S32 encode_vorbis_file(const std::string& in_fname, const std::string& out_fname)
{
#define READ_BUFFER 1024
unsigned char readbuffer[READ_BUFFER*4+44]; /* out of the data segment, not the stack */ /*Flawfinder: ignore*/
ogg_stream_state os; /* take physical pages, weld into a logical stream of packets */
ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
ogg_packet op; /* one raw packet of data for decode */
vorbis_info vi; /* struct that stores all the static vorbis bitstream settings */
vorbis_comment vc; /* struct that stores all the user comments */
vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
vorbis_block vb; /* local working space for packet->PCM decode */
int eos=0;
int result;
U16 num_channels = 0;
U32 sample_rate = 0;
U32 bits_per_sample = 0;
S32 format_error = 0;
std::string error_msg;
if ((format_error = check_for_invalid_wav_formats(in_fname, error_msg)))
{
llwarns << error_msg << ": " << in_fname << llendl;
return(format_error);
}
#if 1
unsigned char wav_header[44]; /*Flawfinder: ignore*/
S32 data_left = 0;
LLAPRFile infile ;
infile.open(in_fname,LL_APR_RB, LLAPRFile::global);
if (!infile.getFileHandle())
{
llwarns << "Couldn't open temporary ogg file for writing: " << in_fname
<< llendl;
return(LLVORBISENC_SOURCE_OPEN_ERR);
}
LLAPRFile outfile ;
outfile.open(out_fname,LL_APR_WPB, LLAPRFile::global);
if (!outfile.getFileHandle())
{
llwarns << "Couldn't open upload sound file for reading: " << in_fname
<< llendl;
return(LLVORBISENC_DEST_OPEN_ERR);
}
// parse the chunks
U32 chunk_length = 0;
U32 file_pos = 12; // start at the first chunk (usually fmt but not always)
while (infile.eof() != APR_EOF)
{
infile.seek(APR_SET,file_pos);
infile.read(wav_header, 44);
chunk_length = ((U32) wav_header[7] << 24)
+ ((U32) wav_header[6] << 16)
+ ((U32) wav_header[5] << 8)
+ wav_header[4];
// llinfos << "chunk found: '" << wav_header[0] << wav_header[1] << wav_header[2] << wav_header[3] << "'" << llendl;
if (!(strncmp((char *)&(wav_header[0]),"fmt ",4)))
{
num_channels = ((U16) wav_header[11] << 8) + wav_header[10];
sample_rate = ((U32) wav_header[15] << 24)
+ ((U32) wav_header[14] << 16)
+ ((U32) wav_header[13] << 8)
+ wav_header[12];
bits_per_sample = ((U16) wav_header[23] << 8) + wav_header[22];
}
else if (!(strncmp((char *)&(wav_header[0]),"data",4)))
{
infile.seek(APR_SET,file_pos+8);
// leave the file pointer at the beginning of the data chunk data
data_left = chunk_length;
break;
}
file_pos += (chunk_length + 8);
chunk_length = 0;
}
/********** Encode setup ************/
/* choose an encoding mode */
/* (mode 0: 44kHz stereo uncoupled, roughly 128kbps VBR) */
vorbis_info_init(&vi);
// always encode to mono
// SL-52913 & SL-53779 determined this quality level to be our 'good
// enough' general-purpose quality level with a nice low bitrate.
// Equivalent to oggenc -q0.5
F32 quality = 0.05f;
// quality = (bitrate==128000 ? 0.4f : 0.1);
// if (vorbis_encode_init(&vi, /* num_channels */ 1 ,sample_rate, -1, bitrate, -1))
if (vorbis_encode_init_vbr(&vi, /* num_channels */ 1 ,sample_rate, quality))
// if (vorbis_encode_setup_managed(&vi,1,sample_rate,-1,bitrate,-1) ||
// vorbis_encode_ctl(&vi,OV_ECTL_RATEMANAGE_AVG,NULL) ||
// vorbis_encode_setup_init(&vi))
{
llwarns << "unable to initialize vorbis codec at quality " << quality << llendl;
// llwarns << "unable to initialize vorbis codec at bitrate " << bitrate << llendl;
return(LLVORBISENC_DEST_OPEN_ERR);
}
/* add a comment */
vorbis_comment_init(&vc);
// vorbis_comment_add(&vc,"Linden");
/* set up the analysis state and auxiliary encoding storage */
vorbis_analysis_init(&vd,&vi);
vorbis_block_init(&vd,&vb);
/* set up our packet->stream encoder */
/* pick a random serial number; that way we can more likely build
chained streams just by concatenation */
ogg_stream_init(&os, ll_rand());
/* Vorbis streams begin with three headers; the initial header (with
most of the codec setup parameters) which is mandated by the Ogg
bitstream spec. The second header holds any comment fields. The
third header holds the bitstream codebook. We merely need to
make the headers, then pass them to libvorbis one at a time;
libvorbis handles the additional Ogg bitstream constraints */
{
ogg_packet header;
ogg_packet header_comm;
ogg_packet header_code;
vorbis_analysis_headerout(&vd,&vc,&header,&header_comm,&header_code);
ogg_stream_packetin(&os,&header); /* automatically placed in its own
page */
ogg_stream_packetin(&os,&header_comm);
ogg_stream_packetin(&os,&header_code);
/* We don't have to write out here, but doing so makes streaming
* much easier, so we do, flushing ALL pages. This ensures the actual
* audio data will start on a new page
*/
while(!eos){
int result=ogg_stream_flush(&os,&og);
if(result==0)break;
outfile.write(og.header, og.header_len);
outfile.write(og.body, og.body_len);
}
}
while(!eos)
{
long bytes_per_sample = bits_per_sample/8;
long bytes=(long)infile.read(readbuffer,llclamp((S32)(READ_BUFFER*num_channels*bytes_per_sample),0,data_left)); /* stereo hardwired here */
if (bytes==0)
{
/* end of file. this can be done implicitly in the mainline,
but it's easier to see here in non-clever fashion.
Tell the library we're at end of stream so that it can handle
the last frame and mark end of stream in the output properly */
vorbis_analysis_wrote(&vd,0);
// eos = 1;
}
else
{
long i;
long samples;
int temp;
data_left -= bytes;
/* data to encode */
/* expose the buffer to submit data */
float **buffer=vorbis_analysis_buffer(&vd,READ_BUFFER);
i = 0;
samples = bytes / (num_channels * bytes_per_sample);
if (num_channels == 2)
{
if (bytes_per_sample == 2)
{
/* uninterleave samples */
for(i=0; i<samples ;i++)
{
temp = ((signed char *)readbuffer)[i*4+1]; /*Flawfinder: ignore*/
temp += ((signed char *)readbuffer)[i*4+3]; /*Flawfinder: ignore*/
temp <<= 8;
temp += readbuffer[i*4];
temp += readbuffer[i*4+2];
buffer[0][i] = ((float)temp) / 65536.f;
}
}
else // presume it's 1 byte per which is unsigned (F#@%ing wav "standard")
{
/* uninterleave samples */
for(i=0; i<samples ;i++)
{
temp = readbuffer[i*2+0];
temp += readbuffer[i*2+1];
temp -= 256;
buffer[0][i] = ((float)temp) / 256.f;
}
}
}
else if (num_channels == 1)
{
if (bytes_per_sample == 2)
{
for(i=0; i < samples ;i++)
{
temp = ((signed char*)readbuffer)[i*2+1];
temp <<= 8;
temp += readbuffer[i*2];
buffer[0][i] = ((float)temp) / 32768.f;
}
}
else // presume it's 1 byte per which is unsigned (F#@%ing wav "standard")
{
for(i=0; i < samples ;i++)
{
temp = readbuffer[i];
temp -= 128;
buffer[0][i] = ((float)temp) / 128.f;
}
}
}
/* tell the library how much we actually submitted */
vorbis_analysis_wrote(&vd,i);
}
/* vorbis does some data preanalysis, then divvies up blocks for
more involved (potentially parallel) processing. Get a single
block for encoding now */
while(vorbis_analysis_blockout(&vd,&vb)==1)
{
/* analysis */
/* Do the main analysis, creating a packet */
vorbis_analysis(&vb, NULL);
vorbis_bitrate_addblock(&vb);
while(vorbis_bitrate_flushpacket(&vd, &op))
{
/* weld the packet into the bitstream */
ogg_stream_packetin(&os,&op);
/* write out pages (if any) */
while(!eos)
{
result = ogg_stream_pageout(&os,&og);
if(result==0)
break;
outfile.write(og.header, og.header_len);
outfile.write(og.body, og.body_len);
/* this could be set above, but for illustrative purposes, I do
it here (to show that vorbis does know where the stream ends) */
if(ogg_page_eos(&og))
eos=1;
}
}
}
}
