static audio_channel *libavcodec_compress(void *state, audio_channel * channel)
{
struct libavcodec_codec_state *s = (struct libavcodec_codec_state *) state;
assert(s->magic == MAGIC);
if(channel) {
if(!audio_desc_eq(s->saved_desc, audio_desc_from_audio_channel(channel))) {
if(!reinitialize_coder(s, audio_desc_from_audio_channel(channel))) {
fprintf(stderr, "Unable to reinitialize audio compress!\n");
return NULL;
}
}
if(s->change_bps_to) {
change_bps(s->tmp.data, s->saved_desc.bps, channel->data,
s->change_bps_to, channel->data_len);
s->tmp.data_len += channel->data_len / s->saved_desc.bps * s->change_bps_to;
} else {
memcpy(s->tmp.data + s->tmp.data_len, channel->data, channel->data_len);
s->tmp.data_len += channel->data_len;
}
}
int bps = s->output_channel.bps;
int offset = 0;
s->output_channel.data_len = 0;
int chunk_size = s->codec_ctx->frame_size * bps;
//while(offset + chunk_size <= s->tmp.data_len) {
while(offset + chunk_size <= s->tmp.data_len) {
s->pkt.data = (unsigned char *) s->output_channel.data + s->output_channel.data_len;
s->pkt.size = 1024*1024 - s->output_channel.data_len;
int got_packet;
memcpy(s->samples, s->tmp.data + offset, chunk_size);
int ret = avcodec_encode_audio2(s->codec_ctx, &s->pkt, s->av_frame,
&got_packet);
if(ret) {
char errbuf[1024];
av_strerror(ret, errbuf, sizeof(errbuf));
fprintf(stderr, "Warning: unable to compress audio: %s\n",
errbuf);
}
if(got_packet) {
s->output_channel.data_len += s->pkt.size;
}
offset += chunk_size;
if(!(s->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE))
break;
}
s->tmp.data_len -= offset;
memmove(s->tmp.data, s->tmp.data + offset, s->tmp.data_len);
///fprintf(stderr, "%d %d\n", i++% 2, s->output_channel.data_len);
if(s->output_channel.data_len) {
return &s->output_channel;
} else {
return NULL;
}
}
// Filter data through filter
static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
{
struct mp_audio* l = af->data; // Local data
struct mp_audio* c = data; // Current working data
int len = c->len/c->bps; // Length in samples of current audio block
if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
return NULL;
// Change to cpu native endian format
if((c->format&AF_FORMAT_END_MASK)!=AF_FORMAT_NE)
endian(c->audio,c->audio,len,c->bps);
// Conversion table
if((c->format & AF_FORMAT_POINT_MASK) == AF_FORMAT_F) {
float2int(c->audio, l->audio, len, l->bps);
if((l->format&AF_FORMAT_SIGN_MASK) == AF_FORMAT_US)
si2us(l->audio,len,l->bps);
} else {
// Input must be int
// Change signed/unsigned
if((c->format&AF_FORMAT_SIGN_MASK) != (l->format&AF_FORMAT_SIGN_MASK)){
si2us(c->audio,len,c->bps);
}
// Convert to special formats
switch(l->format&AF_FORMAT_POINT_MASK){
case(AF_FORMAT_F):
int2float(c->audio, l->audio, len, c->bps);
break;
default:
// Change the number of bits
if(c->bps != l->bps)
change_bps(c->audio,l->audio,len,c->bps,l->bps);
else
memcpy(l->audio,c->audio,len*c->bps);
break;
}
}
// Switch from cpu native endian to the correct endianness
if((l->format&AF_FORMAT_END_MASK)!=AF_FORMAT_NE)
endian(l->audio,l->audio,len,l->bps);
// Set output data
c->audio = l->audio;
c->len = len*l->bps;
c->bps = l->bps;
c->format = l->format;
return c;
}
