///< current samples per frame, pointer to samples_per_frame
AVRational time_base; ///< time base of output audio packets AVRational time_base; ///< time base of output audio packets
} AudioInterleaveContext; } 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); void ff_audio_interleave_close(AVFormatContext *s);
/** /**

2
libavformat/gxfenc.c
View File

@ -663,7 +663,7 @@ static int gxf_write_umf_packet(AVFormatContext *s)
return updatePacketSize(pb, pos); return updatePacketSize(pb, pos);
} }
static const int GXF_samples_per_frame[] = { 32768, 0 }; static const int GXF_samples_per_frame = 32768;
static void gxf_init_timecode_track(GXFStreamContext *sc, GXFStreamContext *vsc) static void gxf_init_timecode_track(GXFStreamContext *sc, GXFStreamContext *vsc)
{ {

7
libavformat/mxfenc.c
View File

@ -1747,7 +1747,7 @@ static void mxf_write_index_table_segment(AVFormatContext *s)
avio_wb32(pb, KAG_SIZE); // system item size including klv fill avio_wb32(pb, KAG_SIZE); // system item size including klv fill
} else { // audio or data track } else { // audio or data track
if (!audio_frame_size) { 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); audio_frame_size += klv_fill_size(audio_frame_size);
} }
avio_w8(pb, 1); avio_w8(pb, 1);
@ -2650,10 +2650,7 @@ static int mxf_write_header(AVFormatContext *s)
return AVERROR(ENOMEM); return AVERROR(ENOMEM);
mxf->timecode_track->index = -1; mxf->timecode_track->index = -1;
if (!spf) if (ff_audio_interleave_init(s, 0, av_inv_q(mxf->tc.rate)) < 0)
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)
return -1; return -1;
return 0; return 0;