static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
unsigned char *start=NULL;
int y,len=-1;
while(len<minlen){
AVPacket pkt;
int len2=maxlen;
double pts;
int x=ds_get_packet_pts(sh_audio->ds,&start, &pts);
if(x<=0) {
start = NULL;
x = 0;
ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0);
if (x <= 0)
break; // error
} else {
int in_size = x;
int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0);
sh_audio->ds->buffer_pos -= in_size - consumed;
}
av_init_packet(&pkt);
pkt.data = start;
pkt.size = x;
if (pts != MP_NOPTS_VALUE) {
sh_audio->pts = pts;
sh_audio->pts_bytes = 0;
}
y=avcodec_decode_audio3(sh_audio->context,(int16_t*)buf,&len2,&pkt);
//printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout);
// LATM may need many packets to find mux info
if (y == AVERROR(EAGAIN))
continue;
if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
if(!sh_audio->parser && y<x)
sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!)
if(len2>0){
if (((AVCodecContext *)sh_audio->context)->channels >= 5) {
int samplesize = av_get_bytes_per_sample(((AVCodecContext *)
sh_audio->context)->sample_fmt);
reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
((AVCodecContext *)sh_audio->context)->channels,
len2 / samplesize, samplesize);
}
//len=len2;break;
if(len<0) len=len2; else len+=len2;
buf+=len2;
maxlen -= len2;
sh_audio->pts_bytes += len2;
}
mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2);
if (setup_format(sh_audio, sh_audio->context))
break;
}
return len;
}
static int decode_audio(sh_audio_t *sh, unsigned char *buf,
int minlen, int maxlen)
{
struct spdifContext *spdif_ctx = sh->context;
AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
AVPacket pkt;
double pts;
int ret, in_size, consumed, x;
unsigned char *start = NULL;
consumed = spdif_ctx->out_buffer_len = 0;
spdif_ctx->out_buffer_size = maxlen;
spdif_ctx->out_buffer = buf;
while (spdif_ctx->out_buffer_len + spdif_ctx->iec61937_packet_size < maxlen
&& spdif_ctx->out_buffer_len < minlen) {
if (sh->ds->eof)
break;
x = ds_get_packet_pts(sh->ds, &start, &pts);
if (x <= 0) {
x = 0;
ds_parse(sh->ds, &start, &x, MP_NOPTS_VALUE, 0);
if (x == 0)
continue; // END_NOT_FOUND
in_size = x;
} else {
in_size = x;
consumed = ds_parse(sh->ds, &start, &x, pts, 0);
if (x == 0) {
mp_msg(MSGT_DECAUDIO,MSGL_V,
"start[%p] in_size[%d] consumed[%d] x[%d].\n",
start, in_size, consumed, x);
continue; // END_NOT_FOUND
}
sh->ds->buffer_pos -= in_size - consumed;
}
av_init_packet(&pkt);
pkt.data = start;
pkt.size = x;
mp_msg(MSGT_DECAUDIO,MSGL_V,
"start[%p] pkt.size[%d] in_size[%d] consumed[%d] x[%d].\n",
start, pkt.size, in_size, consumed, x);
if (pts != MP_NOPTS_VALUE) {
sh->pts = pts;
sh->pts_bytes = 0;
}
ret = lavf_ctx->oformat->write_packet(lavf_ctx, &pkt);
if (ret < 0)
break;
}
sh->pts_bytes += spdif_ctx->out_buffer_len;
return spdif_ctx->out_buffer_len;
}
static int init(sh_audio_t *sh, const char *decoder)
{
int x, in_size, srate, bps, *dtshd_rate;
unsigned char *start;
double pts;
AVFormatContext *lavf_ctx = NULL;
AVStream *stream = NULL;
const AVOption *opt = NULL;
struct spdifContext *spdif_ctx = NULL;
spdif_ctx = av_mallocz(sizeof(*spdif_ctx));
if (!spdif_ctx)
goto fail;
spdif_ctx->lavf_ctx = avformat_alloc_context();
if (!spdif_ctx->lavf_ctx)
goto fail;
sh->context = spdif_ctx;
lavf_ctx = spdif_ctx->lavf_ctx;
lavf_ctx->oformat = av_guess_format(FILENAME_SPDIFENC, NULL, NULL);
if (!lavf_ctx->oformat)
goto fail;
lavf_ctx->priv_data = av_mallocz(lavf_ctx->oformat->priv_data_size);
if (!lavf_ctx->priv_data)
goto fail;
lavf_ctx->pb = avio_alloc_context(spdif_ctx->pb_buffer, OUTBUF_SIZE, 1, spdif_ctx,
read_packet, write_packet, seek);
if (!lavf_ctx->pb)
goto fail;
stream = avformat_new_stream(lavf_ctx, 0);
if (!stream)
