OBJS-$(CONFIG_MTV_DEMUXER) += mtv.o
OBJS-$(CONFIG_MVI_DEMUXER) += mvi.o
OBJS-$(CONFIG_MXF_DEMUXER) += mxfdec.o mxf.o
-OBJS-$(CONFIG_MXF_MUXER) += mxfenc.o mxf.o
+OBJS-$(CONFIG_MXF_MUXER) += mxfenc.o mxf.o audiointerleave.o
OBJS-$(CONFIG_NSV_DEMUXER) += nsvdec.o
OBJS-$(CONFIG_NULL_MUXER) += raw.o
OBJS-$(CONFIG_NUT_DEMUXER) += nutdec.o nut.o riff.o
--- /dev/null
+/*
+ * Audio Interleaving functions
+ *
+ * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/fifo.h"
+#include "avformat.h"
+#include "audiointerleave.h"
+
+void ff_audio_interleave_close(AVFormatContext *s)
+{
+ int i;
+ for (i = 0; i < s->nb_streams; i++) {
+ AVStream *st = s->streams[i];
+ AudioInterleaveContext *aic = st->priv_data;
+
+ if (st->codec->codec_type == CODEC_TYPE_AUDIO)
+ av_fifo_free(&aic->fifo);
+ }
+}
+
+int ff_audio_interleave_init(AVFormatContext *s,
+ const int *samples_per_frame,
+ AVRational time_base)
+{
+ int i;
+
+ if (!samples_per_frame)
+ return -1;
+
+ for (i = 0; i < s->nb_streams; i++) {
+ AVStream *st = s->streams[i];
+ AudioInterleaveContext *aic = st->priv_data;
+
+ if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+ aic->sample_size = (st->codec->channels *
+ av_get_bits_per_sample(st->codec->codec_id)) / 8;
+ if (!aic->sample_size) {
+ av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
+ return -1;
+ }
+ aic->samples_per_frame = samples_per_frame;
+ aic->samples = aic->samples_per_frame;
+ aic->time_base = time_base;
+
+ av_fifo_init(&aic->fifo, 100 * *aic->samples);
+ }
+ }
+
+ return 0;
+}
+
+int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
+ int stream_index, int flush)
+{
+ AVStream *st = s->streams[stream_index];
+ AudioInterleaveContext *aic = st->priv_data;
+
+ int size = FFMIN(av_fifo_size(&aic->fifo), *aic->samples * aic->sample_size);
+ if (!size || (!flush && size == av_fifo_size(&aic->fifo)))
+ return 0;
+
+ av_new_packet(pkt, size);
+ av_fifo_read(&aic->fifo, pkt->data, size);
+
+ pkt->dts = pkt->pts = aic->dts;
+ pkt->duration = av_rescale_q(*aic->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;
+
+ return size;
+}
+
+int ff_audio_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
+ int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
+ int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
+{
+ int i;
+
+ if (pkt) {
+ AVStream *st = s->streams[pkt->stream_index];
+ AudioInterleaveContext *aic = st->priv_data;
+ if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+ av_fifo_generic_write(&aic->fifo, pkt->data, pkt->size, NULL);
+ } else {
+ // rewrite pts and dts to be decoded time line position
+ pkt->dts = aic->dts;
+ aic->dts += pkt->duration;
+ ff_interleave_add_packet(s, pkt, compare_ts);
+ }
+ pkt = NULL;
+ }
+
+ for (i = 0; i < s->nb_streams; i++) {
+ AVStream *st = s->streams[i];
+ if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+ AVPacket new_pkt;
+ while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush))
+ ff_interleave_add_packet(s, &new_pkt, compare_ts);
+ }
+ }
+
+ return get_packet(s, out, pkt, flush);
+}
--- /dev/null
+/*
+ * Audio Interleaving prototypes and declarations
+ *
+ * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVFORMAT_AUDIOINTERLEAVE_H
+#define AVFORMAT_AUDIOINTERLEAVE_H
+
+#include "libavutil/fifo.h"
+#include "avformat.h"
+
+typedef struct {
+ AVFifoBuffer fifo;
+ unsigned fifo_size; ///< current fifo size allocated
+ 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
+ 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);
+void ff_audio_interleave_close(AVFormatContext *s);
+
+int ff_interleave_compare_dts(AVFormatContext *s, AVPacket *next, AVPacket *pkt);
+int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