/* clean up and exit. vorbis_info_clear() must be called last */
ogg_stream_clear(&os);
vorbis_block_clear(&vb);
vorbis_dsp_clear(&vd);
vorbis_comment_clear(&vc);
vorbis_info_clear(&vi);
/* ogg_page and ogg_packet structs always point to storage in
libvorbis. They're never freed or manipulated directly */
// fprintf(stderr,"Vorbis encoding: Done.\n");
llinfos << "Vorbis encoding: Done." << llendl;
#endif
return(LLVORBISENC_NOERR);
}
static void *ucil_theora_encode_thread( ucil_theora_video_file_object_t *vobj )
{
yuv_buffer yuv;
double videopos = 0;
double audiopos = 0;
int gotpage = 0;
unsigned char *ds_y_buffer = NULL;
unsigned char *ds_u_buffer = NULL;
unsigned char *ds_v_buffer = NULL;
yuv.y_width = vobj->ti.width;
yuv.y_height = vobj->ti.height;
yuv.y_stride = vobj->ti.width;
yuv.uv_width = vobj->ti.width / 2;
yuv.uv_height = vobj->ti.height / 2;
yuv.uv_stride = vobj->ti.width / 2;
if( vobj->downsize > 1 || vobj->requires_resizing_frames )
{
ds_y_buffer = malloc( yuv.y_width * yuv.y_height );
ds_u_buffer = malloc( yuv.uv_width * yuv.uv_height );
ds_v_buffer = malloc( yuv.uv_width * yuv.uv_height );
}
vobj->last_frame = NULL;
while( !vobj->quit_thread )
{
unicap_data_buffer_t *data_buffer;
ogg_page og;
sem_wait( &vobj->lock );
data_buffer = ( unicap_data_buffer_t *)g_queue_pop_head( vobj->full_queue );
sem_post( &vobj->lock );
if( !data_buffer )
{
audiopos = fetch_and_process_audio( vobj, audiopos ); usleep( 1000 );
continue;
}
if( vobj->frame_count == 0 )
{
memcpy( &vobj->recording_start_time, &data_buffer->fill_time, sizeof( struct timeval ) );
}
audiopos = fetch_and_process_audio( vobj, audiopos );
/* printf( "v: %f a: %f\n", videopos, audiopos ); */
if( vobj->audio && ( videopos > audiopos ) )
{
data_buffer->flags &= ~UNICAP_FLAGS_BUFFER_LOCKED;
sem_wait( &vobj->lock );
g_queue_push_head( vobj->empty_queue, data_buffer );
sem_post( &vobj->lock );
continue;
}
if( vobj->fill_frames )
{
if( vobj->audio )
{
if( vobj->last_frame )
{
unicap_data_buffer_t *last_data_buffer;
double streampos;
struct timeval streamtime;
last_data_buffer = vobj->last_frame;
if( vobj->downsize > 1 || vobj->requires_resizing_frames )
{
yuv.y = ds_y_buffer;
yuv.u = ds_u_buffer;
yuv.v = ds_v_buffer;
}
else
{
yuv.y = last_data_buffer->data;
yuv.u = last_data_buffer->data + ( yuv.y_stride * yuv.y_height );
yuv.v = yuv.u + ( yuv.uv_stride * yuv.uv_height );
}
streamtime.tv_sec = data_buffer->fill_time.tv_sec - vobj->recording_start_time.tv_sec;
streamtime.tv_usec = data_buffer->fill_time.tv_usec;
if( data_buffer->fill_time.tv_usec < vobj->recording_start_time.tv_usec )
{
streamtime.tv_sec--;
streamtime.tv_usec += 1000000;
}
streamtime.tv_usec -= vobj->recording_start_time.tv_usec;
streampos = streamtime.tv_sec + ( (double)streamtime.tv_usec / 1000000.0f );
// If the streampos is ahead, this means that we get
// less than 30 frames per seconds.
// --> Fill up the stream
while( streampos > videopos )
{
/* printf( "s v: %f a: %f\n", videopos, audiopos ); */
gotpage = 0;
if( theora_encode_YUVin( &vobj->th, &yuv ) )
{
TRACE( "theora_encode_YUVin FAILED!\n" );
}
theora_encode_packetout( &vobj->th, 0, &vobj->op );
ogg_stream_packetin( &vobj->os, &vobj->op );
while( ogg_stream_pageout( &vobj->os, &og ) )
{
double gt;
fwrite( og.header, og.header_len, 1, vobj->f );
fwrite( og.body, og.body_len, 1, vobj->f );
gt = theora_granule_time( &vobj->th, ogg_page_granulepos( &og ) );
if( gt < 0 )
{
continue;
}
gotpage = 1;
videopos = gt;
/* printf( "THEORA: %f\n", videopos ); */
}
if( !gotpage )
{
videopos += vobj->frame_interval / 1000000.0f;
}
vobj->frame_count++;
audiopos = fetch_and_process_audio( vobj, audiopos );
}
//
while( ( videopos + ( vobj->frame_interval / 1000000.0f ) ) < audiopos )
{
/* printf( "a v: %f a: %f\n", videopos, audiopos ); */
gotpage = 0;
if( theora_encode_YUVin( &vobj->th, &yuv ) )
{
TRACE( "theora_encode_YUVin FAILED!\n" );
}
theora_encode_packetout( &vobj->th, 0, &vobj->op );
ogg_stream_packetin( &vobj->os, &vobj->op );
while( ogg_stream_pageout( &vobj->os, &og ) )
{
double gt;
fwrite( og.header, og.header_len, 1, vobj->f );
fwrite( og.body, og.body_len, 1, vobj->f );
gt = theora_granule_time( &vobj->th, ogg_page_granulepos( &og ) );
if( gt < 0 )
{
continue;
}
gotpage = 1;
videopos = gt;
/* printf( "THEORA: %f\n", videopos ); */
}
if( !gotpage )
{
videopos += vobj->frame_interval / 1000000.0f;
}
vobj->frame_count++;
audiopos = fetch_and_process_audio( vobj, audiopos );
}
last_data_buffer->flags &= ~UNICAP_FLAGS_BUFFER_LOCKED;
sem_wait( &vobj->lock );
g_queue_push_head( vobj->empty_queue, vobj->last_frame );
sem_post( &vobj->lock );
vobj->last_frame = NULL;
}
}
else
{
fill_frames( vobj, data_buffer, &yuv, ds_y_buffer, ds_u_buffer, ds_v_buffer );
}
}
else// ( !vobj->fill_frames )
{
if( vobj->last_frame )
{
unicap_data_buffer_t *last_data_buffer;
last_data_buffer = vobj->last_frame;
last_data_buffer->flags &= ~UNICAP_FLAGS_BUFFER_LOCKED;
sem_wait( &vobj->lock );
g_queue_push_head( vobj->empty_queue, vobj->last_frame );
sem_post( &vobj->lock );
}
vobj->last_frame = NULL;
}
//
// Encode the new buffer
//
if( vobj->encode_frame_cb )
{
vobj->encode_frame_cb( UNICAP_EVENT_NEW_FRAME, NULL, data_buffer, vobj->encode_frame_cb_data );
}
vobj->frame_count++;
if( vobj->downsize > 1 || vobj->requires_resizing_frames )
{
downsize_yuv420p( vobj->format.size.width, vobj->format.size.height, vobj->downsize,
vobj->ti.width, vobj->ti.height,
ds_y_buffer, ds_u_buffer, ds_v_buffer,
data_buffer->data, data_buffer->data + ( vobj->format.size.width * vobj->format.size.height ),
data_buffer->data + ( vobj->format.size.width * vobj->format.size.height ) +
( ( vobj->format.size.width * vobj->format.size.height ) / 4 ) );
yuv.y = ds_y_buffer;
yuv.u = ds_u_buffer;
yuv.v = ds_v_buffer;
}
else
{
yuv.y = data_buffer->data;
yuv.u = data_buffer->data + ( yuv.y_stride * yuv.y_height );
yuv.v = yuv.u + ( yuv.uv_stride * yuv.uv_height );
}
if( theora_encode_YUVin( &vobj->th, &yuv ) )
{
TRACE( "theora_encode_YUVin FAILED!\n" );
}
memcpy( &vobj->last_frame_time, &data_buffer->fill_time, sizeof( struct timeval ) );
vobj->last_frame = data_buffer;
theora_encode_packetout( &vobj->th, 0, &vobj->op );
ogg_stream_packetin( &vobj->os, &vobj->op );
/* printf( "= v: %f a: %f\n", videopos, audiopos ); */
gotpage = 0;
while( ogg_stream_pageout( &vobj->os, &og ) )
{
double gt;
fwrite( og.header, og.header_len, 1, vobj->f );
fwrite( og.body, og.body_len, 1, vobj->f );
gt = theora_granule_time( &vobj->th, ogg_page_granulepos( &og ) );
if( gt < 0 )
{
continue;
}
gotpage = 1;
videopos = gt;
/* printf( "THEORA: %f\n", videopos ); */
}
if( !gotpage )
{
videopos += vobj->frame_interval / 1000000.0f;
}
}
if( vobj->last_frame )
{
// encode again to set eos
unicap_data_buffer_t *last_data_buffer;
ogg_page og;
ogg_packet op;
#if HAVE_ALSA
if( vobj->audio && !vobj->async_audio_encoding )
{
audiopos = fetch_and_process_audio( vobj, audiopos );
vorbis_analysis_wrote( &vobj->vd, 0 );
while( vorbis_analysis_blockout( &vobj->vd, &vobj->vb ) == 1 )
{
vorbis_analysis( &vobj->vb, NULL );
vorbis_bitrate_addblock( &vobj->vb );
while( vorbis_bitrate_flushpacket( &vobj->vd, &op ) )
{
ogg_stream_packetin( &vobj->vo, &op );
}
}
while( ogg_stream_pageout( &vobj->vo, &og ) )
{
fwrite( og.header, og.header_len, 1, vobj->f );
fwrite( og.body, og.body_len, 1, vobj->f );
}
}
else if( vobj->audio )
{
audiopos = fetch_and_process_audio( vobj, audiopos );
}
#endif
