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Code Issues Releases Activity

use sample rate as audio input time base

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Originally committed as revision 16985 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Baptiste Coudurier 2009-02-04 04:50:47 +00:00
parent 488227c5d7
commit 5a897cfa3c
1 changed files with 16 additions and 17 deletions
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33
libavformat/mxfenc.c
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@ -49,6 +49,7 @@ typedef struct {
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 const int *samples_per_frame; ///< must be 0 terminated
const int *samples; ///< current samples per frame, pointer to samples_per_frame const int *samples; ///< current samples per frame, pointer to samples_per_frame
AVRational time_base; ///< time base of output audio packets
} AudioInterleaveContext; } AudioInterleaveContext;
typedef struct { typedef struct {
@ -463,6 +464,7 @@ static void mxf_write_content_storage(AVFormatContext *s)
static void mxf_write_track(AVFormatContext *s, AVStream *st, enum MXFMetadataSetType type) static void mxf_write_track(AVFormatContext *s, AVStream *st, enum MXFMetadataSetType type)
{ {
MXFContext *mxf = s->priv_data;
ByteIOContext *pb = s->pb; ByteIOContext *pb = s->pb;
MXFStreamContext *sc = st->priv_data; MXFStreamContext *sc = st->priv_data;
@ -487,8 +489,8 @@ static void mxf_write_track(AVFormatContext *s, AVStream *st, enum MXFMetadataSe
put_buffer(pb, sc->track_essence_element_key + 12, 4); put_buffer(pb, sc->track_essence_element_key + 12, 4);
mxf_write_local_tag(pb, 8, 0x4B01); mxf_write_local_tag(pb, 8, 0x4B01);
put_be32(pb, st->time_base.den); put_be32(pb, mxf->time_base.den);
put_be32(pb, st->time_base.num); put_be32(pb, mxf->time_base.num);
// write origin // write origin
mxf_write_local_tag(pb, 8, 0x4B02); mxf_write_local_tag(pb, 8, 0x4B02);
@ -1059,7 +1061,9 @@ static int mxf_parse_mpeg2_frame(AVFormatContext *s, AVStream *st, AVPacket *pkt
return !!sc->codec_ul; return !!sc->codec_ul;
} }
static int ff_audio_interleave_init(AVFormatContext *s, const int *samples_per_frame) static int ff_audio_interleave_init(AVFormatContext *s,
const int *samples_per_frame,
AVRational time_base)
{ {
int i; int i;
@ -1079,6 +1083,7 @@ static int ff_audio_interleave_init(AVFormatContext *s, const int *samples_per_f
} }
aic->samples_per_frame = samples_per_frame; aic->samples_per_frame = samples_per_frame;
aic->samples = aic->samples_per_frame; aic->samples = aic->samples_per_frame;
aic->time_base = time_base;
av_fifo_init(&aic->fifo, 100 * *aic->samples); av_fifo_init(&aic->fifo, 100 * *aic->samples);
} }
@ -1130,11 +1135,13 @@ static int mxf_write_header(AVFormatContext *s)
return -1; return -1;
} }
mxf->edit_unit_start = st->index; mxf->edit_unit_start = st->index;
av_set_pts_info(st, 64, mxf->time_base.num, mxf->time_base.den);
} else if (st->codec->codec_type == CODEC_TYPE_AUDIO) { } else if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
if (st->codec->sample_rate != 48000) { if (st->codec->sample_rate != 48000) {
av_log(s, AV_LOG_ERROR, "only 48khz is implemented\n"); av_log(s, AV_LOG_ERROR, "only 48khz is implemented\n");
return -1; return -1;
} }
av_set_pts_info(st, 64, 1, st->codec->sample_rate);
} }
sc->duration = -1; sc->duration = -1;
@ -1159,7 +1166,6 @@ static int mxf_write_header(AVFormatContext *s)
for (i = 0; i < s->nb_streams; i++) { for (i = 0; i < s->nb_streams; i++) {
MXFStreamContext *sc = s->streams[i]->priv_data; MXFStreamContext *sc = s->streams[i]->priv_data;
av_set_pts_info(s->streams[i], 64, mxf->time_base.num, mxf->time_base.den);
// update element count // update element count
sc->track_essence_element_key[13] = present[sc->index]; sc->track_essence_element_key[13] = present[sc->index];
sc->order = AV_RB32(sc->track_essence_element_key+12); sc->order = AV_RB32(sc->track_essence_element_key+12);
@ -1168,7 +1174,7 @@ static int mxf_write_header(AVFormatContext *s)
if (!samples_per_frame) if (!samples_per_frame)
samples_per_frame = PAL_samples_per_frame; samples_per_frame = PAL_samples_per_frame;
if (ff_audio_interleave_init(s, samples_per_frame) < 0) if (ff_audio_interleave_init(s, samples_per_frame, mxf->time_base) < 0)
return -1; return -1;
return 0; return 0;
@ -1284,9 +1290,7 @@ static int mxf_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
av_fifo_read(&aic->fifo, pkt->data, size); av_fifo_read(&aic->fifo, pkt->data, size);
pkt->dts = pkt->pts = aic->dts; pkt->dts = pkt->pts = aic->dts;
pkt->duration = av_rescale_q(*aic->samples, pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
(AVRational){ 1, st->codec->sample_rate },
st->time_base);
pkt->stream_index = stream_index; pkt->stream_index = stream_index;
aic->dts += pkt->duration; aic->dts += pkt->duration;
@ -1353,16 +1357,11 @@ static int mxf_interleave_get_packet(AVFormatContext *s, AVPacket *out, AVPacket
static int mxf_compare_timestamps(AVFormatContext *s, AVPacket *next, AVPacket *pkt) static int mxf_compare_timestamps(AVFormatContext *s, AVPacket *next, AVPacket *pkt)
{ {
AVStream *st = s->streams[pkt ->stream_index]; MXFStreamContext *sc = s->streams[pkt ->stream_index]->priv_data;
AVStream *st2 = s->streams[next->stream_index]; MXFStreamContext *sc2 = s->streams[next->stream_index]->priv_data;
MXFStreamContext *sc = st ->priv_data;
MXFStreamContext *sc2 = st2->priv_data;
int64_t left = st2->time_base.num * (int64_t)st ->time_base.den; return next->dts > pkt->dts ||
int64_t right = st ->time_base.num * (int64_t)st2->time_base.den; (next->dts == pkt->dts && sc->order < sc2->order);
return next->dts * left > pkt->dts * right || // FIXME this can overflow
(next->dts * left == pkt->dts * right && sc->order < sc2->order);
} }
static int mxf_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush) static int mxf_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush)