vorbisencode.cpp

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#include "linden_common.h"

#include "vorbisencode.h"
#include "vorbis/vorbisenc.h"
#include "llerror.h"
#include "llrand.h"
#include "llmath.h"
#include "llapr.h"

//#if LL_DARWIN
// MBW -- XXX -- Getting rid of SecondLifeVorbis for now -- no fmod means no name collisions.
#if 0
#include "VorbisFramework.h"

#define vorbis_analysis                         mac_vorbis_analysis
#define vorbis_analysis_headerout       mac_vorbis_analysis_headerout
#define vorbis_analysis_init            mac_vorbis_analysis_init
#define vorbis_encode_ctl                       mac_vorbis_encode_ctl
#define vorbis_encode_setup_init        mac_vorbis_encode_setup_init
#define vorbis_encode_setup_managed     mac_vorbis_encode_setup_managed

#define vorbis_info_init                        mac_vorbis_info_init
#define vorbis_info_clear                       mac_vorbis_info_clear
#define vorbis_comment_init                     mac_vorbis_comment_init
#define vorbis_comment_clear            mac_vorbis_comment_clear
#define vorbis_block_init                       mac_vorbis_block_init
#define vorbis_block_clear                      mac_vorbis_block_clear
#define vorbis_dsp_clear                        mac_vorbis_dsp_clear
#define vorbis_analysis_buffer          mac_vorbis_analysis_buffer
#define vorbis_analysis_wrote           mac_vorbis_analysis_wrote
#define vorbis_analysis_blockout        mac_vorbis_analysis_blockout

#define ogg_stream_packetin                     mac_ogg_stream_packetin
#define ogg_stream_init                         mac_ogg_stream_init
#define ogg_stream_flush                        mac_ogg_stream_flush
#define ogg_stream_pageout                      mac_ogg_stream_pageout
#define ogg_page_eos                            mac_ogg_page_eos
#define ogg_stream_clear                        mac_ogg_stream_clear

#endif

S32 check_for_invalid_wav_formats(const char *in_fname, char *error_msg)
{
        U16 num_channels = 0;
        U32 sample_rate = 0;
        U32 bits_per_sample = 0;
        U32 physical_file_size = 0;
        U32 chunk_length = 0;
        U32 raw_data_length = 0;
        U32 bytes_per_sec = 0;
        BOOL uncompressed_pcm = FALSE;

        unsigned char wav_header[44];           /*Flawfinder: ignore*/

        error_msg[0] = '\0';

    apr_file_t* infp = ll_apr_file_open(in_fname,LL_APR_RB);
        if (!infp)
        {
                strcpy(error_msg, "CannotUploadSoundFile");     /*Flawfinder: ignore*/
                return(LLVORBISENC_SOURCE_OPEN_ERR);
        }

        ll_apr_file_read(infp, wav_header, 44);
        physical_file_size = ll_apr_file_seek(infp,APR_END,0);

        if (strncmp((char *)&(wav_header[0]),"RIFF",4))
        {
                strcpy(error_msg, "SoundFileNotRIFF");  /*Flawfinder: ignore*/
                apr_file_close(infp);
            return(LLVORBISENC_WAV_FORMAT_ERR);
        }

        if (strncmp((char *)&(wav_header[8]),"WAVE",4))
        {
                strcpy(error_msg, "SoundFileNotRIFF");  /*Flawfinder: ignore*/
                apr_file_close(infp);
            return(LLVORBISENC_WAV_FORMAT_ERR);
        }
        
        // parse the chunks
        
        U32 file_pos = 12;  // start at the first chunk (usually fmt but not always)
        
        while ((file_pos + 8)< physical_file_size)
        {
                ll_apr_file_seek(infp,APR_SET,file_pos);
                ll_apr_file_read(infp, 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)))
                {
                        if ((wav_header[8] == 0x01) && (wav_header[9] == 0x00))
                        {
                                uncompressed_pcm = TRUE;
                        }
                        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];
                        bytes_per_sec = ((U32) wav_header[19] << 24) 
                                + ((U32) wav_header[18] << 16) 
                                + ((U32) wav_header[17] << 8) 
                                + wav_header[16];
                }
                else if (!(strncmp((char *)&(wav_header[0]),"data",4)))
                {
                        raw_data_length = chunk_length;                 
                }
                file_pos += (chunk_length + 8);
                chunk_length = 0;
        } 