last_data_buffer = vobj->last_frame;
if( vobj->downsize > 1 || vobj->requires_resizing_frames )
{
yuv.y = ds_y_buffer;
yuv.u = ds_u_buffer;
yuv.v = ds_v_buffer;
}
else
{
yuv.y = last_data_buffer->data;
yuv.u = last_data_buffer->data + ( yuv.y_stride * yuv.y_height );
yuv.v = yuv.u + ( yuv.uv_stride * yuv.uv_height );
}
if( theora_encode_YUVin( &vobj->th, &yuv ) )
{
TRACE( "theora_encode_YUVin FAILED!\n" );
}
theora_encode_packetout( &vobj->th, 1, &vobj->op );
ogg_stream_packetin( &vobj->os, &vobj->op );
while( ogg_stream_pageout( &vobj->os, &og ) )
{
/* printf( "THEORA: %f\n", theora_granule_time( &vobj->th, ogg_page_granulepos( &og ) ) ); */
fwrite( og.header, og.header_len, 1, vobj->f );
fwrite( og.body, og.body_len, 1, vobj->f );
}
last_data_buffer->flags &= ~UNICAP_FLAGS_BUFFER_LOCKED;
sem_wait( &vobj->lock );
g_queue_push_head( vobj->empty_queue, vobj->last_frame );
sem_post( &vobj->lock );
vobj->last_frame = NULL;
}
return NULL;
}
static int oggvorbis_encode_frame(AVCodecContext *avccontext,
unsigned char *packets,
int buf_size, void *data)
{
OggVorbisContext *context = avccontext->priv_data ;
ogg_packet op ;
signed short *audio = data ;
int l;
if(data) {
int samples = OGGVORBIS_FRAME_SIZE;
float **buffer ;
buffer = vorbis_analysis_buffer(&context->vd, samples) ;
if(context->vi.channels == 1) {
for(l = 0 ; l < samples ; l++)
buffer[0][l]=audio[l]/32768.f;
} else {
for(l = 0 ; l < samples ; l++){
buffer[0][l]=audio[l*2]/32768.f;
buffer[1][l]=audio[l*2+1]/32768.f;
}
}
vorbis_analysis_wrote(&context->vd, samples) ;
} else {
if(!context->eof)
vorbis_analysis_wrote(&context->vd, 0) ;
context->eof = 1;
}
while(vorbis_analysis_blockout(&context->vd, &context->vb) == 1) {
vorbis_analysis(&context->vb, NULL);
vorbis_bitrate_addblock(&context->vb) ;
while(vorbis_bitrate_flushpacket(&context->vd, &op)) {
/* i'd love to say the following line is a hack, but sadly it's
* not, apparently the end of stream decision is in libogg. */
if(op.bytes==1)
continue;
memcpy(context->buffer + context->buffer_index, &op, sizeof(ogg_packet));
context->buffer_index += sizeof(ogg_packet);
memcpy(context->buffer + context->buffer_index, op.packet, op.bytes);
context->buffer_index += op.bytes;
// av_log(avccontext, AV_LOG_DEBUG, "e%d / %d\n", context->buffer_index, op.bytes);
}
}
l=0;
if(context->buffer_index){
ogg_packet *op2= (ogg_packet*)context->buffer;
op2->packet = context->buffer + sizeof(ogg_packet);
l= op2->bytes;
avccontext->coded_frame->pts= av_rescale_q(op2->granulepos, (AVRational){1, avccontext->sample_rate}, avccontext->time_base);
//FIXME we should reorder the user supplied pts and not assume that they are spaced by 1/sample_rate
memcpy(packets, op2->packet, l);
context->buffer_index -= l + sizeof(ogg_packet);
memmove(context->buffer, context->buffer + l + sizeof(ogg_packet), context->buffer_index);
// av_log(avccontext, AV_LOG_DEBUG, "E%d\n", l);
}
return l;
}
HOTSPOT PUBLIC SW32 vorbis_encode( const char *filename, void *data, W32 size, W32 in_channels, W32 in_samplesize,
W32 rate, W32 quality, W32 max_bitrate, W32 min_bitrate )
{
FILE *fp;
ogg_stream_state os;
ogg_page og;
ogg_packet op;
vorbis_dsp_state vd;
vorbis_block vb;
vorbis_info vi;
ogg_packet header_main;
ogg_packet header_comments;
ogg_packet header_codebooks;
SW32 result;
W32 serialno = 0;
vorbis_comment comments;
SW32 ret = 0;
SW32 eos;
W32 samplesdone = 0;
W32 packetsdone = 0;
W32 bytes_written = 0;
fp = fopen( filename, "wb" );
if( fp == NULL )
{
return 0;
}
memset( &comments, 0, sizeof( comments ) );
channels = in_channels;
samplesize = in_samplesize;
ptrCurrent = (PW8)data;
ptrEnd = (PW8)data + size;
vorbis_info_init( &vi );
if( vorbis_encode_setup_vbr( &vi, channels, rate, quality ) )
{
fprintf( stderr, "Mode initialisation failed: invalid parameters for quality\n" );
vorbis_info_clear( &vi );
// Añadido como consejo de cppcheck.
fclose(fp);
return 1;
}
/* do we have optional hard quality restrictions? */
if( max_bitrate > 0 || min_bitrate > 0 )
{
struct ovectl_ratemanage_arg ai;
vorbis_encode_ctl( &vi, OV_ECTL_RATEMANAGE_GET, &ai );
ai.bitrate_hard_min = min_bitrate;
ai.bitrate_hard_max = max_bitrate;
ai.management_active = 1;
vorbis_encode_ctl( &vi, OV_ECTL_RATEMANAGE_SET, &ai );
}
/* Turn off management entirely (if it was turned on). */
vorbis_encode_ctl( &vi, OV_ECTL_RATEMANAGE_SET, NULL );
vorbis_encode_setup_init( &vi );
vorbis_analysis_init( &vd, &vi );
vorbis_block_init( &vd, &vb );
ogg_stream_init( &os, serialno );
/* Now, build the three header packets and send through to the stream
output stage (but defer actual file output until the main encode loop) */
/* Build the packets */
ret = vorbis_analysis_headerout( &vd, &comments,
&header_main, &header_comments, &header_codebooks );
/* And stream them out */
ogg_stream_packetin( &os, &header_main );
ogg_stream_packetin( &os, &header_comments );
ogg_stream_packetin( &os, &header_codebooks );
while( (result = ogg_stream_flush( &os, &og )) )
{
ret = fwrite( og.header, 1, og.header_len, fp );
ret += fwrite( og.body, 1, og.body_len, fp );
if(ret != og.header_len + og.body_len)
{
fprintf( stderr, "[vorbis_encode]: Failed writing header to output stream\n") ;
ret = 1;
goto cleanup; /* Bail and try to clean up stuff */
}
}
eos = 0;
/* Main encode loop - continue until end of file */
while( ! eos )
{
float **buffer = vorbis_analysis_buffer( &vd, READSIZE );
SW32 samples_read = read_samples( buffer, READSIZE );
if( samples_read == 0 )
{
/* Tell the library that we wrote 0 bytes - signalling the end */
vorbis_analysis_wrote( &vd, 0 );
}
else
{
samplesdone += samples_read;
/* Call progress update every 40 pages */
if( packetsdone >= 40 )
{
packetsdone = 0;
// progress bar here
}
/* Tell the library how many samples (per channel) we wrote
into the supplied buffer */
vorbis_analysis_wrote( &vd, samples_read );
}
/* While we can get enough data from the library to analyse, one
block at a time... */
while( vorbis_analysis_blockout( &vd, &vb ) == 1 )
{
/* Do the main analysis, creating a packet */
vorbis_analysis( &vb, NULL );
vorbis_bitrate_addblock( &vb );
while( vorbis_bitrate_flushpacket( &vd, &op ) )
{
/* Add packet to bitstream */
ogg_stream_packetin( &os, &op );
packetsdone++;
/* If we've gone over a page boundary, we can do actual output,
so do so (for however many pages are available) */
while( ! eos )
{
SW32 result = ogg_stream_pageout( &os, &og );
if( ! result )
{
break;
}
ret = fwrite( og.header, 1, og.header_len, fp );
ret += fwrite( og.body, 1, og.body_len, fp );
if(ret != og.header_len + og.body_len)
{
fprintf( stderr, "[vorbis_encode]: Failed writing data to output stream\n" );
ret = 1;
goto cleanup; /* Bail */
}
else
{
bytes_written += ret;
}
if( ogg_page_eos( &og ) )
{
eos = 1;
}
}
}
}
}
cleanup:
fclose( fp );
ogg_stream_clear( &os );
vorbis_block_clear( &vb );
vorbis_dsp_clear( &vd );
vorbis_info_clear( &vi );
return 0;
}
const char *
_edje_multisense_encode_to_ogg_vorbis(char *snd_path, double quality, SF_INFO sfinfo)
{
ogg_stream_state os; /* take physical pages, weld into a logical stream of packets */
ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
ogg_packet op; /* one raw packet of data for decode */
vorbis_info vi; /* struct that stores all the static vorbis bitstream settings */
vorbis_comment vc; /* struct that stores all the user comments */
vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
vorbis_block vb; /* local working space for packet->PCM decode */
