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https://github.com/FFmpeg/FFmpeg.git
synced 2025-01-24 13:56:33 +02:00
8svx/iff: fix decoding of compressed stereo 8svx files.
Make the iff demuxer send the whole audio chunk to the decoder as a single packet and move stereo interleaving from the iff demuxer to the decoder. Based on a patch by Stefano Sabatini. git.videolan.org/ffmpeg.git commit e280a4da2ae6fd44f0079358ecc5aa08e388a5ed
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@ -32,8 +32,14 @@
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/** decoder context */
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typedef struct EightSvxContext {
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uint8_t fib_acc;
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uint8_t fib_acc[2];
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const int8_t *table;
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/* buffer used to store the whole first packet.
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data is only sent as one large packet */
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uint8_t *data[2];
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int data_size;
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int data_idx;
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} EightSvxContext;
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static const int8_t fibonacci[16] = { -34, -21, -13, -8, -5, -3, -2, -1,
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@ -41,6 +47,8 @@ static const int8_t fibonacci[16] = { -34, -21, -13, -8, -5, -3, -2, -1,
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static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1,
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0, 1, 2, 4, 8, 16, 32, 64 };
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#define MAX_FRAME_SIZE 32768
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/**
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* Delta decode the compressed values in src, and put the resulting
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* decoded samples in dst.
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@ -48,16 +56,18 @@ static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1,
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* @param[in,out] state starting value. it is saved for use in the next call.
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*/
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static void delta_decode(uint8_t *dst, const uint8_t *src, int src_size,
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uint8_t *state, const int8_t *table)
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uint8_t *state, const int8_t *table, int channels)
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{
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uint8_t val = *state;
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while (src_size--) {
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uint8_t d = *src++;
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val = av_clip_uint8(val + table[d & 0xF]);
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*dst++ = val;
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*dst = val;
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dst += channels;
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val = av_clip_uint8(val + table[d >> 4]);
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*dst++ = val;
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*dst = val;
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dst += channels;
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}
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*state = val;
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@ -67,33 +77,69 @@ static void delta_decode(uint8_t *dst, const uint8_t *src, int src_size,
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static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
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AVPacket *avpkt)
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{
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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EightSvxContext *esc = avctx->priv_data;
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int buf_size;
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uint8_t *out_data = data;
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int consumed = buf_size;
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int out_data_size;
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if(avctx->frame_number == 0) {
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if (buf_size < 2) {
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/* for the first packet, copy data to buffer */
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if (avpkt->data) {
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int chan_size = (avpkt->size / avctx->channels) - 2;
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if (avpkt->size < 2) {
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av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
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return AVERROR(EINVAL);
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}
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esc->fib_acc = (int8_t)buf[1] + 128;
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buf_size -= 2;
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buf += 2;
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if (esc->data[0]) {
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av_log(avctx, AV_LOG_ERROR, "unexpected data after first packet\n");
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return AVERROR(EINVAL);
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}
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esc->fib_acc[0] = avpkt->data[1] + 128;
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if (avctx->channels == 2)
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esc->fib_acc[1] = avpkt->data[2+chan_size+1] + 128;
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esc->data_idx = 0;
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esc->data_size = chan_size;
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if (!(esc->data[0] = av_malloc(chan_size)))
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return AVERROR(ENOMEM);
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if (avctx->channels == 2) {
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if (!(esc->data[1] = av_malloc(chan_size))) {
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av_freep(&esc->data[0]);
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return AVERROR(ENOMEM);
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}
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}
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memcpy(esc->data[0], &avpkt->data[2], chan_size);
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if (avctx->channels == 2)
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memcpy(esc->data[1], &avpkt->data[2+chan_size+2], chan_size);
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}
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if (!esc->data[0]) {
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av_log(avctx, AV_LOG_ERROR, "unexpected empty packet\n");
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return AVERROR(EINVAL);
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}
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if (*data_size < buf_size * 2) {
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/* decode next piece of data from the buffer */
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buf_size = FFMIN(MAX_FRAME_SIZE, esc->data_size - esc->data_idx);
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if (buf_size <= 0) {
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*data_size = 0;
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return avpkt->size;
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}
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out_data_size = buf_size * 2 * avctx->channels;
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if (*data_size < out_data_size) {
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av_log(avctx, AV_LOG_ERROR, "Provided buffer with size %d is too small.\n",
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*data_size);
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return AVERROR(EINVAL);
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}
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delta_decode(out_data, &esc->data[0][esc->data_idx], buf_size,
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&esc->fib_acc[0], esc->table, avctx->channels);
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if (avctx->channels == 2) {
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delta_decode(&out_data[1], &esc->data[1][esc->data_idx], buf_size,
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&esc->fib_acc[1], esc->table, avctx->channels);
