const CodecTag codec_wav_tags[] = {
{ CODEC_ID_MP2, 0x50 },
- { CODEC_ID_MP3LAME, 0x55 },
+ { CODEC_ID_MP3, 0x55 },
{ CODEC_ID_AC3, 0x2000 },
{ CODEC_ID_PCM_S16LE, 0x01 },
{ CODEC_ID_PCM_U8, 0x01 }, /* must come after s16le in this list */
{ CODEC_ID_PCM_MULAW, 0x07 },
{ CODEC_ID_ADPCM_MS, 0x02 },
{ CODEC_ID_ADPCM_IMA_WAV, 0x11 },
+ { CODEC_ID_ADPCM_G726, 0x45 },
+ { CODEC_ID_ADPCM_IMA_DK4, 0x61 }, /* rogue format number */
+ { CODEC_ID_ADPCM_IMA_DK3, 0x62 }, /* rogue format number */
{ CODEC_ID_WMAV1, 0x160 },
{ CODEC_ID_WMAV2, 0x161 },
+ { CODEC_ID_AAC, 0x706d },
+ { CODEC_ID_VORBIS, ('V'<<8)+'o' }, //HACK/FIXME, does vorbis in WAV/AVI have an (in)official id?
+ { CODEC_ID_SONIC, 0x2048 },
+ { CODEC_ID_SONIC_LS, 0x2048 },
+ { CODEC_ID_ADPCM_CT, 0x200 },
{ 0, 0 },
};
+#ifdef CONFIG_ENCODERS
/* WAVEFORMATEX header */
/* returns the size or -1 on error */
int put_wav_header(ByteIOContext *pb, AVCodecContext *enc)
{
- int tag, bps, blkalign, bytespersec;
+ int bps, blkalign, bytespersec;
int hdrsize = 18;
- tag = codec_get_tag(codec_wav_tags, enc->codec_id);
- if (tag == 0)
+ if(!enc->codec_tag)
+ enc->codec_tag = codec_get_tag(codec_wav_tags, enc->codec_id);
+ if(!enc->codec_tag)
return -1;
- put_le16(pb, tag);
+
+ put_le16(pb, enc->codec_tag);
put_le16(pb, enc->channels);
put_le32(pb, enc->sample_rate);
if (enc->codec_id == CODEC_ID_PCM_U8 ||
enc->codec_id == CODEC_ID_PCM_ALAW ||
enc->codec_id == CODEC_ID_PCM_MULAW) {
bps = 8;
- } else if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3LAME) {
+ } else if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3) {
bps = 0;
} else if (enc->codec_id == CODEC_ID_ADPCM_IMA_WAV || enc->codec_id == CODEC_ID_ADPCM_MS) {
bps = 4;
bps = 16;
}
- if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3LAME) {
- blkalign = 1;
+ if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3) {
+ blkalign = enc->frame_size; //this is wrong, but seems many demuxers dont work if this is set correctly
//blkalign = 144 * enc->bit_rate/enc->sample_rate;
} else if (enc->block_align != 0) { /* specified by the codec */
blkalign = enc->block_align;
put_le32(pb, bytespersec); /* bytes per second */
put_le16(pb, blkalign); /* block align */
put_le16(pb, bps); /* bits per sample */
- if (enc->codec_id == CODEC_ID_MP3LAME) {
+ if (enc->codec_id == CODEC_ID_MP3) {
put_le16(pb, 12); /* wav_extra_size */
hdrsize += 12;
put_le16(pb, 1); /* wID */
put_le32(pb, 0); /* dwPTSHigh */
} else if (enc->codec_id == CODEC_ID_ADPCM_IMA_WAV) {
put_le16(pb, 2); /* wav_extra_size */
+ hdrsize += 2;
put_le16(pb, ((enc->block_align - 4 * enc->channels) / (4 * enc->channels)) * 8 + 1); /* wSamplesPerBlock */
- } else
- put_le16(pb, 0); /* wav_extra_size */
+ } else if(enc->extradata_size){
+ put_le16(pb, enc->extradata_size);
+ put_buffer(pb, enc->extradata, enc->extradata_size);
+ hdrsize += enc->extradata_size;
+ if(hdrsize&1){
+ hdrsize++;
+ put_byte(pb, 0);
+ }
+ } else {
+ hdrsize -= 2;
+ }
return hdrsize;
}
-
-void get_wav_header(ByteIOContext *pb, AVCodecContext *codec,
- int has_extra_data)
+#endif //CONFIG_ENCODERS
+
+/* We could be given one of the three possible structures here:
+ * 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;
id = get_le16(pb);