goto fail;
lavf_ctx->duration = AV_NOPTS_VALUE;
lavf_ctx->start_time = AV_NOPTS_VALUE;
lavf_ctx->streams[0]->codec->codec_id = mp_codec_to_av_codec_id(decoder);
lavf_ctx->raw_packet_buffer_remaining_size = RAW_PACKET_BUFFER_SIZE;
if (AVERROR_PATCHWELCOME == lavf_ctx->oformat->write_header(lavf_ctx)) {
mp_msg(MSGT_DECAUDIO,MSGL_INFO,
"This codec is not supported by spdifenc.\n");
goto fail;
}
// get sample_rate & bitrate from parser
x = ds_get_packet_pts(sh->ds, &start, &pts);
in_size = x;
if (x <= 0) {
pts = MP_NOPTS_VALUE;
x = 0;
}
ds_parse(sh->ds, &start, &x, pts, 0);
srate = 48000; //fake value
bps = 768000/8; //fake value
if (x && sh->avctx) { // we have parser and large enough buffer
if (sh->avctx->sample_rate < 44100) {
mp_msg(MSGT_DECAUDIO,MSGL_INFO,
"This stream sample_rate[%d Hz] may be broken. "
"Force reset 48000Hz.\n",
sh->avctx->sample_rate);
srate = 48000; //fake value
} else
srate = sh->avctx->sample_rate;
bps = sh->avctx->bit_rate/8;
}
sh->ds->buffer_pos -= in_size;
switch (lavf_ctx->streams[0]->codec->codec_id) {
case AV_CODEC_ID_AAC:
spdif_ctx->iec61937_packet_size = 16384;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
break;
case AV_CODEC_ID_AC3:
spdif_ctx->iec61937_packet_size = 6144;
sh->sample_format = AF_FORMAT_AC3_LE;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
break;
case AV_CODEC_ID_DTS:
if(sh->opts->dtshd) {
opt = av_opt_find(&lavf_ctx->oformat->priv_class,
"dtshd_rate", NULL, 0, 0);
if (!opt)
goto fail;
dtshd_rate = (int*)(((uint8_t*)lavf_ctx->priv_data) +
opt->offset);
*dtshd_rate = 192000*4;
spdif_ctx->iec61937_packet_size = 32768;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000; // DTS core require 48000
sh->channels = 2*4;
sh->i_bps = bps;
} else {
spdif_ctx->iec61937_packet_size = 32768;
sh->sample_format = AF_FORMAT_AC3_LE;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
}
break;
case AV_CODEC_ID_EAC3:
spdif_ctx->iec61937_packet_size = 24576;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000;
sh->channels = 2;
sh->i_bps = bps;
break;
case AV_CODEC_ID_MP3:
spdif_ctx->iec61937_packet_size = 4608;
sh->sample_format = AF_FORMAT_MPEG2;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
break;
case AV_CODEC_ID_TRUEHD:
spdif_ctx->iec61937_packet_size = 61440;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000;
sh->channels = 8;
sh->i_bps = bps;
break;
default:
break;
}
return 1;
fail:
uninit(sh);
return 0;
}
static int decode_new_packet(struct sh_audio *sh)
{
struct priv *priv = sh->context;
AVCodecContext *avctx = priv->avctx;
double pts = MP_NOPTS_VALUE;
int insize;
bool packet_already_used = priv->previous_data_left;
struct demux_packet *mpkt = ds_get_packet2(sh->ds,
priv->previous_data_left);
unsigned char *start;
if (!mpkt) {
assert(!priv->previous_data_left);
start = NULL;
insize = 0;
ds_parse(sh->ds, &start, &insize, pts, 0);
if (insize <= 0)
return -1; // error or EOF
} else {
assert(mpkt->len >= priv->previous_data_left);
if (!priv->previous_data_left) {
priv->previous_data_left = mpkt->len;
pts = mpkt->pts;
}
insize = priv->previous_data_left;
start = mpkt->buffer + mpkt->len - priv->previous_data_left;
int consumed = ds_parse(sh->ds, &start, &insize, pts, 0);
priv->previous_data_left -= consumed;
priv->previous_data_left = FFMAX(priv->previous_data_left, 0);
}
AVPacket pkt;
av_init_packet(&pkt);
pkt.data = start;
pkt.size = insize;
if (mpkt && mpkt->avpacket) {
pkt.side_data = mpkt->avpacket->side_data;
pkt.side_data_elems = mpkt->avpacket->side_data_elems;
}
if (pts != MP_NOPTS_VALUE && !packet_already_used) {
sh->pts = pts;
sh->pts_bytes = 0;
}
int got_frame = 0;
int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
// LATM may need many packets to find mux info
if (ret == AVERROR(EAGAIN))
