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https://github.com/librempeg/librempeg
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correct AUDIO strf parsing patch by (Roman Shaposhnick <rvs at sun dot com>)
Originally committed as revision 1664 to svn://svn.ffmpeg.org/ffmpeg/trunk
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69db4e10f2
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2e7973bbe7
@ -806,7 +806,7 @@ static int asf_read_header(AVFormatContext *s, AVFormatParameters *ap)
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asf->packet_size = asf->hdr.max_pktsize;
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asf->nb_packets = asf->hdr.packets_count;
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} else if (!memcmp(&g, &stream_header, sizeof(GUID))) {
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int type, total_size;
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int type, total_size, type_specific_size;
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unsigned int tag1;
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int64_t pos1, pos2;
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@ -832,7 +832,7 @@ static int asf_read_header(AVFormatContext *s, AVFormatParameters *ap)
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}
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get_guid(pb, &g);
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total_size = get_le64(pb);
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get_le32(pb);
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type_specific_size = get_le32(pb);
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get_le32(pb);
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st->id = get_le16(pb) & 0x7f; /* stream id */
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// mapping of asf ID to AV stream ID;
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@ -842,7 +842,7 @@ static int asf_read_header(AVFormatContext *s, AVFormatParameters *ap)
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st->codec.codec_type = type;
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st->codec.frame_rate = 15 * s->pts_den / s->pts_num; // 15 fps default
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if (type == CODEC_TYPE_AUDIO) {
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get_wav_header(pb, &st->codec, 1);
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get_wav_header(pb, &st->codec, type_specific_size);
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/* We have to init the frame size at some point .... */
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pos2 = url_ftell(pb);
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if (gsize > (pos2 + 8 - pos1 + 24)) {
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@ -18,8 +18,7 @@ typedef struct CodecTag {
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void put_bmp_header(ByteIOContext *pb, AVCodecContext *enc, const CodecTag *tags, int for_asf);
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int put_wav_header(ByteIOContext *pb, AVCodecContext *enc);
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int wav_codec_get_id(unsigned int tag, int bps);
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void get_wav_header(ByteIOContext *pb, AVCodecContext *codec,
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int has_extra_data);
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void get_wav_header(ByteIOContext *pb, AVCodecContext *codec, int size);
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extern const CodecTag codec_bmp_tags[];
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extern const CodecTag codec_wav_tags[];
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@ -187,7 +187,7 @@ static int avi_read_header(AVFormatContext *s, AVFormatParameters *ap)
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// url_fskip(pb, size - 5 * 4);
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break;
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case CODEC_TYPE_AUDIO:
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get_wav_header(pb, &st->codec, (size >= 18));
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get_wav_header(pb, &st->codec, size);
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if (size%2) /* 2-aligned (fix for Stargate SG-1 - 3x18 - Shades of Grey.avi) */
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url_fskip(pb, 1);
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break;
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@ -103,26 +103,44 @@ int put_wav_header(ByteIOContext *pb, AVCodecContext *enc)
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return hdrsize;
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}
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void get_wav_header(ByteIOContext *pb, AVCodecContext *codec,
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int has_extra_data)
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/* We could be given one of the three possible structures here:
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* WAVEFORMAT, PCMWAVEFORMAT or WAVEFORMATEX. Each structure
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* is an expansion of the previous one with the fields added
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* at the bottom. PCMWAVEFORMAT adds 'WORD wBitsPerSample' and
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* WAVEFORMATEX adds 'WORD cbSize' and basically makes itself
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* an openended structure.
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*/
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void get_wav_header(ByteIOContext *pb, AVCodecContext *codec, int size)
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{
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int id;
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id = get_le16(pb);
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codec->codec_id = wav_codec_get_id(id, codec->frame_bits);
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codec->codec_type = CODEC_TYPE_AUDIO;
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codec->codec_tag = id;
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codec->channels = get_le16(pb);
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codec->sample_rate = get_le32(pb);
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codec->bit_rate = get_le32(pb) * 8;
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codec->block_align = get_le16(pb);
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codec->bits_per_sample = get_le16(pb); /* bits per sample */
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codec->codec_id = wav_codec_get_id(id, codec->frame_bits);
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if (has_extra_data) {
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if (size == 14) { /* We're dealing with plain vanilla WAVEFORMAT */
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codec->bits_per_sample = 8;
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return;
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}
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codec->bits_per_sample = get_le16(pb);
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if (size > 16) { /* We're obviously dealing with WAVEFORMATEX */
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codec->extradata_size = get_le16(pb);
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if (codec->extradata_size > 0) {
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if (codec->extradata_size > size - 18)
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codec->extradata_size = size - 18;
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codec->extradata = av_mallocz(codec->extradata_size);
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get_buffer(pb, codec->extradata, codec->extradata_size);
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}
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} else
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codec->extradata_size = 0;
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/* It is possible for the chunk to contain garbage at the end */
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if (size - codec->extradata_size - 18 > 0)
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url_fskip(pb, size - codec->extradata_size - 18);
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}
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}
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@ -259,7 +277,7 @@ static int wav_read_header(AVFormatContext *s,
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if (!st)
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return AVERROR_NOMEM;
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get_wav_header(pb, &st->codec, (size >= 18));
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get_wav_header(pb, &st->codec, size);
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size = find_tag(pb, MKTAG('d', 'a', 't', 'a'));
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if (size < 0)
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