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correct AUDIO strf parsing patch by (Roman Shaposhnick <rvs at sun dot com>)

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Originally committed as revision 1664 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Roman Shaposhnik 2003-03-12 01:35:47 +00:00 committed by Michael Niedermayer
parent 69db4e10f2
commit 2e7973bbe7
4 changed files with 30 additions and 13 deletions
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6
libavformat/asf.c
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@ -806,7 +806,7 @@ static int asf_read_header(AVFormatContext *s, AVFormatParameters *ap)
asf->packet_size = asf->hdr.max_pktsize; asf->packet_size = asf->hdr.max_pktsize;
asf->nb_packets = asf->hdr.packets_count; asf->nb_packets = asf->hdr.packets_count;
} else if (!memcmp(&g, &stream_header, sizeof(GUID))) { } else if (!memcmp(&g, &stream_header, sizeof(GUID))) {
int type, total_size; int type, total_size, type_specific_size;
unsigned int tag1; unsigned int tag1;
int64_t pos1, pos2; int64_t pos1, pos2;
@ -832,7 +832,7 @@ static int asf_read_header(AVFormatContext *s, AVFormatParameters *ap)
} }
get_guid(pb, &g); get_guid(pb, &g);
total_size = get_le64(pb); total_size = get_le64(pb);
get_le32(pb); type_specific_size = get_le32(pb);
get_le32(pb); get_le32(pb);
st->id = get_le16(pb) & 0x7f; /* stream id */ st->id = get_le16(pb) & 0x7f; /* stream id */
// mapping of asf ID to AV stream ID; // mapping of asf ID to AV stream ID;
@ -842,7 +842,7 @@ static int asf_read_header(AVFormatContext *s, AVFormatParameters *ap)
st->codec.codec_type = type; st->codec.codec_type = type;
st->codec.frame_rate = 15 * s->pts_den / s->pts_num; // 15 fps default st->codec.frame_rate = 15 * s->pts_den / s->pts_num; // 15 fps default
if (type == CODEC_TYPE_AUDIO) { if (type == CODEC_TYPE_AUDIO) {
get_wav_header(pb, &st->codec, 1); get_wav_header(pb, &st->codec, type_specific_size);
/* We have to init the frame size at some point .... */ /* We have to init the frame size at some point .... */
pos2 = url_ftell(pb); pos2 = url_ftell(pb);
if (gsize > (pos2 + 8 - pos1 + 24)) { if (gsize > (pos2 + 8 - pos1 + 24)) {

3
libavformat/avi.h
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@ -18,8 +18,7 @@ typedef struct CodecTag {
void put_bmp_header(ByteIOContext *pb, AVCodecContext *enc, const CodecTag *tags, int for_asf); void put_bmp_header(ByteIOContext *pb, AVCodecContext *enc, const CodecTag *tags, int for_asf);
int put_wav_header(ByteIOContext *pb, AVCodecContext *enc); int put_wav_header(ByteIOContext *pb, AVCodecContext *enc);
int wav_codec_get_id(unsigned int tag, int bps); int wav_codec_get_id(unsigned int tag, int bps);
void get_wav_header(ByteIOContext *pb, AVCodecContext *codec, void get_wav_header(ByteIOContext *pb, AVCodecContext *codec, int size);
int has_extra_data);
extern const CodecTag codec_bmp_tags[]; extern const CodecTag codec_bmp_tags[];
extern const CodecTag codec_wav_tags[]; extern const CodecTag codec_wav_tags[];

2
libavformat/avidec.c
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@ -187,7 +187,7 @@ static int avi_read_header(AVFormatContext *s, AVFormatParameters *ap)
// url_fskip(pb, size - 5 * 4); // url_fskip(pb, size - 5 * 4);
break; break;
case CODEC_TYPE_AUDIO: case CODEC_TYPE_AUDIO:
get_wav_header(pb, &st->codec, (size >= 18)); get_wav_header(pb, &st->codec, size);
if (size%2) /* 2-aligned (fix for Stargate SG-1 - 3x18 - Shades of Grey.avi) */ if (size%2) /* 2-aligned (fix for Stargate SG-1 - 3x18 - Shades of Grey.avi) */
url_fskip(pb, 1); url_fskip(pb, 1);
break; break;

32
libavformat/wav.c
View File

@ -103,26 +103,44 @@ int put_wav_header(ByteIOContext *pb, AVCodecContext *enc)
return hdrsize; return hdrsize;
} }
void get_wav_header(ByteIOContext *pb, AVCodecContext *codec, /* We could be given one of the three possible structures here:
int has_extra_data) * WAVEFORMAT, PCMWAVEFORMAT or WAVEFORMATEX. Each structure
* is an expansion of the previous one with the fields added
* at the bottom. PCMWAVEFORMAT adds 'WORD wBitsPerSample' and
* WAVEFORMATEX adds 'WORD cbSize' and basically makes itself
* an openended structure.
*/
void get_wav_header(ByteIOContext *pb, AVCodecContext *codec, int size)
{ {
int id; int id;
id = get_le16(pb); id = get_le16(pb);
codec->codec_id = wav_codec_get_id(id, codec->frame_bits);
codec->codec_type = CODEC_TYPE_AUDIO; codec->codec_type = CODEC_TYPE_AUDIO;
codec->codec_tag = id; codec->codec_tag = id;
codec->channels = get_le16(pb); codec->channels = get_le16(pb);
codec->sample_rate = get_le32(pb); codec->sample_rate = get_le32(pb);
codec->bit_rate = get_le32(pb) * 8; codec->bit_rate = get_le32(pb) * 8;
codec->block_align = get_le16(pb); codec->block_align = get_le16(pb);
codec->bits_per_sample = get_le16(pb); /* bits per sample */ if (size == 14) { /* We're dealing with plain vanilla WAVEFORMAT */
codec->codec_id = wav_codec_get_id(id, codec->frame_bits); codec->bits_per_sample = 8;
if (has_extra_data) { return;
}
codec->bits_per_sample = get_le16(pb);
if (size > 16) { /* We're obviously dealing with WAVEFORMATEX */
codec->extradata_size = get_le16(pb); codec->extradata_size = get_le16(pb);
if (codec->extradata_size > 0) { if (codec->extradata_size > 0) {
if (codec->extradata_size > size - 18)
codec->extradata_size = size - 18;
codec->extradata = av_mallocz(codec->extradata_size); codec->extradata = av_mallocz(codec->extradata_size);
get_buffer(pb, codec->extradata, codec->extradata_size); get_buffer(pb, codec->extradata, codec->extradata_size);
} } else
codec->extradata_size = 0;
/* It is possible for the chunk to contain garbage at the end */
if (size - codec->extradata_size - 18 > 0)
url_fskip(pb, size - codec->extradata_size - 18);
} }
} }
@ -259,7 +277,7 @@ static int wav_read_header(AVFormatContext *s,
if (!st) if (!st)
return AVERROR_NOMEM; return AVERROR_NOMEM;
get_wav_header(pb, &st->codec, (size >= 18)); get_wav_header(pb, &st->codec, size);
size = find_tag(pb, MKTAG('d', 'a', 't', 'a')); size = find_tag(pb, MKTAG('d', 'a', 't', 'a'));
if (size < 0) if (size < 0)