return 0;
if (ret < 0) {
mp_msg(MSGT_DECAUDIO, MSGL_V, "lavc_audio: error\n");
return -1;
}
// The "insize >= ret" test is sanity check against decoder overreads
if (!sh->parser && insize >= ret)
priv->previous_data_left = insize - ret;
if (!got_frame)
return 0;
int format_result = setup_format(sh);
if (format_result < 0)
return format_result;
#ifdef CONFIG_LIBAVRESAMPLE
if (priv->avr) {
int ret;
uint64_t needed_size = av_samples_get_buffer_size(
NULL, priv->resample_channels, priv->avframe->nb_samples,
priv->resample_fmt, 0);
if (needed_size > priv->resample_buf_size) {
priv->resample_buf = talloc_realloc(priv, priv->resample_buf,
uint8_t, needed_size);
priv->resample_buf_size = needed_size;
}
ret = avresample_convert(priv->avr, &priv->resample_buf,
priv->resample_buf_size, priv->avframe->nb_samples,
priv->avframe->extended_data, priv->avframe->linesize[0],
priv->avframe->nb_samples);
if (ret < 0) {
uint8_t error[128];
av_strerror(ret, error, sizeof(error));
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"Error during sample format conversion: %s.\n", error);
return -1;
}
assert(ret == priv->avframe->nb_samples);
priv->output = priv->resample_buf;
priv->output_left = priv->unitsize * ret;
} else
#endif
{
uint64_t unitsize = av_get_bytes_per_sample(avctx->sample_fmt) *
(uint64_t)avctx->channels;
if (unitsize > 100000)
abort();
priv->unitsize = unitsize;
uint64_t output_left = unitsize * priv->avframe->nb_samples;
if (output_left > 500000000)
abort();
priv->output_left = output_left;
priv->output = priv->avframe->data[0];
}
mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d \n", insize,
priv->output_left);
return format_result;
}
static int decode_new_packet(struct sh_audio *sh)
{
struct priv *priv = sh->context;
AVCodecContext *avctx = priv->avctx;
double pts = MP_NOPTS_VALUE;
int insize;
bool packet_already_used = priv->previous_data_left;
struct demux_packet *mpkt = ds_get_packet2(sh->ds,
priv->previous_data_left);
unsigned char *start;
if (!mpkt) {
assert(!priv->previous_data_left);
start = NULL;
insize = 0;
ds_parse(sh->ds, &start, &insize, pts, 0);
if (insize <= 0)
return -1; // error or EOF
} else {
assert(mpkt->len >= priv->previous_data_left);
if (!priv->previous_data_left) {
priv->previous_data_left = mpkt->len;
pts = mpkt->pts;
}
insize = priv->previous_data_left;
start = mpkt->buffer + mpkt->len - priv->previous_data_left;
int consumed = ds_parse(sh->ds, &start, &insize, pts, 0);
priv->previous_data_left -= consumed;
}
AVPacket pkt;
av_init_packet(&pkt);
pkt.data = start;
pkt.size = insize;
if (mpkt && mpkt->avpacket) {
pkt.side_data = mpkt->avpacket->side_data;
pkt.side_data_elems = mpkt->avpacket->side_data_elems;
}
if (pts != MP_NOPTS_VALUE && !packet_already_used) {
sh->pts = pts;
sh->pts_bytes = 0;
}
int got_frame = 0;
int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
// LATM may need many packets to find mux info
if (ret == AVERROR(EAGAIN))
return 0;
if (ret < 0) {
mp_msg(MSGT_DECAUDIO, MSGL_V, "lavc_audio: error\n");
return -1;
}
if (!sh->parser)
priv->previous_data_left += insize - ret;
if (!got_frame)
return 0;
/* An error is reported later from output format checking, but make
* sure we don't crash by overreading first plane. */
if (av_sample_fmt_is_planar(avctx->sample_fmt) && avctx->channels > 1)
return 0;
uint64_t unitsize = (uint64_t)av_get_bytes_per_sample(avctx->sample_fmt) *
avctx->channels;
if (unitsize > 100000)
abort();
priv->unitsize = unitsize;
uint64_t output_left = unitsize * priv->avframe->nb_samples;
if (output_left > 500000000)
abort();
priv->output_left = output_left;
priv->output = priv->avframe->data[0];
mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d \n", insize,
priv->output_left);
return 0;
}
