コード例 #1
0
ファイル: twolame_encoder.c プロジェクト: GunioRobot/mpd
static bool
twolame_encoder_write(struct encoder *_encoder,
		      const void *data, size_t length,
		      G_GNUC_UNUSED GError **error)
{
	struct twolame_encoder *encoder = (struct twolame_encoder *)_encoder;
	unsigned num_frames;
	const int16_t *src = (const int16_t*)data;
	int bytes_out;

	assert(encoder->buffer_length == 0);

	num_frames =
		length / audio_format_frame_size(&encoder->audio_format);

	bytes_out = twolame_encode_buffer_interleaved(encoder->options,
						      src, num_frames,
						      encoder->buffer,
						      sizeof(encoder->buffer));
	if (bytes_out < 0) {
		g_set_error(error, twolame_encoder_quark(), 0,
			    "twolame encoder failed");
		return false;
	}

	encoder->buffer_length = (size_t)bytes_out;
	return true;
}
コード例 #2
0
ファイル: ae_twolame.c プロジェクト: BOTCrusher/sagetv
static int encode_twolame(audio_encoder_t *encoder, uint8_t *dest, void *src, int len, int max_size)
{
	mpae_twolame_ctx *ctx = (mpae_twolame_ctx *)encoder->priv;
	int ret_size = 0, r2;
	
	len /= (2*encoder->params.channels);
	ret_size = twolame_encode_buffer_interleaved(ctx->twolame_ctx, src, len, dest, max_size);
	r2 = mp_decode_mp3_header(dest);	
	mp_msg(MSGT_MENCODER, MSGL_DBG2, "\nSIZE: %d, max: %d, r2: %d\n", ret_size, max_size, r2);
	if(r2 > 0)
		ret_size = r2;
	return ret_size;
}
コード例 #3
0
ファイル: twolame.c プロジェクト: 12307/VLC-for-VS2010
static block_t *Encode( encoder_t *p_enc, block_t *p_aout_buf )
{
    encoder_sys_t *p_sys = p_enc->p_sys;

    /* FIXME:p_aout_buf is NULL when it's time to flush, does twolame has buffer to flush?*/
    if( unlikely( !p_aout_buf ) ) return NULL;

    int16_t *p_buffer = (int16_t *)p_aout_buf->p_buffer;
    int i_nb_samples = p_aout_buf->i_nb_samples;
    block_t *p_chain = NULL;

    p_sys->i_pts = p_aout_buf->i_pts -
                (mtime_t)1000000 * (mtime_t)p_sys->i_nb_samples /
                (mtime_t)p_enc->fmt_out.audio.i_rate;

    while ( p_sys->i_nb_samples + i_nb_samples >= MPEG_FRAME_SIZE )
    {
        int i_used;
        block_t *p_block;

        Bufferize( p_enc, p_buffer, MPEG_FRAME_SIZE - p_sys->i_nb_samples );
        i_nb_samples -= MPEG_FRAME_SIZE - p_sys->i_nb_samples;
        p_buffer += (MPEG_FRAME_SIZE - p_sys->i_nb_samples) * 2;

        i_used = twolame_encode_buffer_interleaved( p_sys->p_twolame,
                               p_sys->p_buffer, MPEG_FRAME_SIZE,
                               p_sys->p_out_buffer, MAX_CODED_FRAME_SIZE );
        p_sys->i_nb_samples = 0;
        p_block = block_Alloc( i_used );
        memcpy( p_block->p_buffer, p_sys->p_out_buffer, i_used );
        p_block->i_length = (mtime_t)1000000 *
                (mtime_t)MPEG_FRAME_SIZE / (mtime_t)p_enc->fmt_out.audio.i_rate;
        p_block->i_dts = p_block->i_pts = p_sys->i_pts;
        p_sys->i_pts += p_block->i_length;
        block_ChainAppend( &p_chain, p_block );
    }

    if ( i_nb_samples )
    {
        Bufferize( p_enc, p_buffer, i_nb_samples );
        p_sys->i_nb_samples += i_nb_samples;
    }

    return p_chain;
}
コード例 #4
0
ファイル: twolame.c プロジェクト: Kafay/vlc
static block_t *Encode( encoder_t *p_enc, aout_buffer_t *p_aout_buf )
{
    encoder_sys_t *p_sys = p_enc->p_sys;
    int16_t *p_buffer = (int16_t *)p_aout_buf->p_buffer;
    int i_nb_samples = p_aout_buf->i_nb_samples;
    block_t *p_chain = NULL;

    p_sys->i_pts = p_aout_buf->start_date -
                (mtime_t)1000000 * (mtime_t)p_sys->i_nb_samples /
                (mtime_t)p_enc->fmt_out.audio.i_rate;

    while ( p_sys->i_nb_samples + i_nb_samples >= MPEG_FRAME_SIZE )
    {
        int i_used;
        block_t *p_block;