        apr_file_close(infp);   

        if (!uncompressed_pcm)
        {       
                 strcpy(error_msg, "SoundFileNotPCM");          /*Flawfinder: ignore*/
                  return(LLVORBISENC_PCM_FORMAT_ERR);
        }
        
        if ((num_channels < 1) || (num_channels > 2))
        {       
                strcpy(error_msg, "SoundFileInvalidChannelCount");      /*Flawfinder: ignore*/
                return(LLVORBISENC_MULTICHANNEL_ERR);
        }

        if (sample_rate != 44100)
        {       
                strcpy(error_msg, "SoundFileInvalidSampleRate");                /*Flawfinder: ignore*/
                return(LLVORBISENC_UNSUPPORTED_SAMPLE_RATE);
        }
        
        if ((bits_per_sample != 16) && (bits_per_sample != 8))
        {                
                strcpy(error_msg, "SoundFileInvalidWordSize");          /*Flawfinder: ignore*/
                return(LLVORBISENC_UNSUPPORTED_WORD_SIZE);
        }

        if (!raw_data_length)
        {
                strcpy(error_msg, "SoundFileInvalidHeader");                    /*Flawfinder: ignore*/
                return(LLVORBISENC_CLIP_TOO_LONG);               
        }

        F32 clip_length = (F32)raw_data_length/(F32)bytes_per_sec;
                
        if (clip_length > 10.0f)
        {
                strcpy(error_msg, "SoundFileInvalidTooLong");                   /*Flawfinder: ignore*/
                return(LLVORBISENC_CLIP_TOO_LONG);               
        }

    return(LLVORBISENC_NOERR);
}

S32 encode_vorbis_file(const char *in_fname, const char *out_fname)
{
        return(encode_vorbis_file_at(in_fname,out_fname, 128000));
}

S32 encode_vorbis_file_at(const char *in_fname, const char *out_fname, S32 bitrate)
{
#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;
        char error_msg[MAX_STRING];     /*Flawfinder: ignore*/
        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;

        apr_file_t* infp = ll_apr_file_open(in_fname,LL_APR_RB);
        if (!infp)
        {
                llwarns << "Couldn't open temporary ogg file for writing: " << in_fname
                        << llendl;
                return(LLVORBISENC_SOURCE_OPEN_ERR);
        }
        apr_file_t* outfp = ll_apr_file_open(out_fname,LL_APR_WPB);
        if (!outfp)
        {
                llwarns << "Couldn't open upload sound file for reading: " << in_fname
                        << llendl;
                apr_file_close (infp);
                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 (apr_file_eof(infp) != APR_EOF)
         {
                 ll_apr_file_seek(infp,APR_SET,file_pos);
                 ll_apr_file_read(infp, 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)))
                 {
                         ll_apr_file_seek(infp,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;
         } 
         
//       apr_file_close(infp);  

         /********** Encode setup ************/
         
         /* choose an encoding mode */
         /* (mode 0: 44kHz stereo uncoupled, roughly 128kbps VBR) */
         vorbis_info_init(&vi);

         // always encode to mono
//       vorbis_encode_init(&vi, /* num_channels */ 1 ,sample_rate, -1, bitrate, -1);
//       if (vorbis_encode_init(&vi, /* num_channels */ 1 ,sample_rate, -1, bitrate, -1))

//       F32 quality = 0;
//       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;
                         ll_apr_file_write(outfp, og.header, og.header_len);
                         ll_apr_file_write(outfp, og.body, og.body_len);
                 }
                 
         }
         
         
         while(!eos)
         {
                 long bytes_per_sample = bits_per_sample/8;

                 long bytes=(long)ll_apr_file_read(infp, 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;

                                 ll_apr_file_write(outfp, og.header, og.header_len);
                                 ll_apr_file_write(outfp, 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;
         apr_file_close(outfp);
         apr_file_close(infp);
         
#endif
         return(LLVORBISENC_NOERR);
         
}

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