ssize_t
ad_read_sndfile(WfDecoder* d, float* out, size_t len)
{
SndfileDecoder* priv = (SndfileDecoder*)d->d;
if (!priv) return -1;
switch(d->info.bit_depth){
case 32:
return sf_read_float (priv->sffile, out, len);
case 16: {
short* d16 = g_malloc0(len * sizeof(short));
ssize_t r = sf_read_short (priv->sffile, d16, len);
int16_to_float(out, d16, d->info.channels, len / d->info.channels, 0);
g_free(d16);
return r;
}
#ifdef DEBUG
default:
gwarn("unhandled bit depth: %i", d->info.bit_depth);
#endif
}
return -1;
}
ssize_t ad_read_ffmpeg(void *sf, float* d, size_t len) {
ffmpeg_audio_decoder *priv = (ffmpeg_audio_decoder*) sf;
if (!priv) return -1;
size_t frames = len / priv->channels;
int written = 0;
int ret = 0;
while (ret >= 0 && written < frames) {
dbg(3, "loop: %i/%i (bl:%lu)",written, frames, priv->m_tmpBufferLen );
if (priv->seek_frame == 0 && priv->m_tmpBufferLen > 0 ) {
int s = MIN(priv->m_tmpBufferLen / priv->channels, frames - written );
int16_to_float(priv->m_tmpBufferStart, d, priv->channels, s , written);
written += s;
priv->output_clock+=s;
s = s * priv->channels;
priv->m_tmpBufferStart += s;
priv->m_tmpBufferLen -= s;
ret = 0;
} else {
priv->m_tmpBufferStart = priv->m_tmpBuffer;
priv->m_tmpBufferLen = 0;
if (!priv->pkt_ptr || priv->pkt_len < 1) {
if (priv->packet.data) av_free_packet(&priv->packet);
ret = av_read_frame(priv->formatContext, &priv->packet);
if (ret < 0) { dbg(2, "reached end of file."); break; }
priv->pkt_len = priv->packet.size;
priv->pkt_ptr = priv->packet.data;
}
if (priv->packet.stream_index != priv->audioStream) {
priv->pkt_ptr = NULL;
continue;
}
/* decode all chunks in packet */
int data_size= AVCODEC_MAX_AUDIO_FRAME_SIZE;
#if 0 // TODO ffcompat.h -- this works but is not optimal (channels may not be planar/interleaved)
AVFrame avf; // TODO statically allocate
memset(&avf, 0, sizeof(AVFrame)); // not sure if that is needed
int got_frame = 0;
ret = avcodec_decode_audio4(priv->codecContext, &avf, &got_frame, &priv->packet);
data_size = avf.linesize[0];
memcpy(priv->m_tmpBuffer, avf.data[0], avf.linesize[0] * sizeof(uint8_t));
#else // this was deprecated in LIBAVCODEC_VERSION_MAJOR 53
ret = avcodec_decode_audio3(priv->codecContext, priv->m_tmpBuffer, &data_size, &priv->packet);
#endif
if (ret < 0 || ret > priv->pkt_len) {
#if 0
dbg(0, "audio decode error");
return -1;
#endif
priv->pkt_len = 0;
ret = 0;
continue;
}
priv->pkt_len -= ret; priv->pkt_ptr += ret;
/* sample exact alignment */
if (priv->packet.pts != AV_NOPTS_VALUE) {
priv->decoder_clock = priv->samplerate * av_q2d(priv->formatContext->streams[priv->audioStream]->time_base) * priv->packet.pts;
} else {
dbg(0, "!!! NO PTS timestamp in file");
priv->decoder_clock += (data_size>>1) / priv->channels;
}
if (data_size > 0) {
priv->m_tmpBufferLen+= (data_size>>1); // 2 bytes per sample
}
/* align buffer after seek. */
if (priv->seek_frame > 0) {
const int diff = priv->output_clock-priv->decoder_clock;
if (diff < 0) {
/* seek ended up past the wanted sample */
gwarn("audio seek failed.");
return -1;
} else if (priv->m_tmpBufferLen < (diff*priv->channels)) {
/* wanted sample not in current buffer - keep going */
dbg(2, " !!! seeked sample was not in decoded buffer. frames-to-go: %li", diff);
priv->m_tmpBufferLen = 0;
} else if (diff!=0 && data_size > 0) {
/* wanted sample is in current buffer but not at the beginnning */
dbg(2, " !!! sync buffer to seek. (diff:%i)", diff);
priv->m_tmpBufferStart+= diff*priv->codecContext->channels;
priv->m_tmpBufferLen -= diff*priv->codecContext->channels;
#if 1
memmove(priv->m_tmpBuffer, priv->m_tmpBufferStart, priv->m_tmpBufferLen);
priv->m_tmpBufferStart = priv->m_tmpBuffer;
#endif
priv->seek_frame = 0;
priv->decoder_clock += diff;
} else if (data_size > 0) {
dbg(2, "Audio exact sync-seek (%"PRIi64" == %"PRIi64")", priv->decoder_clock, priv->seek_frame);
priv->seek_frame = 0;
} else {
dbg(0, "Error: no audio data in packet");
}
}
//dbg(0, "PTS: decoder:%"PRIi64". - want: %"PRIi64, priv->decoder_clock, priv->output_clock);
//dbg(0, "CLK: frame: %"PRIi64" T:%.3fs",priv->decoder_clock, (float) priv->decoder_clock/priv->samplerate);
}
}
