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/*
* copyright ( c ) 2002 Mark Hills < mark @ pogo . org . uk >
*
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* This file is part of Libav .
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*
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* Libav is free software ; you can redistribute it and / or
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* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation ; either
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* version 2.1 of the License , or ( at your option ) any later version .
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*
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* Libav is distributed in the hope that it will be useful ,
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* but WITHOUT ANY WARRANTY ; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE . See the GNU
* Lesser General Public License for more details .
*
* You should have received a copy of the GNU Lesser General Public
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* License along with Libav ; if not , write to the Free Software
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* Foundation , Inc . , 51 Franklin Street , Fifth Floor , Boston , MA 02110 - 1301 USA
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*/
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/**
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* @ file
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* Vorbis encoding support via libvorbisenc .
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* @ author Mark Hills < mark @ pogo . org . uk >
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*/
# include <vorbis/vorbisenc.h>
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# include "libavutil/fifo.h"
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# include "libavutil/opt.h"
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# include "avcodec.h"
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# include "audio_frame_queue.h"
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# include "bytestream.h"
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# include "internal.h"
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# include "vorbis.h"
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# include "vorbis_parser.h"
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# undef NDEBUG
# include <assert.h>
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/* Number of samples the user should send in each call.
* This value is used because it is the LCD of all possible frame sizes , so
* an output packet will always start at the same point as one of the input
* packets .
*/
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# define LIBVORBIS_FRAME_SIZE 64
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# define BUFFER_SIZE (1024 * 64)
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typedef struct LibvorbisContext {
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AVClass * av_class ; /**< class for AVOptions */
vorbis_info vi ; /**< vorbis_info used during init */
vorbis_dsp_state vd ; /**< DSP state used for analysis */
vorbis_block vb ; /**< vorbis_block used for analysis */
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AVFifoBuffer * pkt_fifo ; /**< output packet buffer */
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int eof ; /**< end-of-file flag */
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int dsp_initialized ; /**< vd has been initialized */
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vorbis_comment vc ; /**< VorbisComment info */
ogg_packet op ; /**< ogg packet */
double iblock ; /**< impulse block bias option */
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VorbisParseContext vp ; /**< parse context to get durations */
AudioFrameQueue afq ; /**< frame queue for timestamps */
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} LibvorbisContext ;
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static const AVOption options [ ] = {
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{ " iblock " , " Sets the impulse block bias " , offsetof ( LibvorbisContext , iblock ) , AV_OPT_TYPE_DOUBLE , { . dbl = 0 } , - 15 , 0 , AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM } ,
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{ NULL }
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} ;
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static const AVCodecDefault defaults [ ] = {
{ " b " , " 0 " } ,
{ NULL } ,
} ;
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static const AVClass class = { " libvorbis " , av_default_item_name , options , LIBAVUTIL_VERSION_INT } ;
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static int vorbis_error_to_averror ( int ov_err )
{
switch ( ov_err ) {
case OV_EFAULT : return AVERROR_BUG ;
case OV_EINVAL : return AVERROR ( EINVAL ) ;
case OV_EIMPL : return AVERROR ( EINVAL ) ;
default : return AVERROR_UNKNOWN ;
}
}
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static av_cold int libvorbis_setup ( vorbis_info * vi , AVCodecContext * avctx )
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{
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LibvorbisContext * s = avctx - > priv_data ;
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double cfreq ;
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int ret ;
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if ( avctx - > flags & CODEC_FLAG_QSCALE | | ! avctx - > bit_rate ) {
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/* variable bitrate
* NOTE : we use the oggenc range of - 1 to 10 for global_quality for
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* user convenience , but libvorbis uses - 0.1 to 1.0 .
