extract audio interleaving code from mxf muxer, will be used by gxf and dv

Originally committed as revision 17038 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 125 insertions, 0 deletions

diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c
new file mode 100644
index 0000000000..e34026c408
--- /dev/null
+++ b/libavformat/audiointerleave.c
@@ -0,0 +1,125 @@
+/*
+ * 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);
+}