int eos = 0, ret;
char *tmp;
SNDFILE *sfile;
FILE *fout;
sfile = sf_open(snd_path, SFM_READ, &sfinfo);
if (!sfile) return NULL;
if (!sf_format_check(&sfinfo))
{
sf_close(sfile);
return NULL;
}
tmp = malloc(strlen(snd_path) + 1 + 4);
if (!tmp)
{
sf_close(sfile);
return NULL;
}
strcpy(tmp, snd_path);
snd_path = tmp;
strcat(snd_path, ".ogg");
fout = fopen(snd_path, "wb");
if (!fout)
{
free(snd_path);
sf_close(sfile);
return NULL;
}
/********** Encode setup ************/
vorbis_info_init(&vi);
ret = vorbis_encode_init(&vi, sfinfo.channels, sfinfo.samplerate,
-1, (long)(quality * 1000), -1);
if (ret == OV_EFAULT) printf("OV_EFAULT\n");
if (ret == OV_EINVAL) printf("OV_EINVAL\n");
if (ret == OV_EIMPL) printf("OV_EIMPL\n");
if (ret)
{
fclose(fout);
free(snd_path);
sf_close(sfile);
return NULL;
}
/* add a comment */
vorbis_comment_init(&vc);
vorbis_comment_add_tag(&vc, "", "");
/* set up the analysis state and auxiliary encoding storage */
vorbis_analysis_init(&vd, &vi);
vorbis_block_init(&vd, &vb);
srand(time(NULL));
ogg_stream_init(&os, rand());
ogg_packet header;
ogg_packet header_comm;
ogg_packet header_code;
vorbis_analysis_headerout(&vd, &vc, &header, &header_comm, &header_code);
ogg_stream_packetin(&os, &header); /* automatically placed in its own page */
ogg_stream_packetin(&os, &header_comm);
ogg_stream_packetin(&os, &header_code);
while (!eos)
{
int result = ogg_stream_flush(&os, &og);
if (!result) break;
fwrite(og.header, 1, og.header_len, fout);
fwrite(og.body, 1, og.body_len, fout);
}
while (!eos)
{
int i, ch;
float readbuffer[READBUF * 2];
sf_count_t count;
count = sf_readf_float(sfile, readbuffer, READBUF);
if (!count)
vorbis_analysis_wrote(&vd, 0);
else
{
float **buffer = vorbis_analysis_buffer(&vd, count);
/* uninterleave samples */
for (i = 0; i < count; i++)
{
for (ch = 0; ch < sfinfo.channels; ch++)
buffer[ch][i]= readbuffer[(i * sfinfo.channels) + ch];
}
vorbis_analysis_wrote(&vd, i);
}
while (vorbis_analysis_blockout(&vd, &vb) == 1)
{
vorbis_analysis(&vb, NULL);
vorbis_bitrate_addblock(&vb);
while (vorbis_bitrate_flushpacket(&vd, &op))
{
ogg_stream_packetin(&os, &op);
while (!eos)
{
int result = ogg_stream_pageout(&os, &og);
if (!result) break;
fwrite(og.header, 1, og.header_len, fout);
fwrite(og.body, 1, og.body_len, fout);
if (ogg_page_eos(&og)) eos = 1;
}
}
}
}
ogg_stream_clear(&os);
vorbis_block_clear(&vb);
vorbis_dsp_clear(&vd);
vorbis_comment_clear(&vc);
vorbis_info_clear(&vi);
sf_close(sfile);
fclose (fout);
return snd_path;
}
int StreamEncoder::EncodeBuffer(void* buffer, int length)
{
/*
While there is more audio to encode:
Submit a chunk of audio data using vorbis_analysis_buffer and vorbis_analysis_wrote.
Obtain all available blocks using vorbis_analysis_blockout in a loop. For each block obtained:
Encode the block into a packet (or prepare it for bitrate management) using vorbis_analysis. (It's a good idea to always pass the blocks through the bitrate management mechanism; more information is on the vorbis_analysis page. It does not affect the resulting packets unless you are actually using a bitrate-managed mode.)
If you are using bitrate management, submit the block using vorbis_bitrate_addblock and obtain packets using vorbis_bitrate_flushpacket.
Output any obtained packets.
*/
float** allocatedChannels = vorbis_analysis_buffer(&mVorbisDspState, length);
//I'm not sure of the way that the samples inside of buffer are laid out
float* bufferChannels = static_cast<float*>(buffer);
Write("Casted the buffer channels");
for (int i = 0; i < length; i+=2)
{
//Assume two channels here
allocatedChannels[0][i] = bufferChannels[i];
allocatedChannels[1][i] = bufferChannels[i+1];
}
if (vorbis_analysis_wrote(&mVorbisDspState, length) != 0)
{
//Error
return 0;
}
int processedSamplesCount = 0;
int retValue = 1;
while (retValue == 1)
{
retValue = vorbis_analysis_blockout(&mVorbisDspState, &mVorbisBlock) == 1;
if (retValue >= 0)
{
//No Error
if (vorbis_analysis(&mVorbisBlock, NULL) == 0)
{
vorbis_bitrate_addblock(&mVorbisBlock);
ogg_packet samplePacket;
int fpRetValue = 1;
while (fpRetValue == 1)
{
fpRetValue = vorbis_bitrate_flushpacket(&mVorbisDspState, &samplePacket);
if (fpRetValue >= 0)
{
//Store the bytes inside of buffer
//memcpy(buffer, samplePacket.bytes, len)
//processedSamplesCount += len;
}
}
}
}
if (retValue == 0)
{
break;
}
}
return processedSamplesCount;
}
virtual bool Cook(FName Format, const TArray<uint8>& SrcBuffer, FSoundQualityInfo& QualityInfo, TArray<uint8>& CompressedDataStore) const override
{
check(Format == NAME_OGG);
#if WITH_OGGVORBIS
{
short ReadBuffer[SAMPLES_TO_READ * SAMPLE_SIZE * 2];
ogg_stream_state os; // take physical pages, weld into a logical stream of packets
ogg_page og; // one ogg bitstream page. Vorbis packets are inside
ogg_packet op; // one raw packet of data for decode
vorbis_info vi; // struct that stores all the static vorbis bitstream settings
vorbis_comment vc; // struct that stores all the user comments
vorbis_dsp_state vd; // central working state for the packet->PCM decoder
vorbis_block vb; // local working space for packet->PCM decode
uint32 i;
bool eos;
// Create a buffer to store compressed data
CompressedDataStore.Empty();
FMemoryWriter CompressedData( CompressedDataStore );
uint32 BufferOffset = 0;
float CompressionQuality = ( float )( QualityInfo.Quality * VORBIS_QUALITY_MODIFIER ) / 100.0f;
CompressionQuality = FMath::Clamp( CompressionQuality, -0.1f, 1.0f );
vorbis_info_init( &vi );
if( vorbis_encode_init_vbr( &vi, QualityInfo.NumChannels, QualityInfo.SampleRate, CompressionQuality ) )
{
return false;
}
// add a comment
vorbis_comment_init( &vc );
vorbis_comment_add_tag( &vc, "ENCODER", "UnrealEngine4" );
// set up the analysis state and auxiliary encoding storage
vorbis_analysis_init( &vd, &vi );
vorbis_block_init( &vd, &vb );
// set up our packet->stream encoder
ogg_stream_init( &os, 0 );
ogg_packet header;
ogg_packet header_comm;
ogg_packet header_code;
vorbis_analysis_headerout( &vd, &vc, &header, &header_comm, &header_code);
ogg_stream_packetin( &os, &header );
ogg_stream_packetin( &os, &header_comm );
ogg_stream_packetin( &os, &header_code );
// This ensures the actual audio data will start on a new page, as per spec
while( true )
{
int result = ogg_stream_flush( &os, &og );
if( result == 0 )
{
break;
}
CompressedData.Serialize( og.header, og.header_len );
CompressedData.Serialize( og.body, og.body_len );
}
eos = false;
while( !eos )
{
// Read samples
uint32 BytesToRead = FMath::Min( SAMPLES_TO_READ * QualityInfo.NumChannels * SAMPLE_SIZE, QualityInfo.SampleDataSize - BufferOffset );
FMemory::Memcpy( ReadBuffer, SrcBuffer.GetData() + BufferOffset, BytesToRead );
BufferOffset += BytesToRead;
if( BytesToRead == 0)
{
// end of file
vorbis_analysis_wrote( &vd, 0 );
}
else
{
// expose the buffer to submit data
float **buffer = vorbis_analysis_buffer( &vd, SAMPLES_TO_READ );
if( QualityInfo.NumChannels == 1 )
{
for( i = 0; i < BytesToRead / SAMPLE_SIZE; i++ )
{
buffer[0][i] = ( ReadBuffer[i] ) / 32768.0f;
}
}
else
{
for( i = 0; i < BytesToRead / ( SAMPLE_SIZE * 2 ); i++ )
{
buffer[0][i] = ( ReadBuffer[i * 2] ) / 32768.0f;
buffer[1][i] = ( ReadBuffer[i * 2 + 1] ) / 32768.0f;
}
}
// tell the library how many samples we actually submitted
vorbis_analysis_wrote( &vd, i );
}
// vorbis does some data preanalysis, then divvies up blocks for more involved (potentially parallel) processing.