+ int stream_index, int flush);
+int ff_audio_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
+ int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
+ int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *));
+
+#endif // AVFORMAT_AUDIOINTERLEAVE_H
#include <time.h>
#include "libavutil/fifo.h"
+#include "audiointerleave.h"
#include "mxf.h"
static const int NTSC_samples_per_frame[] = { 1602, 1601, 1602, 1601, 1602, 0 };
#define MXF_INDEX_CLUSTER_SIZE 4096
#define KAG_SIZE 512
-typedef struct {
- AVFifoBuffer fifo;
- unsigned fifo_size; ///< current fifo size allocated
- 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
- AVRational time_base; ///< time base of output audio packets
-} AudioInterleaveContext;
-
typedef struct {
int local_tag;
UID uid;
return !!sc->codec_ul;
}
-static int ff_audio_interleave_init(AVFormatContext *s,
- const int *samples_per_frame,
- AVRational time_base)
-{
- int i;
-
- if (!samples_per_frame)
- return -1;
-
- for (i = 0; i < s->nb_streams; i++) {
- AVStream *st = s->streams[i];
- AudioInterleaveContext *aic = st->priv_data;
-
- if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
- aic->sample_size = (st->codec->channels *
- av_get_bits_per_sample(st->codec->codec_id)) / 8;
- if (!aic->sample_size) {
- av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
- return -1;
- }
- aic->samples_per_frame = samples_per_frame;
- aic->samples = aic->samples_per_frame;
- aic->time_base = time_base;
-
- av_fifo_init(&aic->fifo, 100 * *aic->samples);
- }
- }
-
- return 0;
-}
-
-static void ff_audio_interleave_close(AVFormatContext *s)
-{
- int i;
- for (i = 0; i < s->nb_streams; i++) {
- AVStream *st = s->streams[i];
- AudioInterleaveContext *aic = st->priv_data;
-
- if (st->codec->codec_type == CODEC_TYPE_AUDIO)
- av_fifo_free(&aic->fifo);
- }
-}
-
static uint64_t mxf_parse_timestamp(time_t timestamp)
{
struct tm *time = localtime(×tamp);
return 0;
}
-static int mxf_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
- int stream_index, int flush)
-{
- AVStream *st = s->streams[stream_index];
- AudioInterleaveContext *aic = st->priv_data;
-
- int size = FFMIN(av_fifo_size(&aic->fifo), *aic->samples * aic->sample_size);
- if (!size || (!flush && size == av_fifo_size(&aic->fifo)))
- return 0;
-
- av_new_packet(pkt, size);
- av_fifo_read(&aic->fifo, pkt->data, size);
-
- pkt->dts = pkt->pts = aic->dts;
- pkt->duration = av_rescale_q(*aic->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;
-
- return size;
-}
-
static int mxf_interleave_get_packet(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush)
{
AVPacketList *pktl;
static int mxf_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush)
{
- int i;
-
- if (pkt) {
- AVStream *st = s->streams[pkt->stream_index];
- AudioInterleaveContext *aic = st->priv_data;
- if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
- av_fifo_generic_write(&aic->fifo, pkt->data, pkt->size, NULL);
- } else {
- // rewrite pts and dts to be decoded time line position
- pkt->pts = pkt->dts = aic->dts;
- aic->dts += pkt->duration;
- ff_interleave_add_packet(s, pkt, mxf_compare_timestamps);
- }
- pkt = NULL;
- }
-
- for (i = 0; i < s->nb_streams; i++) {
- AVStream *st = s->streams[i];
- if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
- AVPacket new_pkt;
- while (mxf_interleave_new_audio_packet(s, &new_pkt, i, flush))
- ff_interleave_add_packet(s, &new_pkt, mxf_compare_timestamps);
- }
- }
-
- return mxf_interleave_get_packet(s, out, pkt, flush);
+ return ff_audio_interleave(s, out, pkt, flush,
+ mxf_interleave_get_packet, mxf_compare_timestamps);
}
AVOutputFormat mxf_muxer = {
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