* qualify a file. Refer to wsaud_probe() for the precise parameters.
*/
+#include "libavutil/channel_layout.h"
#include "libavutil/intreadwrite.h"
#include "avformat.h"
#include "internal.h"
#define AUD_CHUNK_PREAMBLE_SIZE 8
#define AUD_CHUNK_SIGNATURE 0x0000DEAF
-typedef struct WsAudDemuxContext {
- int audio_samplerate;
- int audio_channels;
- int audio_bits;
- enum CodecID audio_type;
- int audio_stream_index;
- int64_t audio_frame_counter;
-} WsAudDemuxContext;
-
static int wsaud_probe(AVProbeData *p)
{
int field;
/* note: only check for WS IMA (type 99) right now since there is no
* support for type 1 */
- if (p->buf[11] != 99)
+ if (p->buf[11] != 99 && p->buf[11] != 1)
return 0;
/* read ahead to the first audio chunk and validate the first header signature */
return 0;
/* return 1/2 certainty since this file check is a little sketchy */
- return AVPROBE_SCORE_MAX / 2;
+ return AVPROBE_SCORE_EXTENSION;
}
-static int wsaud_read_header(AVFormatContext *s,
- AVFormatParameters *ap)
+static int wsaud_read_header(AVFormatContext *s)
{
- WsAudDemuxContext *wsaud = s->priv_data;
AVIOContext *pb = s->pb;
AVStream *st;
unsigned char header[AUD_HEADER_SIZE];
+ int sample_rate, channels, codec;
if (avio_read(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE)
return AVERROR(EIO);
- wsaud->audio_samplerate = AV_RL16(&header[0]);
- if (header[11] == 99)
- wsaud->audio_type = CODEC_ID_ADPCM_IMA_WS;
- else
- return AVERROR_INVALIDDATA;
- /* flag 0 indicates stereo */
- wsaud->audio_channels = (header[10] & 0x1) + 1;
- /* flag 1 indicates 16 bit audio */
- wsaud->audio_bits = (((header[10] & 0x2) >> 1) + 1) * 8;
+ sample_rate = AV_RL16(&header[0]);
+ channels = (header[10] & 0x1) + 1;
+ codec = header[11];
/* initialize the audio decoder stream */
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
- avpriv_set_pts_info(st, 33, 1, wsaud->audio_samplerate);
- st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
- st->codec->codec_id = wsaud->audio_type;
- st->codec->codec_tag = 0; /* no tag */
- st->codec->channels = wsaud->audio_channels;
- st->codec->sample_rate = wsaud->audio_samplerate;
- st->codec->bits_per_coded_sample = wsaud->audio_bits;
- st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
- st->codec->bits_per_coded_sample / 4;
- st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
-
- wsaud->audio_stream_index = st->index;
- wsaud->audio_frame_counter = 0;
+
+ switch (codec) {
+ case 1:
+ if (channels != 1) {
+ avpriv_request_sample(s, "Stereo WS-SND1");
+ return AVERROR_PATCHWELCOME;
+ }
+ st->codec->codec_id = AV_CODEC_ID_WESTWOOD_SND1;
+ break;
+ case 99:
+ st->codec->codec_id = AV_CODEC_ID_ADPCM_IMA_WS;
+ st->codec->bits_per_coded_sample = 4;
+ st->codec->bit_rate = channels * sample_rate * 4;
+ break;
+ default:
+ avpriv_request_sample(s, "Unknown codec: %d", codec);
+ return AVERROR_PATCHWELCOME;
+ }
+ avpriv_set_pts_info(st, 64, 1, sample_rate);
+ st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
+ st->codec->channels = channels;
+ st->codec->channel_layout = channels == 1 ? AV_CH_LAYOUT_MONO :
+ AV_CH_LAYOUT_STEREO;
+ st->codec->sample_rate = sample_rate;
return 0;
}
static int wsaud_read_packet(AVFormatContext *s,
AVPacket *pkt)
{
- WsAudDemuxContext *wsaud = s->priv_data;
AVIOContext *pb = s->pb;
unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
unsigned int chunk_size;
int ret = 0;
+ AVStream *st = s->streams[0];
if (avio_read(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
AUD_CHUNK_PREAMBLE_SIZE)
return AVERROR_INVALIDDATA;
chunk_size = AV_RL16(&preamble[0]);
- ret= av_get_packet(pb, pkt, chunk_size);
- if (ret != chunk_size)
- return AVERROR(EIO);
- pkt->stream_index = wsaud->audio_stream_index;
- pkt->pts = wsaud->audio_frame_counter;
- pkt->pts /= wsaud->audio_samplerate;
- /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
- wsaud->audio_frame_counter += (chunk_size * 2) / wsaud->audio_channels;
+ if (st->codec->codec_id == AV_CODEC_ID_WESTWOOD_SND1) {
+ /* For Westwood SND1 audio we need to add the output size and input
+ size to the start of the packet to match what is in VQA.
+ Specifically, this is needed to signal when a packet should be
+ decoding as raw 8-bit pcm or variable-size ADPCM. */
+ int out_size = AV_RL16(&preamble[2]);
+ if ((ret = av_new_packet(pkt, chunk_size + 4)))
+ return ret;
+ if ((ret = avio_read(pb, &pkt->data[4], chunk_size)) != chunk_size)
+ return ret < 0 ? ret : AVERROR(EIO);
+ AV_WL16(&pkt->data[0], out_size);
+ AV_WL16(&pkt->data[2], chunk_size);
+
+ pkt->duration = out_size;
+ } else {
+ ret = av_get_packet(pb, pkt, chunk_size);
+ if (ret != chunk_size)
+ return AVERROR(EIO);
+
+ /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
+ pkt->duration = (chunk_size * 2) / st->codec->channels;
+ }
+ pkt->stream_index = st->index;
return ret;
}
AVInputFormat ff_wsaud_demuxer = {
.name = "wsaud",
- .long_name = NULL_IF_CONFIG_SMALL("Westwood Studios audio format"),
- .priv_data_size = sizeof(WsAudDemuxContext),
+ .long_name = NULL_IF_CONFIG_SMALL("Westwood Studios audio"),
.read_probe = wsaud_probe,
.read_header = wsaud_read_header,
.read_packet = wsaud_read_packet,
projects / ffmpeg / blobdiff