static size_t resample(StreamingSRC &src, const sample_t *in, size_t in_size, sample_t *out, size_t out_size)
{
size_t result_size = 0;
size_t gone = 0;
while (gone < in_size)
{
gone += src.fill(in + gone, in_size - gone);
if (src.can_process())
{
src.process();
if (result_size + src.size() > out_size)
BOOST_FAIL("Not enough buffer size");
copy_samples(out, result_size, src.result(), 0, src.size());
result_size += src.size();
}
}
while (src.need_flushing())
{
src.flush();
if (result_size + src.size() > out_size)
BOOST_FAIL("Not enough buffer size");
copy_samples(out, result_size, src.result(), 0, src.size());
result_size += src.size();
}
return result_size;
}
void
LineGen::gen_samples(samples_t samples, size_t n)
{
for (size_t i = 0; i < n; i++)
samples[0][i] = t + i*k;
t += n*k;
for (int ch = 1; ch < spk.nch(); ch++)
copy_samples(samples[ch], samples[0], n);
}
void
ToneGen::gen_samples(samples_t samples, size_t n)
{
double w = 2 * M_PI * double(freq) / double(spk.sample_rate);
for (size_t i = 0; i < n; i++)
samples[0][i] = sin(t + i*w);
t += n*w;
for (int ch = 1; ch < spk.nch(); ch++)
copy_samples(samples[ch], samples[0], n);
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
unsigned char *start=NULL;
int y,len=-1, got_frame;
AVFrame *frame = avcodec_alloc_frame();
if (!frame)
return AVERROR(ENOMEM);
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_audio4(sh_audio->context, frame, &got_frame, &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 (!got_frame)
continue;
len2 = copy_samples(sh_audio->context, frame, buf, maxlen);
if (len2 < 0)
return len2;
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;
}
av_free(frame);
return len;
}
static GstFlowReturn
vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet,
GstClockTime timestamp, GstClockTime duration)
{
vorbis_sample_t **pcm;
guint sample_count;
GstBuffer *out;
GstFlowReturn result;
gint size;
if (G_UNLIKELY (!vd->initialized))
goto not_initialized;
/* normal data packet */
/* FIXME, we can skip decoding if the packet is outside of the
* segment, this is however not very trivial as we need a previous
* packet to decode the current one so we must be carefull not to
* throw away too much. For now we decode everything and clip right
* before pushing data. */
if (G_UNLIKELY (vorbis_synthesis (&vd->vb, packet)))
goto could_not_read;
if (G_UNLIKELY (vorbis_synthesis_blockin (&vd->vd, &vd->vb) < 0))
goto not_accepted;
/* assume all goes well here */
result = GST_FLOW_OK;
/* count samples ready for reading */
if ((sample_count = vorbis_synthesis_pcmout (&vd->vd, NULL)) == 0)
goto done;
size = sample_count * vd->vi.channels * vd->width;
GST_LOG_OBJECT (vd, "%d samples ready for reading, size %d", sample_count,
size);
/* alloc buffer for it */
result =
gst_pad_alloc_buffer_and_set_caps (vd->srcpad, GST_BUFFER_OFFSET_NONE,
size, GST_PAD_CAPS (vd->srcpad), &out);
if (G_UNLIKELY (result != GST_FLOW_OK))
goto done;
/* get samples ready for reading now, should be sample_count */
if (G_UNLIKELY ((vorbis_synthesis_pcmout (&vd->vd, &pcm)) != sample_count))
goto wrong_samples;
/* copy samples in buffer */
copy_samples ((vorbis_sample_t *) GST_BUFFER_DATA (out), pcm, sample_count,
vd->vi.channels, vd->width);
GST_LOG_OBJECT (vd, "setting output size to %d", size);
GST_BUFFER_SIZE (out) = size;
/* this should not overflow */
if (duration == -1)
duration = sample_count * GST_SECOND / vd->vi.rate;
vorbis_do_timestamps (vd, out, FALSE, timestamp, duration);
if (vd->segment.rate >= 0.0)
result = vorbis_dec_push_forward (vd, out);
else
result = vorbis_dec_push_reverse (vd, out);
done:
vorbis_synthesis_read (&vd->vd, sample_count);
return result;
/* ERRORS */
not_initialized:
{
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("no header sent yet"));
return GST_FLOW_ERROR;
}
could_not_read:
{
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("couldn't read data packet"));
return GST_FLOW_ERROR;
}
not_accepted:
{
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("vorbis decoder did not accept data packet"));
return GST_FLOW_ERROR;
}
wrong_samples:
{
gst_buffer_unref (out);
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("vorbis decoder reported wrong number of samples"));
return GST_FLOW_ERROR;
}
}