        Bufferize( p_enc, p_buffer, MPEG_FRAME_SIZE - p_sys->i_nb_samples );
        i_nb_samples -= MPEG_FRAME_SIZE - p_sys->i_nb_samples;
        p_buffer += (MPEG_FRAME_SIZE - p_sys->i_nb_samples) * 2;

        i_used = twolame_encode_buffer_interleaved( p_sys->p_twolame,
                               p_sys->p_buffer, MPEG_FRAME_SIZE,
                               p_sys->p_out_buffer, MAX_CODED_FRAME_SIZE );
        p_sys->i_nb_samples = 0;
        p_block = block_New( p_enc, i_used );
        vlc_memcpy( p_block->p_buffer, p_sys->p_out_buffer, i_used );
        p_block->i_length = (mtime_t)1000000 *
                (mtime_t)MPEG_FRAME_SIZE / (mtime_t)p_enc->fmt_out.audio.i_rate;
        p_block->i_dts = p_block->i_pts = p_sys->i_pts;
        p_sys->i_pts += p_block->i_length;
        block_ChainAppend( &p_chain, p_block );
    }

    if ( i_nb_samples )
    {
        Bufferize( p_enc, p_buffer, i_nb_samples );
        p_sys->i_nb_samples += i_nb_samples;
    }

    return p_chain;
}
コード例 #5
0
/**
        \fn getPacket
*/
bool 	AUDMEncoder_Twolame::encode(uint8_t *dest, uint32_t *len, uint32_t *samples)
{
  int nbout;
  int channels=wavheader.channels;
  *samples = 1152; //FIXME
  *len = 0;
  ADM_assert(tmptail>=tmphead);
  if(!refillBuffer(_chunk ))
  {
    return false;
  }

  if(tmptail-tmphead<_chunk)
  {
    return false;
  }

  dither16(&(tmpbuffer[tmphead]),_chunk,channels);

  ADM_assert(tmptail>=tmphead);
  if (channels == 1)
  {
    nbout =twolame_encode_buffer(OPTIONS, (int16_t *)&(tmpbuffer[tmphead]),(int16_t *)&(tmpbuffer[tmphead]), _chunk, dest, 16 * 1024);
  }
  else
  {
    nbout = twolame_encode_buffer_interleaved(OPTIONS, (int16_t *)&(tmpbuffer[tmphead]), _chunk/2, dest, 16 * 1024);
  }
  tmphead+=_chunk;
  ADM_assert(tmptail>=tmphead);
  if (nbout < 0) {
    printf("[TwoLame] Error !!! : %d\n", nbout);
    return false;
  }
  *len=nbout;
  return true;
}
コード例 #6
0
uint8_t	AUDMEncoder_Twolame::getPacket(uint8_t *dest, uint32_t *len, uint32_t *samples)
{
  int nbout;
  
  *samples = 1152; //FIXME
  *len = 0;
  ADM_assert(tmptail>=tmphead);
  if(!refillBuffer(_chunk ))
  {
    return 0; 
  }
        
  if(tmptail-tmphead<_chunk)
  {
    return 0; 
  }

  dither16(&(tmpbuffer[tmphead]),_chunk,_wavheader->channels);

  ADM_assert(tmptail>=tmphead);
  if (_wavheader->channels == 1)
  {
    nbout =twolame_encode_buffer(OPTIONS, (int16_t *)&(tmpbuffer[tmphead]),(int16_t *)&(tmpbuffer[tmphead]), _chunk, dest, 16 * 1024);
  }
  else
  {
    nbout = twolame_encode_buffer_interleaved(OPTIONS, (int16_t *)&(tmpbuffer[tmphead]), _chunk/2, dest, 16 * 1024);
  }
  tmphead+=_chunk;
  ADM_assert(tmptail>=tmphead);
  if (nbout < 0) {
    printf("\n Error !!! : %ld\n", nbout);
    return 0;
  }
  *len=nbout;
  return 1;
}
コード例 #7
0
ファイル: ExportMP2.cpp プロジェクト: Avi2011class/audacity
int ExportMP2::Export(AudacityProject *project,
   int channels, const wxString &fName,
   bool selectionOnly, double t0, double t1, MixerSpec *mixerSpec, const Tags *metadata,
   int WXUNUSED(subformat))
{
   bool stereo = (channels == 2);
   long bitrate = gPrefs->Read(wxT("/FileFormats/MP2Bitrate"), 160);
   double rate = project->GetRate();
   const TrackList *tracks = project->GetTracks();