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*/
float q = avctx - > global_quality / ( float ) FF_QP2LAMBDA ;
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/* default to 3 if the user did not set quality or bitrate */
if ( ! ( avctx - > flags & CODEC_FLAG_QSCALE ) )
q = 3.0 ;
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if ( ( ret = vorbis_encode_setup_vbr ( vi , avctx - > channels ,
avctx - > sample_rate ,
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q / 10.0 ) ) )
goto error ;
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} else {
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int minrate = avctx - > rc_min_rate > 0 ? avctx - > rc_min_rate : - 1 ;
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int maxrate = avctx - > rc_max_rate > 0 ? avctx - > rc_max_rate : - 1 ;
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/* average bitrate */
if ( ( ret = vorbis_encode_setup_managed ( vi , avctx - > channels ,
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avctx - > sample_rate , maxrate ,
avctx - > bit_rate , minrate ) ) )
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goto error ;
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/* variable bitrate by estimate, disable slow rate management */
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if ( minrate = = - 1 & & maxrate = = - 1 )
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if ( ( ret = vorbis_encode_ctl ( vi , OV_ECTL_RATEMANAGE2_SET , NULL ) ) )
goto error ;
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}
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/* cutoff frequency */
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if ( avctx - > cutoff > 0 ) {
cfreq = avctx - > cutoff / 1000.0 ;
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if ( ( ret = vorbis_encode_ctl ( vi , OV_ECTL_LOWPASS_SET , & cfreq ) ) )
goto error ;
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}
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/* impulse block bias */
if ( s - > iblock ) {
if ( ( ret = vorbis_encode_ctl ( vi , OV_ECTL_IBLOCK_SET , & s - > iblock ) ) )
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goto error ;
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}
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if ( ( ret = vorbis_encode_setup_init ( vi ) ) )
goto error ;
return 0 ;
error :
return vorbis_error_to_averror ( ret ) ;
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}
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/* How many bytes are needed for a buffer of length 'l' */
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static int xiph_len ( int l )
{
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return 1 + l / 255 + l ;
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}
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static av_cold int libvorbis_encode_close ( AVCodecContext * avctx )
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{
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LibvorbisContext * s = avctx - > priv_data ;
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/* notify vorbisenc this is EOF */
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if ( s - > dsp_initialized )
vorbis_analysis_wrote ( & s - > vd , 0 ) ;
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vorbis_block_clear ( & s - > vb ) ;
vorbis_dsp_clear ( & s - > vd ) ;
vorbis_info_clear ( & s - > vi ) ;
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av_fifo_free ( s - > pkt_fifo ) ;
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ff_af_queue_close ( & s - > afq ) ;
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av_freep ( & avctx - > extradata ) ;
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return 0 ;
}
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static av_cold int libvorbis_encode_init ( AVCodecContext * avctx )
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{
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LibvorbisContext * s = avctx - > priv_data ;
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ogg_packet header , header_comm , header_code ;
uint8_t * p ;
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unsigned int offset ;
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int ret ;
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vorbis_info_init ( & s - > vi ) ;
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if ( ( ret = libvorbis_setup ( & s - > vi , avctx ) ) ) {
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av_log ( avctx , AV_LOG_ERROR , " encoder setup failed \n " ) ;
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goto error ;
}
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if ( ( ret = vorbis_analysis_init ( & s - > vd , & s - > vi ) ) ) {
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av_log ( avctx , AV_LOG_ERROR , " analysis init failed \n " ) ;
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ret = vorbis_error_to_averror ( ret ) ;
goto error ;
}
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s - > dsp_initialized = 1 ;
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if ( ( ret = vorbis_block_init ( & s - > vd , & s - > vb ) ) ) {
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av_log ( avctx , AV_LOG_ERROR , " dsp init failed \n " ) ;
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ret = vorbis_error_to_averror ( ret ) ;
goto error ;
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}
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vorbis_comment_init ( & s - > vc ) ;
vorbis_comment_add_tag ( & s - > vc , " encoder " , LIBAVCODEC_IDENT ) ;
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if ( ( ret = vorbis_analysis_headerout ( & s - > vd , & s - > vc , & header , & header_comm ,
& header_code ) ) ) {
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ret = vorbis_error_to_averror ( ret ) ;
goto error ;
}
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avctx - > extradata_size = 1 + xiph_len ( header . bytes ) +
xiph_len ( header_comm . bytes ) +
header_code . bytes ;
p = avctx - > extradata = av_malloc ( avctx - > extradata_size +
FF_INPUT_BUFFER_PADDING_SIZE ) ;
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if ( ! p ) {
ret = AVERROR ( ENOMEM ) ;
goto error ;
}
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p [ 0 ] = 2 ;
offset = 1 ;
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offset + = av_xiphlacing ( & p [ offset ] , header . bytes ) ;
offset + = av_xiphlacing ( & p [ offset ] , header_comm . bytes ) ;
memcpy ( & p [ offset ] , header . packet , header . bytes ) ;
offset + = header . bytes ;
memcpy ( & p [ offset ] , header_comm . packet , header_comm . bytes ) ;
offset + = header_comm . bytes ;
memcpy ( & p [ offset ] , header_code . packet , header_code . bytes ) ;
offset + = header_code . bytes ;
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assert ( offset = = avctx - > extradata_size ) ;
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if ( ( ret = avpriv_vorbis_parse_extradata ( avctx , & s - > vp ) ) < 0 ) {
av_log ( avctx , AV_LOG_ERROR , " invalid extradata \n " ) ;
return ret ;
}
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vorbis_comment_clear ( & s - > vc ) ;
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avctx - > frame_size = LIBVORBIS_FRAME_SIZE ;