while( vorbis_analysis_blockout( &vd, &vb ) == 1 )
{
// analysis, assume we want to use bitrate management
vorbis_analysis( &vb, NULL );
vorbis_bitrate_addblock( &vb );
while( vorbis_bitrate_flushpacket( &vd, &op ) )
{
// weld the packet into the bitstream
ogg_stream_packetin( &os, &op );
// write out pages (if any)
while( !eos )
{
int result = ogg_stream_pageout( &os, &og );
if( result == 0 )
{
break;
}
CompressedData.Serialize( og.header, og.header_len );
CompressedData.Serialize( og.body, og.body_len );
// this could be set above, but for illustrative purposes, I do it here (to show that vorbis does know where the stream ends)
if( ogg_page_eos( &og ) )
{
eos = true;
}
}
}
}
}
// clean up and exit. vorbis_info_clear() must be called last
ogg_stream_clear( &os );
vorbis_block_clear( &vb );
vorbis_dsp_clear( &vd );
vorbis_comment_clear( &vc );
vorbis_info_clear( &vi );
// ogg_page and ogg_packet structs always point to storage in libvorbis. They're never freed or manipulated directly
}
return CompressedDataStore.Num() > 0;
#else
return false;
#endif // WITH_OGGVOBVIS
}
/**
\fn encode
*/
bool AUDMEncoder_Vorbis::encode(uint8_t *dest, uint32_t *len, uint32_t *samples)
{
uint32_t nbout;
uint32_t consumed=0;
float **float_samples;
int channels=wavheader.channels;
ogg_packet op ;
*len = 0;
_chunk=1024*channels;
int count=ROUNDMAX;
// Check that we have packet from previous pass
while(count--)
{
if(!refillBuffer(_chunk ))
{
return 0;
}
if(tmptail-tmphead<_chunk)
{
return 0;
}
//printf("Round %d\n",ROUNDMAX-count);
if(vorbis_analysis_blockout(&VD, &VB) == 1)
{
vorbis_analysis(&VB, NULL);
vorbis_bitrate_addblock(&VB) ;
//printf("Blockout\n");
if(vorbis_bitrate_flushpacket(&VD, &op))
{
memcpy(dest, op.packet,op.bytes);
*len=op.bytes;
*samples=op.granulepos-_oldpos;
_oldpos=op.granulepos;
// aprintf("1st packet :sampl:%lu len :%lu sample:%lu abs:%llu\n",*samples,op.bytes,total,op.granulepos);
return 1;
}
}
uint32_t nbSample=(tmptail-tmphead)/channels;
if(nbSample>1024) nbSample=1024;
float_samples=vorbis_analysis_buffer(&VD, nbSample) ;
int index=tmphead;
// Put our samples in incoming buffer
//reorderChannels(&(tmpbuffer[tmphead]), nbSample,_incoming->getChannelMapping(),outputChannelMapping);
reorderToPlanar2(&(tmpbuffer[tmphead]),float_samples,nbSample,_incoming->getChannelMapping(),outputChannelMapping);
#if 0
for (int i = 0; i < nbSample; i++)
for (int j = 0; j < channels; j++) {
float_samples[j][i] = tmpbuffer[index++];
if (float_samples[j][i] > 1) float_samples[j][i] = 1;
if (float_samples[j][i] < -1) float_samples[j][i] = -1;
}
#endif
// Buffer full, go go go
vorbis_analysis_wrote(&VD, nbSample) ;
tmphead+=nbSample*channels;
}
return 0;
}
// @todo, nicely set e_o_s flag
void VorbisWriter::onAudioIn(const void* input, unsigned long nframes) {
if(!fp) {
return;
}
if(!is_setup) {
printf("ERROR: onAudioIn(), first call open().\n");
return;
}
float** buffer = vorbis_analysis_buffer(&vd, nframes);
// check if we need to convert the input.
if(format == VW_INT16) {
short int* input_ptr = (short int*)input;
float* out_channel_ptr = NULL;
short int* in_channel_ptr = NULL;
for(int i = 0; i < num_channels; ++i) {
in_channel_ptr = input_ptr + i;
out_channel_ptr = buffer[i];
for(int j = 0; j < nframes; ++j) {
out_channel_ptr[j] = (float(*(in_channel_ptr))) / 32768.0f;
in_channel_ptr += num_channels;
}
}
}
else if(format == VW_FLOAT32) {
int src_dx = 0;
int dest_dx = 0;
float* input_ptr = (float*)input;
float* out_channel_ptr = NULL;
float* in_channel_ptr = NULL;
for(int i = 0; i < num_channels; ++i) {
src_dx = 0;
out_channel_ptr = buffer[i];
in_channel_ptr = input_ptr + i;
for(int j = 0; j < nframes; ++j) {
out_channel_ptr[j] = *in_channel_ptr;
in_channel_ptr += num_channels;
}
}
}
int r = vorbis_analysis_wrote(&vd, nframes);
if (r != 0) {
printf("ERROR: error with vorbis_analysis_wrote\n");
}
// write out
while(vorbis_analysis_blockout(&vd, &vb) == 1) {
vorbis_analysis(&vb, NULL);
vorbis_bitrate_addblock(&vb);
while(vorbis_bitrate_flushpacket(&vd, &op)) {
ogg_stream_packetin(&os, &op);
while(true) {
int result = ogg_stream_pageout(&os, &og);
if(result == 0) {
break;
}
fwrite(og.header, 1, og.header_len, fp);
fwrite(og.body, 1, og.body_len, fp);
}
}
}
}
/****************************************************************************
* Encode: the whole thing
****************************************************************************
* This function spits out ogg packets.
