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authorMarton Balint <cus@passwd.hu>2020-02-28 00:26:20 +0100
committerMarton Balint <cus@passwd.hu>2020-03-14 22:25:25 +0100
commitaef2016bb02fba377481789bf6a84e1176b83c25 (patch)
treee8946d1a1cfff1904a3b5dad2f7d732dfec45715
parentabbb466368c51285ca27d5e3959a16a9591e9a4c (diff)
downloadffmpeg-aef2016bb02fba377481789bf6a84e1176b83c25.tar.gz
avformat/audiointerleave: disallow using a samples_per_frame array
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>
-rw-r--r--libavformat/audiointerleave.c24
-rw-r--r--libavformat/audiointerleave.h7
-rw-r--r--libavformat/gxfenc.c2
-rw-r--r--libavformat/mxfenc.c7
4 files changed, 18 insertions, 22 deletions
diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c
index 6797546a44..2e83031bd6 100644
--- a/libavformat/audiointerleave.c
+++ b/libavformat/audiointerleave.c
@@ -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;
}
diff --git a/libavformat/audiointerleave.h b/libavformat/audiointerleave.h
index f28d5fefcc..0933310f4c 100644
--- a/libavformat/audiointerleave.h
+++ b/libavformat/audiointerleave.h
@@ -29,14 +29,15 @@
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);
/**
diff --git a/libavformat/gxfenc.c b/libavformat/gxfenc.c
index 9eebefc683..e7536a6a7e 100644
--- a/libavformat/gxfenc.c
+++ b/libavformat/gxfenc.c
@@ -663,7 +663,7 @@ static int gxf_write_umf_packet(AVFormatContext *s)
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)
{
diff --git a/libavformat/mxfenc.c b/libavformat/mxfenc.c
index 55c715d776..cbb4d9cc9a 100644
--- a/libavformat/mxfenc.c
+++ b/libavformat/mxfenc.c
@@ -1747,7 +1747,7 @@ static void mxf_write_index_table_segment(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);
@@ -2650,10 +2650,7 @@ static int mxf_write_header(AVFormatContext *s)
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;