audiodsp/x86: clear the high bits of the order parameter on 64bit

[ffmpeg] / libavformat / vocdec.c

diff --git a/libavformat/vocdec.c b/libavformat/vocdec.c
index 23f1b0d579ec02275f601678b40b5a92c6b95b13..5ed71171820a068251db9e9f0c10345de80e27ae 100644 (file)
--- a/libavformat/vocdec.c
+++ b/libavformat/vocdec.c
@@ -23,7 +23,6 @@
 #include "voc.h"
 #include "internal.h"
 
-
 static int voc_probe(AVProbeData *p)
 {
     int version, check;
@@ -38,12 +37,11 @@ static int voc_probe(AVProbeData *p)
     return AVPROBE_SCORE_MAX;
 }
 
-static int voc_read_header(AVFormatContext *s, AVFormatParameters *ap)
+static int voc_read_header(AVFormatContext *s)
 {
     VocDecContext *voc = s->priv_data;
     AVIOContext *pb = s->pb;
     int header_size;
-    AVStream *st;
 
     avio_skip(pb, 20);
     header_size = avio_rl16(pb) - 22;
@@ -52,116 +50,30 @@ static int voc_read_header(AVFormatContext *s, AVFormatParameters *ap)
         return AVERROR(ENOSYS);
     }
     avio_skip(pb, header_size);
-    st = av_new_stream(s, 0);
-    if (!st)
-        return AVERROR(ENOMEM);
-    st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
+
+    s->ctx_flags |= AVFMTCTX_NOHEADER;
 
     voc->remaining_size = 0;
     return 0;
 }
 
-int
-voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
-{
-    VocDecContext *voc = s->priv_data;
-    AVCodecContext *dec = st->codec;
-    AVIOContext *pb = s->pb;
-    VocType type;
-    int size, tmp_codec=-1;
-    int sample_rate = 0;
-    int channels = 1;
-
-    while (!voc->remaining_size) {
-        type = avio_r8(pb);
-        if (type == VOC_TYPE_EOF)
-            return AVERROR(EIO);
-        voc->remaining_size = avio_rl24(pb);
-        if (!voc->remaining_size) {
-            if (!s->pb->seekable)
-                return AVERROR(EIO);
-            voc->remaining_size = avio_size(pb) - avio_tell(pb);
-        }
-        max_size -= 4;
-
-        switch (type) {
-        case VOC_TYPE_VOICE_DATA:
-            dec->sample_rate = 1000000 / (256 - avio_r8(pb));
-            if (sample_rate)
-                dec->sample_rate = sample_rate;
-            dec->channels = channels;
-            tmp_codec = avio_r8(pb);
-            dec->bits_per_coded_sample = av_get_bits_per_sample(dec->codec_id);
-            voc->remaining_size -= 2;
-            max_size -= 2;
-            channels = 1;
-            break;
-
-        case VOC_TYPE_VOICE_DATA_CONT:
-            break;
-
-        case VOC_TYPE_EXTENDED:
-            sample_rate = avio_rl16(pb);
-            avio_r8(pb);
-            channels = avio_r8(pb) + 1;
-            sample_rate = 256000000 / (channels * (65536 - sample_rate));
-            voc->remaining_size = 0;
-            max_size -= 4;
-            break;
-
-        case VOC_TYPE_NEW_VOICE_DATA:
-            dec->sample_rate = avio_rl32(pb);
-            dec->bits_per_coded_sample = avio_r8(pb);
-            dec->channels = avio_r8(pb);
-            tmp_codec = avio_rl16(pb);
-            avio_skip(pb, 4);
-            voc->remaining_size -= 12;
-            max_size -= 12;
-            break;
-
-        default:
-            avio_skip(pb, voc->remaining_size);
-            max_size -= voc->remaining_size;
-            voc->remaining_size = 0;
-            break;
-        }
-    }
-
-    if (tmp_codec >= 0) {
-        tmp_codec = ff_codec_get_id(ff_voc_codec_tags, tmp_codec);
-        if (dec->codec_id == CODEC_ID_NONE)
-            dec->codec_id = tmp_codec;
-        else if (dec->codec_id != tmp_codec)
-            av_log(s, AV_LOG_WARNING, "Ignoring mid-stream change in audio codec\n");
-        if (dec->codec_id == CODEC_ID_NONE) {
-            if (s->audio_codec_id == CODEC_ID_NONE) {
-                av_log(s, AV_LOG_ERROR, "unknown codec tag\n");
-                return AVERROR(EINVAL);
-            }
-            av_log(s, AV_LOG_WARNING, "unknown codec tag\n");
-        }
-    }
-
-    dec->bit_rate = dec->sample_rate * dec->bits_per_coded_sample;
-
-    if (max_size <= 0)
-        max_size = 2048;
-    size = FFMIN(voc->remaining_size, max_size);
-    voc->remaining_size -= size;
-    return av_get_packet(pb, pkt, size);
-}
-
 static int voc_read_packet(AVFormatContext *s, AVPacket *pkt)
 {
-    return voc_get_packet(s, pkt, s->streams[0], 0);
+    if (!s->nb_streams) {
+        AVStream *st = avformat_new_stream(s, NULL);
+        if (!st)
+            return AVERROR(ENOMEM);
+        st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
+    }
+    return ff_voc_get_packet(s, pkt, s->streams[0], 0);
 }
 
 AVInputFormat ff_voc_demuxer = {
     .name           = "voc",
-    .long_name      = NULL_IF_CONFIG_SMALL("Creative Voice file format"),
+    .long_name      = NULL_IF_CONFIG_SMALL("Creative Voice"),
     .priv_data_size = sizeof(VocDecContext),
     .read_probe     = voc_probe,
     .read_header    = voc_read_header,
     .read_packet    = voc_read_packet,
-    .codec_tag=(const AVCodecTag* const []){ff_voc_codec_tags, 0},
+    .codec_tag      = (const AVCodecTag* const []){ ff_voc_codec_tags, 0 },
 };