****************************************************************************/
static block_t *Encode( encoder_t *p_enc, block_t *p_aout_buf )
{
encoder_sys_t *p_sys = p_enc->p_sys;
ogg_packet oggpacket;
block_t *p_block, *p_chain = NULL;
float **buffer;
/* FIXME: flush buffers in here */
if( unlikely( !p_aout_buf ) ) return NULL;
mtime_t i_pts = p_aout_buf->i_pts -
(mtime_t)1000000 * (mtime_t)p_sys->i_samples_delay /
(mtime_t)p_enc->fmt_in.audio.i_rate;
p_sys->i_samples_delay += p_aout_buf->i_nb_samples;
buffer = vorbis_analysis_buffer( &p_sys->vd, p_aout_buf->i_nb_samples );
/* convert samples to float and uninterleave */
for( unsigned int i = 0; i < p_sys->i_channels; i++ )
{
for( unsigned int j = 0 ; j < p_aout_buf->i_nb_samples ; j++ )
{
buffer[i][j]= ((float *)p_aout_buf->p_buffer)
[j * p_sys->i_channels + p_sys->pi_chan_table[i]];
}
}
vorbis_analysis_wrote( &p_sys->vd, p_aout_buf->i_nb_samples );
while( vorbis_analysis_blockout( &p_sys->vd, &p_sys->vb ) == 1 )
{
int i_samples;
vorbis_analysis( &p_sys->vb, NULL );
vorbis_bitrate_addblock( &p_sys->vb );
while( vorbis_bitrate_flushpacket( &p_sys->vd, &oggpacket ) )
{
int i_block_size;
p_block = block_Alloc( oggpacket.bytes );
memcpy( p_block->p_buffer, oggpacket.packet, oggpacket.bytes );
i_block_size = vorbis_packet_blocksize( &p_sys->vi, &oggpacket );
if( i_block_size < 0 ) i_block_size = 0;
i_samples = ( p_sys->i_last_block_size + i_block_size ) >> 2;
p_sys->i_last_block_size = i_block_size;
p_block->i_length = (mtime_t)1000000 *
(mtime_t)i_samples / (mtime_t)p_enc->fmt_in.audio.i_rate;
p_block->i_dts = p_block->i_pts = i_pts;
p_sys->i_samples_delay -= i_samples;
/* Update pts */
i_pts += p_block->i_length;
block_ChainAppend( &p_chain, p_block );
}
}
return p_chain;
}
void *EncodeSoundBuffer(ProgData *pdata){
int sampread=pdata->periodsize;
#ifdef HAVE_LIBJACK
void *jackbuf=NULL;
if(pdata->args.use_jack){
jackbuf=malloc(pdata->sound_framesize*pdata->jdata->buffersize);
}
#endif
pdata->v_encoding_clean=0;
while((pdata->running)){
float **vorbis_buffer;
int count=0,i,j;
SndBuffer *buff=NULL;
if (pdata->paused) {
pthread_mutex_lock(&pdata->pause_mutex);
pthread_cond_wait(&pdata->pause_cond, &pdata->pause_mutex);
pthread_mutex_unlock(&pdata->pause_mutex);
}
if(!pdata->args.use_jack){
if(pdata->sound_buffer==NULL){
pdata->v_enc_thread_waiting=1;
pthread_mutex_lock(&pdata->snd_buff_ready_mutex);
pthread_cond_wait(&pdata->sound_data_read,
&pdata->snd_buff_ready_mutex);
pthread_mutex_unlock(&pdata->snd_buff_ready_mutex);
pdata->v_enc_thread_waiting=0;
}
if(pdata->sound_buffer==NULL || !pdata->running)
break;
pthread_mutex_lock(&pdata->sound_buffer_mutex);
buff=pdata->sound_buffer;
//advance the list
pdata->sound_buffer=pdata->sound_buffer->next;
pthread_mutex_unlock(&pdata->sound_buffer_mutex);
vorbis_buffer=vorbis_analysis_buffer(&pdata->enc_data->m_vo_dsp,
sampread);
for(i=0;i<sampread;i++){
for(j=0;j<pdata->args.channels;j++){
vorbis_buffer[j][i]=((buff->data[count+1]<<8)|
(0x00ff&(int)buff->data[count]))/
32768.f;
count+=2;
}
}
free(buff->data);
free(buff);
}
else{
#ifdef HAVE_LIBJACK
if((*jack_ringbuffer_read_space)(pdata->jdata->sound_buffer)>=
(pdata->sound_framesize*pdata->jdata->buffersize)){
(*jack_ringbuffer_read)(pdata->jdata->sound_buffer,
jackbuf,
(pdata->sound_framesize*
pdata->jdata->buffersize));
vorbis_buffer=vorbis_analysis_buffer(&pdata->enc_data->m_vo_dsp,
sampread);
for(j=0;j<pdata->args.channels;j++){
for(i=0;i<sampread;i++){
vorbis_buffer[j][i]=((float*)jackbuf)[count];
count++;
}
}
}
else{
pdata->v_enc_thread_waiting=1;
pthread_mutex_lock(&pdata->snd_buff_ready_mutex);
pthread_cond_wait(&pdata->sound_data_read,
&pdata->snd_buff_ready_mutex);
pthread_mutex_unlock(&pdata->snd_buff_ready_mutex);
pdata->v_enc_thread_waiting=0;
continue;
}
#endif
}
vorbis_analysis_wrote(&pdata->enc_data->m_vo_dsp,sampread);
pthread_mutex_lock(&pdata->libogg_mutex);
while(vorbis_analysis_blockout(&pdata->enc_data->m_vo_dsp,
&pdata->enc_data->m_vo_block)==1){
vorbis_analysis(&pdata->enc_data->m_vo_block,NULL);
vorbis_bitrate_addblock(&pdata->enc_data->m_vo_block);
while(vorbis_bitrate_flushpacket(&pdata->enc_data->m_vo_dsp,
&pdata->enc_data->m_ogg_pckt2)){
ogg_stream_packetin(&pdata->enc_data->m_ogg_vs,
&pdata->enc_data->m_ogg_pckt2);
}
}
pthread_mutex_unlock(&pdata->libogg_mutex);
pdata->avd-=pdata->periodtime;
}
pdata->v_encoding_clean=1;
pthread_mutex_lock(&pdata->vorbis_lib_mutex);
pthread_cond_signal(&pdata->vorbis_lib_clean);
pthread_mutex_unlock(&pdata->vorbis_lib_mutex);
pthread_exit(&errno);
}
//Method to start encoding
int startEncoding(JNIEnv *env, jclass *cls_ptr, jlong *sampleRate_ptr, jlong *channels_ptr, jfloat *quality_ptr, jlong *bitrate_ptr, jobject *encoderDataFeed_ptr, int type) {
//Dereference our variables
jclass cls = (*cls_ptr);
jlong sampleRate = (*sampleRate_ptr);
jlong channels = (*channels_ptr);
jfloat quality = (*quality_ptr);
jlong bitrate = (*bitrate_ptr);
jobject encoderDataFeed = (*encoderDataFeed_ptr);
//Create our PCM data buffer
signed char readbuffer[READ*4+44];
//Create a new java byte array to pass to the data feed method
jbyteArray jByteArrayBuffer = (*env)->NewByteArray(env, READ*4);
//Create a new java byte buffer to write to
jbyteArray jByteArrayWriteBuffer = (*env)->NewByteArray(env, READ*8);
//Find our java classes we'll be calling
jclass encoderDataFeedClass = (*env)->FindClass(env, "org/xiph/vorbis/encoder/EncodeFeed");
//Find our java method id's we'll be calling
jmethodID writeVorbisDataMethodId = (*env)->GetMethodID(env, encoderDataFeedClass, "writeVorbisData", "([BI)I");
jmethodID readPCMDataMethodId = (*env)->GetMethodID(env, encoderDataFeedClass, "readPCMData", "([BI)J");
jmethodID startMethodId = (*env)->GetMethodID(env, encoderDataFeedClass, "start", "()V");
jmethodID stopMethodId = (*env)->GetMethodID(env, encoderDataFeedClass, "stop", "()V");
ogg_stream_state os; /* take physical pages, weld into a logical
stream of packets */
ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
ogg_packet op; /* one raw packet of data for decode */
vorbis_info vi; /* struct that stores all the static vorbis bitstream
settings */
vorbis_comment vc; /* struct that stores all the user comments */
vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
vorbis_block vb; /* local working space for packet->PCM decode */
int eos=0,ret;
int i, founddata;
/********** Encode setup ************/
__android_log_print(ANDROID_LOG_INFO, "VorbisEncoder", "Setting up encoding");
vorbis_info_init(&vi);
/* choose an encoding mode. A few possibilities commented out, one
actually used: */
/*********************************************************************
Encoding using a VBR quality mode. The usable range is -.1
(lowest quality, smallest file) to 1. (highest quality, largest file).
Example quality mode .4: 44kHz stereo coupled, roughly 128kbps VBR
ret = vorbis_encode_init_vbr(&vi,2,44100,.4);
---------------------------------------------------------------------
Encoding using an average bitrate mode (ABR).
example: 44kHz stereo coupled, average 128kbps VBR
ret = vorbis_encode_init(&vi,2,44100,-1,128000,-1);
---------------------------------------------------------------------
Encode using a quality mode, but select that quality mode by asking for
an approximate bitrate. This is not ABR, it is true VBR, but selected
using the bitrate interface, and then turning bitrate management off:
ret = ( vorbis_encode_setup_managed(&vi,2,44100,-1,128000,-1) ||
vorbis_encode_ctl(&vi,OV_ECTL_RATEMANAGE2_SET,NULL) ||
vorbis_encode_setup_init(&vi));
*********************************************************************/
switch(type) {
case WITH_BITRATE:
__android_log_print(ANDROID_LOG_INFO, "VorbisEncoder", "Initializing with %lld channels %lldHz sample rate and %lld bitrate", channels, sampleRate, bitrate);
ret=vorbis_encode_init(&vi, (long)channels, (long)sampleRate, (long)-1, (long)bitrate, (long)-1);
break;
case WITH_QUALITY:
__android_log_print(ANDROID_LOG_INFO, "VorbisEncoder", "Initializing with %lld channels %lldHz sample rate and %f quality", channels, sampleRate, quality);
ret=vorbis_encode_init_vbr(&vi, (long)channels, (long)sampleRate, (float)quality);
break;
default:
__android_log_print(ANDROID_LOG_ERROR, "VorbisEncoder", "Failed to initialize");
stopEncodeFeed(env, &encoderDataFeed, &stopMethodId);
return ERROR_INITIALIZING;
}
/* do not continue if setup failed; this can happen if we ask for a
mode that libVorbis does not support (eg, too low a bitrate, etc,
will return 'OV_EIMPL') */
if(ret) {
__android_log_print(ANDROID_LOG_ERROR, "VorbisEncoder", "Failed to initialize");
stopEncodeFeed(env, &encoderDataFeed, &stopMethodId);
return ERROR_INITIALIZING;
}
startEncodeFeed(env, &encoderDataFeed, &startMethodId);
/* add a comment */
__android_log_print(ANDROID_LOG_DEBUG, "VorbisEncoder", "Adding comments");
vorbis_comment_init(&vc);
vorbis_comment_add_tag(&vc,"ENCODER","JNIVorbisEncoder");
/* set up the analysis state and auxiliary encoding storage */
vorbis_analysis_init(&vd,&vi);
vorbis_block_init(&vd,&vb);
/* set up our packet->stream encoder */
/* pick a random serial number; that way we can more likely build
chained streams just by concatenation */
srand(time(NULL));
ogg_stream_init(&os,rand());
/* Vorbis streams begin with three headers; the initial header (with
most of the codec setup parameters) which is mandated by the Ogg
bitstream spec. The second header holds any comment fields. The
third header holds the bitstream codebook. We merely need to
make the headers, then pass them to libvorbis one at a time;
libvorbis handles the additional Ogg bitstream constraints */
{
ogg_packet header;
ogg_packet header_comm;
ogg_packet header_code;
vorbis_analysis_headerout(&vd,&vc,&header,&header_comm,&header_code);
ogg_stream_packetin(&os,&header); /* automatically placed in its own
page */
ogg_stream_packetin(&os,&header_comm);
ogg_stream_packetin(&os,&header_code);
/* This ensures the actual
* audio data will start on a new page, as per spec
*/
__android_log_print(ANDROID_LOG_INFO, "VorbisEncoder", "Writting header");
while(!eos){
int result=ogg_stream_flush(&os,&og);
if(result==0)break;
writeVorbisDataToEncoderDataFeed(env, &encoderDataFeed, &writeVorbisDataMethodId, og.header, og.header_len, &jByteArrayWriteBuffer);
writeVorbisDataToEncoderDataFeed(env, &encoderDataFeed, &writeVorbisDataMethodId, og.body, og.body_len, &jByteArrayWriteBuffer);
}
}
__android_log_print(ANDROID_LOG_INFO, "VorbisEncoder", "Starting to read from pcm callback");
while(!eos){
long i;
long bytes = readPCMDataFromEncoderDataFeed(env, &encoderDataFeed, &readPCMDataMethodId, readbuffer, READ*4, &jByteArrayBuffer);
if(bytes==0){
/* end of file. this can be done implicitly in the mainline,
but it's easier to see here in non-clever fashion.