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}
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esc->data_idx += buf_size;
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*data_size = out_data_size;
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delta_decode(out_data, buf, buf_size, &esc->fib_acc, esc->table);
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*data_size = buf_size * 2;
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return consumed;
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return avpkt->size;
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}
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/** initialize 8svx decoder */
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@ -101,6 +147,11 @@ static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
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{
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EightSvxContext *esc = avctx->priv_data;
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if (avctx->channels < 1 || avctx->channels > 2) {
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av_log(avctx, AV_LOG_ERROR, "8SVX does not support more than 2 channels\n");
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return AVERROR(EINVAL);
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}
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switch(avctx->codec->id) {
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case CODEC_ID_8SVX_FIB:
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esc->table = fibonacci;
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@ -115,13 +166,25 @@ static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
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return 0;
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}
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static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
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{
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EightSvxContext *esc = avctx->priv_data;
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av_freep(&esc->data[0]);
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av_freep(&esc->data[1]);
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return 0;
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}
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AVCodec ff_eightsvx_fib_decoder = {
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.name = "8svx_fib",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_8SVX_FIB,
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.priv_data_size = sizeof (EightSvxContext),
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.init = eightsvx_decode_init,
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.close = eightsvx_decode_close,
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.decode = eightsvx_decode_frame,
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.capabilities = CODEC_CAP_DELAY,
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.long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
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};
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@ -131,6 +194,8 @@ AVCodec ff_eightsvx_exp_decoder = {
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.id = CODEC_ID_8SVX_EXP,
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.priv_data_size = sizeof (EightSvxContext),
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.init = eightsvx_decode_init,
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.close = eightsvx_decode_close,
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.decode = eightsvx_decode_frame,
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.capabilities = CODEC_CAP_DELAY,
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.long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"),
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};
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@ -59,8 +59,6 @@
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#define RIGHT 4
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#define STEREO 6
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#define PACKET_SIZE 1024
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typedef enum {
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COMP_NONE,
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COMP_FIB,
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@ -76,22 +74,9 @@ typedef struct {
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uint64_t body_pos;
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uint32_t body_size;
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uint32_t sent_bytes;
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uint32_t audio_frame_count;
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} IffDemuxContext;
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static void interleave_stereo(const uint8_t *src, uint8_t *dest, int size)
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{
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uint8_t *end = dest + size;
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size = size>>1;
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while(dest < end) {
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*dest++ = *src;
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*dest++ = *(src+size);
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src++;
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}
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}
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/* Metadata string read */
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static int get_metadata(AVFormatContext *s,
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const char *const tag,
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@ -278,40 +263,20 @@ static int iff_read_packet(AVFormatContext *s,
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{
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IffDemuxContext *iff = s->priv_data;
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AVIOContext *pb = s->pb;
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AVStream *st = s->streams[0];
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int ret;
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if(iff->sent_bytes >= iff->body_size)
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return AVERROR(EIO);
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return AVERROR_EOF;
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if(st->codec->channels == 2) {
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uint8_t sample_buffer[PACKET_SIZE];
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ret = avio_read(pb, sample_buffer, PACKET_SIZE);
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if(av_new_packet(pkt, PACKET_SIZE) < 0) {
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av_log(s, AV_LOG_ERROR, "cannot allocate packet\n");
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return AVERROR(ENOMEM);
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}
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interleave_stereo(sample_buffer, pkt->data, PACKET_SIZE);
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} else if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
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ret = av_get_packet(pb, pkt, iff->body_size);
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} else {
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ret = av_get_packet(pb, pkt, PACKET_SIZE);
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}
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ret = av_get_packet(pb, pkt, iff->body_size);
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if (ret < 0)
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return ret;
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if(iff->sent_bytes == 0)
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pkt->flags |= AV_PKT_FLAG_KEY;
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iff->sent_bytes = iff->body_size;
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if(st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
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iff->sent_bytes += PACKET_SIZE;
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} else {
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iff->sent_bytes = iff->body_size;
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}
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pkt->stream_index = 0;
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if(st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
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pkt->pts = iff->audio_frame_count;
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iff->audio_frame_count += ret / st->codec->channels;
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}
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return ret;
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}
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