   wxLogNull logNo;             /* temporarily disable wxWidgets error messages */

   twolame_options *encodeOptions;
   encodeOptions = twolame_init();

   twolame_set_in_samplerate(encodeOptions, (int)(rate + 0.5));
   twolame_set_out_samplerate(encodeOptions, (int)(rate + 0.5));
   twolame_set_bitrate(encodeOptions, bitrate);
   twolame_set_num_channels(encodeOptions, stereo ? 2 : 1);

   if (twolame_init_params(encodeOptions) != 0)
   {
      wxMessageBox(_("Cannot export MP2 with this sample rate and bit rate"),
         _("Error"), wxICON_STOP);
      twolame_close(&encodeOptions);
      return false;
   }

   // Put ID3 tags at beginning of file
   if (metadata == NULL)
      metadata = project->GetTags();

   FileIO outFile(fName, FileIO::Output);
   if (!outFile.IsOpened()) {
      wxMessageBox(_("Unable to open target file for writing"));
      twolame_close(&encodeOptions);
      return false;
   }

   char *id3buffer = NULL;
   int id3len;
   bool endOfFile;
   id3len = AddTags(project, &id3buffer, &endOfFile, metadata);
   if (id3len && !endOfFile)
      outFile.Write(id3buffer, id3len);

   // Values taken from the twolame simple encoder sample
   const int pcmBufferSize = 9216 / 2; // number of samples
   const int mp2BufferSize = 16384; // bytes

   // We allocate a buffer which is twice as big as the
   // input buffer, which should always be enough.
   // We have to multiply by 4 because one sample is 2 bytes wide!
   unsigned char* mp2Buffer = new unsigned char[mp2BufferSize];

   const WaveTrackConstArray waveTracks =
      tracks->GetWaveTrackConstArray(selectionOnly, false);
   int updateResult = eProgressSuccess;
   {
      auto mixer = CreateMixer(waveTracks,
         tracks->GetTimeTrack(),
         t0, t1,
         stereo ? 2 : 1, pcmBufferSize, true,
         rate, int16Sample, true, mixerSpec);

      ProgressDialog progress(wxFileName(fName).GetName(),
         selectionOnly ?
         wxString::Format(_("Exporting selected audio at %ld kbps"), bitrate) :
         wxString::Format(_("Exporting entire file at %ld kbps"), bitrate));

      while (updateResult == eProgressSuccess) {
         sampleCount pcmNumSamples = mixer->Process(pcmBufferSize);

         if (pcmNumSamples == 0)
            break;

         short *pcmBuffer = (short *)mixer->GetBuffer();

         int mp2BufferNumBytes = twolame_encode_buffer_interleaved(
            encodeOptions,
            pcmBuffer,
            pcmNumSamples,
            mp2Buffer,
            mp2BufferSize);

         outFile.Write(mp2Buffer, mp2BufferNumBytes);

         updateResult = progress.Update(mixer->MixGetCurrentTime() - t0, t1 - t0);
      }
   }

   int mp2BufferNumBytes = twolame_encode_flush(
      encodeOptions,
      mp2Buffer,
      mp2BufferSize);

   if (mp2BufferNumBytes > 0)
      outFile.Write(mp2Buffer, mp2BufferNumBytes);

   twolame_close(&encodeOptions);

   delete[] mp2Buffer;

   /* Write ID3 tag if it was supposed to be at the end of the file */

   if (id3len && endOfFile)
      outFile.Write(id3buffer, id3len);

   if (id3buffer) {
   free(id3buffer);
   }

   /* Close file */

   outFile.Close();

   return updateResult;
}
コード例 #8
0
ファイル: gsttwolame.c プロジェクト: jwzl/ossbuild
static GstFlowReturn
gst_two_lame_chain (GstPad * pad, GstBuffer * buf)
{
    GstTwoLame *twolame;
    guchar *mp3_data;
    gint mp3_buffer_size, mp3_size;
    gint64 duration;
    GstFlowReturn result;
    gint num_samples;
    guint8 *data;
    guint size;

    twolame = GST_TWO_LAME (GST_PAD_PARENT (pad));