Tell the library we're at end of stream so that it can handle
the last frame and mark end of stream in the output properly */
__android_log_print(ANDROID_LOG_INFO, "VorbisEncoder", "End of file");
vorbis_analysis_wrote(&vd,0);
}else{
/* data to encode */
/* expose the buffer to submit data */
float **buffer=vorbis_analysis_buffer(&vd,bytes/(2*channels));
/* uninterleave samples */
int channel;
for(i=0;i<bytes/(2*channels);i++) {
for(channel = 0; channel < channels; channel++) {
buffer[channel][i]=((readbuffer[i*(2*channels)+(channel*2+1)]<<8)|
(0x00ff&(int)readbuffer[i*(2*channels)+(channel*2)]))/32768.f;
}
}
/* tell the library how much we actually submitted */
vorbis_analysis_wrote(&vd,i);
}
/* vorbis does some data preanalysis, then divvies up blocks for
more involved (potentially parallel) processing. Get a single
block for encoding now */
while(vorbis_analysis_blockout(&vd,&vb)==1){
/* analysis, assume we want to use bitrate management */
vorbis_analysis(&vb,NULL);
vorbis_bitrate_addblock(&vb);
while(vorbis_bitrate_flushpacket(&vd,&op)){
/* weld the packet into the bitstream */
ogg_stream_packetin(&os,&op);
/* write out pages (if any) */
while(!eos){
int result=ogg_stream_pageout(&os,&og);
if(result==0)break;
writeVorbisDataToEncoderDataFeed(env, &encoderDataFeed, &writeVorbisDataMethodId, og.header, og.header_len, &jByteArrayWriteBuffer);
writeVorbisDataToEncoderDataFeed(env, &encoderDataFeed, &writeVorbisDataMethodId, og.body, og.body_len, &jByteArrayWriteBuffer);
/* this could be set above, but for illustrative purposes, I do
it here (to show that vorbis does know where the stream ends) */
if(ogg_page_eos(&og))eos=1;
}
}
}
}
/* clean up and exit. vorbis_info_clear() must be called last */
__android_log_print(ANDROID_LOG_INFO, "VorbisEncoder", "Cleaning up encoder");
ogg_stream_clear(&os);
vorbis_block_clear(&vb);
vorbis_dsp_clear(&vd);
vorbis_comment_clear(&vc);
vorbis_info_clear(&vi);
/* ogg_page and ogg_packet structs always point to storage in
libvorbis. They're never freed or manipulated directly */
__android_log_print(ANDROID_LOG_INFO, "VorbisEncoder", "Completed encoding.");
stopEncodeFeed(env, &encoderDataFeed, &stopMethodId);
//Clean up encode buffers
(*env)->DeleteLocalRef(env, jByteArrayBuffer);
(*env)->DeleteLocalRef(env, jByteArrayWriteBuffer);
return SUCCESS;
}
int main(){
ogg_stream_state os; /* take physical pages, weld into a logical
stream of packets */
ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
ogg_packet op; /* one raw packet of data for decode */
vorbis_info vi; /* struct that stores all the static vorbis bitstream
settings */
vorbis_comment vc; /* struct that stores all the user comments */
vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
vorbis_block vb; /* local working space for packet->PCM decode */
int eos=0,ret;
int i, founddata;
int numReads=0;
FILE *outFile = fopen("../test.ogg","wb");
/* we cheat on the WAV header; we just bypass 44 bytes (simplest WAV
header is 44 bytes) and assume that the data is 44.1khz, stereo, 16 bit
little endian pcm samples. This is just an example, after all. */
/********** Encode setup ************/
vorbis_info_init(&vi);
/* choose an encoding mode. A few possibilities commented out, one
actually used: */
/*********************************************************************
Encoding using a VBR quality mode. The usable range is -.1
(lowest quality, smallest file) to 1. (highest quality, largest file).
Example quality mode .4: 44kHz stereo coupled, roughly 128kbps VBR
ret = vorbis_encode_init_vbr(&vi,2,44100,.4);
---------------------------------------------------------------------
Encoding using an average bitrate mode (ABR).
example: 44kHz stereo coupled, average 128kbps VBR
ret = vorbis_encode_init(&vi,2,44100,-1,128000,-1);
---------------------------------------------------------------------
Encode using a quality mode, but select that quality mode by asking for
an approximate bitrate. This is not ABR, it is true VBR, but selected
using the bitrate interface, and then turning bitrate management off:
ret = ( vorbis_encode_setup_managed(&vi,2,44100,-1,128000,-1) ||
vorbis_encode_ctl(&vi,OV_ECTL_RATEMANAGE2_SET,NULL) ||
vorbis_encode_setup_init(&vi));
*********************************************************************/
ret=vorbis_encode_init_vbr(&vi,2,44100,1.0);
/* do not continue if setup failed; this can happen if we ask for a
mode that libVorbis does not support (eg, too low a bitrate, etc,
will return 'OV_EIMPL') */
if(ret)exit(1);
/* add a comment */
vorbis_comment_init(&vc);
vorbis_comment_add_tag(&vc,"ENCODER","encoder_example.c");
/* set up the analysis state and auxiliary encoding storage */
vorbis_analysis_init(&vd,&vi);
vorbis_block_init(&vd,&vb);
/* set up our packet->stream encoder */
/* pick a random serial number; that way we can more likely build
chained streams just by concatenation */
srand(time(NULL));
ogg_stream_init(&os,rand());
/* Vorbis streams begin with three headers; the initial header (with
most of the codec setup parameters) which is mandated by the Ogg
bitstream spec. The second header holds any comment fields. The
third header holds the bitstream codebook. We merely need to
make the headers, then pass them to libvorbis one at a time;
libvorbis handles the additional Ogg bitstream constraints */
{
ogg_packet header;
ogg_packet header_comm;
ogg_packet header_code;
vorbis_analysis_headerout(&vd,&vc,&header,&header_comm,&header_code);
ogg_stream_packetin(&os,&header); /* automatically placed in its own
page */
ogg_stream_packetin(&os,&header_comm);
ogg_stream_packetin(&os,&header_code);
/* This ensures the actual
* audio data will start on a new page, as per spec
*/
while(!eos){
int result=ogg_stream_flush(&os,&og);
if(result==0)break;
fwrite(og.header,1,og.header_len,outFile);
fwrite(og.body,1,og.body_len,outFile);
}
}
while(!eos){
long i;
long bytes=READ*4; /* stereo hardwired here */
numReads++;
if(numReads>10000) bytes=0;
if(bytes==0){
/* end of file. this can be done implicitly in the mainline,
but it's easier to see here in non-clever fashion.