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ff_af_queue_init ( avctx , & s - > afq ) ;
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s - > pkt_fifo = av_fifo_alloc ( BUFFER_SIZE ) ;
if ( ! s - > pkt_fifo ) {
ret = AVERROR ( ENOMEM ) ;
goto error ;
}
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return 0 ;
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error :
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libvorbis_encode_close ( avctx ) ;
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return ret ;
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}
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static int libvorbis_encode_frame ( AVCodecContext * avctx , AVPacket * avpkt ,
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const AVFrame * frame , int * got_packet_ptr )
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{
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LibvorbisContext * s = avctx - > priv_data ;
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ogg_packet op ;
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int ret , duration ;
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/* send samples to libvorbis */
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if ( frame ) {
const int samples = frame - > nb_samples ;
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float * * buffer ;
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int c , channels = s - > vi . channels ;
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buffer = vorbis_analysis_buffer ( & s - > vd , samples ) ;
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for ( c = 0 ; c < channels ; c + + ) {
int co = ( channels > 8 ) ? c :
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ff_vorbis_encoding_channel_layout_offsets [ channels - 1 ] [ c ] ;
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memcpy ( buffer [ c ] , frame - > extended_data [ co ] ,
samples * sizeof ( * buffer [ c ] ) ) ;
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}
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if ( ( ret = vorbis_analysis_wrote ( & s - > vd , samples ) ) < 0 ) {
av_log ( avctx , AV_LOG_ERROR , " error in vorbis_analysis_wrote() \n " ) ;
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return vorbis_error_to_averror ( ret ) ;
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}
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if ( ( ret = ff_af_queue_add ( & s - > afq , frame ) ) < 0 )
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return ret ;
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} else {
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if ( ! s - > eof )
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if ( ( ret = vorbis_analysis_wrote ( & s - > vd , 0 ) ) < 0 ) {
av_log ( avctx , AV_LOG_ERROR , " error in vorbis_analysis_wrote() \n " ) ;
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return vorbis_error_to_averror ( ret ) ;
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}
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s - > eof = 1 ;
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}
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/* retrieve available packets from libvorbis */
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while ( ( ret = vorbis_analysis_blockout ( & s - > vd , & s - > vb ) ) = = 1 ) {
if ( ( ret = vorbis_analysis ( & s - > vb , NULL ) ) < 0 )
break ;
if ( ( ret = vorbis_bitrate_addblock ( & s - > vb ) ) < 0 )
break ;
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/* add any available packets to the output packet buffer */
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while ( ( ret = vorbis_bitrate_flushpacket ( & s - > vd , & op ) ) = = 1 ) {
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if ( av_fifo_space ( s - > pkt_fifo ) < sizeof ( ogg_packet ) + op . bytes ) {
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av_log ( avctx , AV_LOG_ERROR , " packet buffer is too small " ) ;
return AVERROR_BUG ;
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}
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av_fifo_generic_write ( s - > pkt_fifo , & op , sizeof ( ogg_packet ) , NULL ) ;
av_fifo_generic_write ( s - > pkt_fifo , op . packet , op . bytes , NULL ) ;
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}
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if ( ret < 0 ) {
av_log ( avctx , AV_LOG_ERROR , " error getting available packets \n " ) ;
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break ;
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}
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}
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if ( ret < 0 ) {
av_log ( avctx , AV_LOG_ERROR , " error getting available packets \n " ) ;
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return vorbis_error_to_averror ( ret ) ;
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}
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/* check for available packets */
if ( av_fifo_size ( s - > pkt_fifo ) < sizeof ( ogg_packet ) )
return 0 ;
av_fifo_generic_read ( s - > pkt_fifo , & op , sizeof ( ogg_packet ) , NULL ) ;
if ( ( ret = ff_alloc_packet ( avpkt , op . bytes ) ) ) {
av_log ( avctx , AV_LOG_ERROR , " Error getting output packet \n " ) ;
return ret ;
}
av_fifo_generic_read ( s - > pkt_fifo , avpkt - > data , op . bytes , NULL ) ;
avpkt - > pts = ff_samples_to_time_base ( avctx , op . granulepos ) ;
duration = avpriv_vorbis_parse_frame ( & s - > vp , avpkt - > data , avpkt - > size ) ;
if ( duration > 0 ) {
/* we do not know encoder delay until we get the first packet from
* libvorbis , so we have to update the AudioFrameQueue counts */
if ( ! avctx - > delay ) {
avctx - > delay = duration ;
s - > afq . remaining_delay + = duration ;
s - > afq . remaining_samples + = duration ;
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}
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ff_af_queue_remove ( & s - > afq , duration , & avpkt - > pts , & avpkt - > duration ) ;
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}
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* got_packet_ptr = 1 ;
return 0 ;
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}
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AVCodec ff_libvorbis_encoder = {
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. name = " libvorbis " ,
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. long_name = NULL_IF_CONFIG_SMALL ( " libvorbis Vorbis " ) ,
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. type = AVMEDIA_TYPE_AUDIO ,
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. id = AV_CODEC_ID_VORBIS ,
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. priv_data_size = sizeof ( LibvorbisContext ) ,
. init = libvorbis_encode_init ,
. encode2 = libvorbis_encode_frame ,
. close = libvorbis_encode_close ,
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. capabilities = CODEC_CAP_DELAY ,
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. sample_fmts = ( const enum AVSampleFormat [ ] ) { AV_SAMPLE_FMT_FLTP ,
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AV_SAMPLE_FMT_NONE } ,
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. priv_class = & class ,
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. defaults = defaults ,
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} ;