codec->codec_type = CODEC_TYPE_AUDIO;
codec->codec_tag = id;
- codec->fourcc = id;
codec->channels = get_le16(pb);
codec->sample_rate = get_le32(pb);
codec->bit_rate = get_le32(pb) * 8;
codec->block_align = get_le16(pb);
- codec->frame_bits = get_le16(pb); /* bits per sample */
- codec->codec_id = wav_codec_get_id(id, codec->frame_bits);
- if (has_extra_data) {
+ if (size == 14) { /* We're dealing with plain vanilla WAVEFORMAT */
+ codec->bits_per_sample = 8;
+ }else
+ codec->bits_per_sample = get_le16(pb);
+ codec->codec_id = wav_codec_get_id(id, codec->bits_per_sample);
+
+ if (size > 16) { /* We're obviously dealing with WAVEFORMATEX */
codec->extradata_size = get_le16(pb);
if (codec->extradata_size > 0) {
- codec->extradata = av_mallocz(codec->extradata_size);
+ if (codec->extradata_size > size - 18)
+ codec->extradata_size = size - 18;
+ codec->extradata = av_mallocz(codec->extradata_size + FF_INPUT_BUFFER_PADDING_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);
}
}
return id;
}
+#ifdef CONFIG_ENCODERS
typedef struct {
offset_t data;
} WAVContext;
return 0;
}
-static int wav_write_packet(AVFormatContext *s, int stream_index_ptr,
- uint8_t *buf, int size, int force_pts)
+static int wav_write_packet(AVFormatContext *s, AVPacket *pkt)
{
ByteIOContext *pb = &s->pb;
- put_buffer(pb, buf, size);
+ put_buffer(pb, pkt->data, pkt->size);
return 0;
}
}
return 0;
}
+#endif //CONFIG_ENCODERS
/* return the size of the found tag */
/* XXX: > 2GB ? */
if (!st)
return AVERROR_NOMEM;
- get_wav_header(pb, &st->codec, (size >= 18));
-
+ get_wav_header(pb, &st->codec, size);
+ st->need_parsing = 1;
+
+ av_set_pts_info(st, 64, 1, st->codec.sample_rate);
+
size = find_tag(pb, MKTAG('d', 'a', 't', 'a'));
if (size < 0)
return -1;
static int wav_read_packet(AVFormatContext *s,
AVPacket *pkt)
{
- int ret;
+ int ret, size;
+ AVStream *st;
if (url_feof(&s->pb))
- return -EIO;
- if (av_new_packet(pkt, MAX_SIZE))
- return -EIO;
+ return AVERROR_IO;
+ st = s->streams[0];
+
+ size = MAX_SIZE;
+ if (st->codec.block_align > 1) {
+ if (size < st->codec.block_align)
+ size = st->codec.block_align;
+ size = (size / st->codec.block_align) * st->codec.block_align;
+ }
+ if (av_new_packet(pkt, size))
+ return AVERROR_IO;
pkt->stream_index = 0;
ret = get_buffer(&s->pb, pkt->data, pkt->size);
return 0;
}
+static int wav_read_seek(AVFormatContext *s,
+ int stream_index, int64_t timestamp, int flags)
+{
+ AVStream *st;
+
+ st = s->streams[0];
+ switch(st->codec.codec_id) {
+ case CODEC_ID_MP2:
+ case CODEC_ID_MP3:
+ case CODEC_ID_AC3:
+ case CODEC_ID_DTS:
+ /* use generic seeking with dynamically generated indexes */
+ return -1;
+ default:
+ break;
+ }
+ return pcm_read_seek(s, stream_index, timestamp, flags);
+}
+
+
static AVInputFormat wav_iformat = {
"wav",
"wav format",
wav_read_header,
wav_read_packet,
wav_read_close,
+ wav_read_seek,
};
+#ifdef CONFIG_ENCODERS
static AVOutputFormat wav_oformat = {
"wav",
"wav format",
wav_write_packet,
wav_write_trailer,
};
+#endif //CONFIG_ENCODERS
-int wav_init(void)
+int ff_wav_init(void)
{
av_register_input_format(&wav_iformat);
+#ifdef CONFIG_ENCODERS
av_register_output_format(&wav_oformat);
+#endif //CONFIG_ENCODERS
return 0;
}
projects / ffmpeg / blobdiff