Tell the library we're at end of stream so that it can handle
the last frame and mark end of stream in the output properly */
vorbis_analysis_wrote(&vd,0);
}else{
/* data to encode */
/* expose the buffer to submit data */
float **buffer=vorbis_analysis_buffer(&vd,READ);
/* uninterleave samples */
for(i=0;i<bytes/4;i++){
static float curamp = 0;
curamp += 0.15f;
buffer[0][i] = sinf(curamp);
buffer[1][i] = sinf(curamp);
}
/* tell the library how much we actually submitted */
vorbis_analysis_wrote(&vd,i);
}
/* vorbis does some data preanalysis, then divvies up blocks for
more involved (potentially parallel) processing. Get a single
block for encoding now */
while(vorbis_analysis_blockout(&vd,&vb)==1){
/* analysis, assume we want to use bitrate management */
vorbis_analysis(&vb,NULL);
vorbis_bitrate_addblock(&vb);
while(vorbis_bitrate_flushpacket(&vd,&op)){
/* weld the packet into the bitstream */
ogg_stream_packetin(&os,&op);
/* write out pages (if any) */
while(!eos){
int result=ogg_stream_pageout(&os,&og);
if(result==0)break;
fwrite(og.header,1,og.header_len,outFile);
fwrite(og.body,1,og.body_len,outFile);
/* this could be set above, but for illustrative purposes, I do
it here (to show that vorbis does know where the stream ends) */
if(ogg_page_eos(&og))eos=1;
}
}
}
}
/* clean up and exit. vorbis_info_clear() must be called last */
ogg_stream_clear(&os);
vorbis_block_clear(&vb);
vorbis_dsp_clear(&vd);
vorbis_comment_clear(&vc);
vorbis_info_clear(&vi);
/* ogg_page and ogg_packet structs always point to storage in
libvorbis. They're never freed or manipulated directly */
fprintf(stderr,"Done.\n");
return(0);
}
int ExportOGG::Export(AudacityProject *project,
int numChannels,
wxString fName,
bool selectionOnly,
double t0,
double t1,
MixerSpec *mixerSpec,
Tags *metadata,
int WXUNUSED(subformat))
{
double rate = project->GetRate();
TrackList *tracks = project->GetTracks();
double quality = (gPrefs->Read(wxT("/FileFormats/OggExportQuality"), 50)/(float)100.0);
wxLogNull logNo; // temporarily disable wxWidgets error messages
int updateResult = eProgressSuccess;
int eos = 0;
FileIO outFile(fName, FileIO::Output);
if (!outFile.IsOpened()) {
wxMessageBox(_("Unable to open target file for writing"));
return false;
}
// All the Ogg and Vorbis encoding data
ogg_stream_state stream;
ogg_page page;
ogg_packet packet;
vorbis_info info;
vorbis_comment comment;
vorbis_dsp_state dsp;
vorbis_block block;
// Encoding setup
vorbis_info_init(&info);
vorbis_encode_init_vbr(&info, numChannels, int(rate + 0.5), quality);
// Retrieve tags
if (!FillComment(project, &comment, metadata)) {
return false;
}
// Set up analysis state and auxiliary encoding storage
vorbis_analysis_init(&dsp, &info);
vorbis_block_init(&dsp, &block);
// Set up packet->stream encoder. According to encoder example,
// a random serial number makes it more likely that you can make
// chained streams with concatenation.
srand(time(NULL));
ogg_stream_init(&stream, rand());
// First we need to write the required headers:
// 1. The Ogg bitstream header, which contains codec setup params
// 2. The Vorbis comment header
// 3. The bitstream codebook.
//
// After we create those our responsibility is complete, libvorbis will
// take care of any other ogg bistream constraints (again, according
// to the example encoder source)
ogg_packet bitstream_header;
ogg_packet comment_header;
ogg_packet codebook_header;
vorbis_analysis_headerout(&dsp, &comment, &bitstream_header, &comment_header,
&codebook_header);
// Place these headers into the stream
ogg_stream_packetin(&stream, &bitstream_header);
ogg_stream_packetin(&stream, &comment_header);
ogg_stream_packetin(&stream, &codebook_header);
// Flushing these headers now guarentees that audio data will
// start on a new page, which apparently makes streaming easier
while (ogg_stream_flush(&stream, &page)) {
outFile.Write(page.header, page.header_len);
outFile.Write(page.body, page.body_len);
}
int numWaveTracks;
WaveTrack **waveTracks;
tracks->GetWaveTracks(selectionOnly, &numWaveTracks, &waveTracks);
Mixer *mixer = CreateMixer(numWaveTracks, waveTracks,
tracks->GetTimeTrack(),
t0, t1,
numChannels, SAMPLES_PER_RUN, false,
rate, floatSample, true, mixerSpec);
delete [] waveTracks;
ProgressDialog *progress = new ProgressDialog(wxFileName(fName).GetName(),
selectionOnly ?
_("Exporting the selected audio as Ogg Vorbis") :
_("Exporting the entire project as Ogg Vorbis"));
while (updateResult == eProgressSuccess && !eos) {
float **vorbis_buffer = vorbis_analysis_buffer(&dsp, SAMPLES_PER_RUN);
sampleCount samplesThisRun = mixer->Process(SAMPLES_PER_RUN);
if (samplesThisRun == 0) {
// Tell the library that we wrote 0 bytes - signalling the end.
vorbis_analysis_wrote(&dsp, 0);
}
else {
for (int i = 0; i < numChannels; i++) {
float *temp = (float *)mixer->GetBuffer(i);
memcpy(vorbis_buffer[i], temp, sizeof(float)*SAMPLES_PER_RUN);
}
// tell the encoder how many samples we have
vorbis_analysis_wrote(&dsp, samplesThisRun);
}
// I don't understand what this call does, so here is the comment
// from the example, verbatim:
//
// vorbis does some data preanalysis, then divvies up blocks
// for more involved (potentially parallel) processing. Get
// a single block for encoding now
while (vorbis_analysis_blockout(&dsp, &block) == 1) {
// analysis, assume we want to use bitrate management
vorbis_analysis(&block, NULL);
vorbis_bitrate_addblock(&block);
while (vorbis_bitrate_flushpacket(&dsp, &packet)) {
// add the packet to the bitstream
ogg_stream_packetin(&stream, &packet);
// From vorbis-tools-1.0/oggenc/encode.c:
// If we've gone over a page boundary, we can do actual output,
// so do so (for however many pages are available).
while (!eos) {
int result = ogg_stream_pageout(&stream, &page);
if (!result) {
break;
}
outFile.Write(page.header, page.header_len);
outFile.Write(page.body, page.body_len);
if (ogg_page_eos(&page)) {
eos = 1;
}
}
}
}
updateResult = progress->Update(mixer->MixGetCurrentTime()-t0, t1-t0);
}
delete progress;;
delete mixer;
ogg_stream_clear(&stream);
vorbis_block_clear(&block);
vorbis_dsp_clear(&dsp);
vorbis_info_clear(&info);
vorbis_comment_clear(&comment);
outFile.Close();
return updateResult;
}
/* The following function is basically a hacked version of the code in
* examples/encoder_example.c */
void
write_vorbis_data_or_die (const char *filename, int srate, float q, const float * data, int count, int ch)
{
FILE * file ;
ogg_stream_state os;
ogg_page og;
ogg_packet op;
vorbis_info vi;
vorbis_comment vc;
vorbis_dsp_state vd;
vorbis_block vb;
int eos = 0, ret;
if ((file = fopen (filename, "wb")) == NULL) {
printf("\n\nError : fopen failed : %s\n", strerror (errno)) ;
exit (1) ;
}
/********** Encode setup ************/
vorbis_info_init (&vi);
ret = vorbis_encode_init_vbr (&vi,ch,srate,q);
if (ret) {
printf ("vorbis_encode_init_vbr return %d\n", ret) ;
exit (1) ;
}
vorbis_comment_init (&vc);
vorbis_comment_add_tag (&vc,"ENCODER","test/util.c");
vorbis_analysis_init (&vd,&vi);
vorbis_block_init (&vd,&vb);
ogg_stream_init (&os,12345678);
{
ogg_packet header;
ogg_packet header_comm;
ogg_packet header_code;
vorbis_analysis_headerout (&vd,&vc,&header,&header_comm,&header_code);
ogg_stream_packetin (&os,&header);
ogg_stream_packetin (&os,&header_comm);
ogg_stream_packetin (&os,&header_code);
/* Ensures the audio data will start on a new page. */
while (!eos){
int result = ogg_stream_flush (&os,&og);
if (result == 0)
break;
fwrite (og.header,1,og.header_len,file);
fwrite (og.body,1,og.body_len,file);
}
}
{
/* expose the buffer to submit data */
float **buffer = vorbis_analysis_buffer (&vd,count);
int i;
for(i=0;i<ch;i++)
memcpy (buffer [i], data, count * sizeof (float)) ;
/* tell the library how much we actually submitted */
vorbis_analysis_wrote (&vd,count);
vorbis_analysis_wrote (&vd,0);
}
while (vorbis_analysis_blockout (&vd,&vb) == 1) {
vorbis_analysis (&vb,NULL);
vorbis_bitrate_addblock (&vb);
while (vorbis_bitrate_flushpacket (&vd,&op)) {
ogg_stream_packetin (&os,&op);
while (!eos) {
int result = ogg_stream_pageout (&os,&og);
if (result == 0)
break;
fwrite (og.header,1,og.header_len,file);
fwrite (og.body,1,og.body_len,file);
if (ogg_page_eos (&og))
eos = 1;
}
}
}
ogg_stream_clear (&os);
vorbis_block_clear (&vb);
vorbis_dsp_clear (&vd);
vorbis_comment_clear (&vc);
vorbis_info_clear (&vi);
fclose (file) ;
}