lavf: use conditional notation for default codec in muxer declarations.

[ffmpeg] / libavformat / oggparsespeex.c

diff --git a/libavformat/oggparsespeex.c b/libavformat/oggparsespeex.c
index cc00dd2207871e73f05e2c65c6822ddfc67ade19..435ed0cedf9ca5617aca42d44c6f201053a7eedd 100644 (file)
--- a/libavformat/oggparsespeex.c
+++ b/libavformat/oggparsespeex.c
@@ -28,49 +28,99 @@
 #include "libavcodec/get_bits.h"
 #include "libavcodec/bytestream.h"
 #include "avformat.h"
+#include "internal.h"
 #include "oggdec.h"
 
+struct speex_params {
+    int packet_size;
+    int final_packet_duration;
+    int seq;
+};
+
 static int speex_header(AVFormatContext *s, int idx) {
     struct ogg *ogg = s->priv_data;
     struct ogg_stream *os = ogg->streams + idx;
+    struct speex_params *spxp = os->private;
     AVStream *st = s->streams[idx];
     uint8_t *p = os->buf + os->pstart;
 
-    if (os->seq > 1)
+    if (!spxp) {
+        spxp = av_mallocz(sizeof(*spxp));
+        os->private = spxp;
+    }
+
+    if (spxp->seq > 1)
         return 0;
 
-    if (os->seq == 0) {
+    if (spxp->seq == 0) {
         int frames_per_packet;
-        st->codec->codec_type = CODEC_TYPE_AUDIO;
+        st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
         st->codec->codec_id = CODEC_ID_SPEEX;
 
         st->codec->sample_rate = AV_RL32(p + 36);
         st->codec->channels = AV_RL32(p + 48);
 
-        /* We treat the whole Speex packet as a single frame everywhere Speex
-           is handled in FFmpeg.  This avoids the complexities of splitting
-           and joining individual Speex frames, which are not always
-           byte-aligned. */
-        st->codec->frame_size = AV_RL32(p + 56);
-        frames_per_packet     = AV_RL32(p + 64);
+        spxp->packet_size  = AV_RL32(p + 56);
+        frames_per_packet  = AV_RL32(p + 64);
         if (frames_per_packet)
-            st->codec->frame_size *= frames_per_packet;
+            spxp->packet_size *= frames_per_packet;
 
         st->codec->extradata_size = os->psize;
         st->codec->extradata = av_malloc(st->codec->extradata_size
                                          + FF_INPUT_BUFFER_PADDING_SIZE);
         memcpy(st->codec->extradata, p, st->codec->extradata_size);
 
-        st->time_base.num = 1;
-        st->time_base.den = st->codec->sample_rate;
+        avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
     } else
-        vorbis_comment(s, p, os->psize);
+        ff_vorbis_comment(s, &st->metadata, p, os->psize);
 
+    spxp->seq++;
     return 1;
 }
 
+static int ogg_page_packets(struct ogg_stream *os)
+{
+    int i;
+    int packets = 0;
+    for (i = 0; i < os->nsegs; i++)
+        if (os->segments[i] < 255)
+            packets++;
+    return packets;
+}
+
+static int speex_packet(AVFormatContext *s, int idx)
+{
+    struct ogg *ogg = s->priv_data;
+    struct ogg_stream *os = ogg->streams + idx;
+    struct speex_params *spxp = os->private;
+    int packet_size = spxp->packet_size;
+
+    if (os->flags & OGG_FLAG_EOS && os->lastpts != AV_NOPTS_VALUE &&
+        os->granule > 0) {
+        /* first packet of final page. we have to calculate the final packet
+           duration here because it is the only place we know the next-to-last
+           granule position. */
+        spxp->final_packet_duration = os->granule - os->lastpts -
+                                      packet_size * (ogg_page_packets(os) - 1);
+    }
+
+    if (!os->lastpts && os->granule > 0)
+        /* first packet */
+        os->lastpts = os->lastdts = os->granule - packet_size *
+                                    ogg_page_packets(os);
+    if (os->flags & OGG_FLAG_EOS && os->segp == os->nsegs &&
+        spxp->final_packet_duration)
+        /* final packet */
+        os->pduration = spxp->final_packet_duration;
+    else
+        os->pduration = packet_size;
+
+    return 0;
+}
+
 const struct ogg_codec ff_speex_codec = {
     .magic = "Speex   ",
     .magicsize = 8,
-    .header = speex_header
+    .header = speex_header,
+    .packet = speex_packet
 };