    GST_LOG_OBJECT (twolame, "entered chain");

    if (!twolame->setup)
        goto not_setup;

    data = GST_BUFFER_DATA (buf);
    size = GST_BUFFER_SIZE (buf);

    if (twolame->float_input)
        num_samples = size / 4;
    else
        num_samples = size / 2;

    /* allocate space for output */
    mp3_buffer_size = 1.25 * num_samples + 16384;
    mp3_data = g_malloc (mp3_buffer_size);

    if (twolame->num_channels == 1) {
        if (twolame->float_input)
            mp3_size = twolame_encode_buffer_float32 (twolame->glopts,
                       (float *) data,
                       (float *) data, num_samples, mp3_data, mp3_buffer_size);
        else
            mp3_size = twolame_encode_buffer (twolame->glopts,
                                              (short int *) data,
                                              (short int *) data, num_samples, mp3_data, mp3_buffer_size);
    } else {
        if (twolame->float_input)
            mp3_size = twolame_encode_buffer_float32_interleaved (twolame->glopts,
                       (float *) data,
                       num_samples / twolame->num_channels, mp3_data, mp3_buffer_size);
        else
            mp3_size = twolame_encode_buffer_interleaved (twolame->glopts,
                       (short int *) data,
                       num_samples / twolame->num_channels, mp3_data, mp3_buffer_size);
    }

    GST_LOG_OBJECT (twolame, "encoded %d bytes of audio to %d bytes of mp3",
                    size, mp3_size);

    if (twolame->float_input)
        duration = gst_util_uint64_scale_int (size, GST_SECOND,
                                              4 * twolame->samplerate * twolame->num_channels);
    else
        duration = gst_util_uint64_scale_int (size, GST_SECOND,
                                              2 * twolame->samplerate * twolame->num_channels);

    if (GST_BUFFER_DURATION (buf) != GST_CLOCK_TIME_NONE &&
            GST_BUFFER_DURATION (buf) != duration) {
        GST_DEBUG_OBJECT (twolame, "incoming buffer had incorrect duration %"
                          GST_TIME_FORMAT ", outgoing buffer will have correct duration %"
                          GST_TIME_FORMAT,
                          GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_TIME_ARGS (duration));
    }

    if (twolame->last_ts == GST_CLOCK_TIME_NONE) {
        twolame->last_ts = GST_BUFFER_TIMESTAMP (buf);
        twolame->last_offs = GST_BUFFER_OFFSET (buf);
        twolame->last_duration = duration;
    } else {
        twolame->last_duration += duration;
    }

    gst_buffer_unref (buf);

    if (mp3_size < 0) {
        g_warning ("error %d", mp3_size);
    }

    if (mp3_size > 0) {
        GstBuffer *outbuf;

        outbuf = gst_buffer_new ();
        GST_BUFFER_DATA (outbuf) = mp3_data;
        GST_BUFFER_MALLOCDATA (outbuf) = mp3_data;
        GST_BUFFER_SIZE (outbuf) = mp3_size;
        GST_BUFFER_TIMESTAMP (outbuf) = twolame->last_ts;
        GST_BUFFER_OFFSET (outbuf) = twolame->last_offs;
        GST_BUFFER_DURATION (outbuf) = twolame->last_duration;
        gst_buffer_set_caps (outbuf, GST_PAD_CAPS (twolame->srcpad));

        result = gst_pad_push (twolame->srcpad, outbuf);
        twolame->last_flow = result;
        if (result != GST_FLOW_OK) {
            GST_DEBUG_OBJECT (twolame, "flow return: %s", gst_flow_get_name (result));
        }

        if (GST_CLOCK_TIME_IS_VALID (twolame->last_ts))
            twolame->eos_ts = twolame->last_ts + twolame->last_duration;
        else
            twolame->eos_ts = GST_CLOCK_TIME_NONE;
        twolame->last_ts = GST_CLOCK_TIME_NONE;
    } else {
        g_free (mp3_data);
        result = GST_FLOW_OK;
    }

    return result;

    /* ERRORS */
not_setup:
    {
        gst_buffer_unref (buf);
        GST_ELEMENT_ERROR (twolame, CORE, NEGOTIATION, (NULL),
                           ("encoder not initialized (input is not audio?)"));
        return GST_FLOW_ERROR;
    }
}
コード例 #9
0
ファイル: frontend.c プロジェクト: ruthmagnus/audacity
int
main(int argc, char **argv)
{
    twolame_options *encopts = NULL;
    SNDFILE         *inputfile = NULL;
    FILE            *outputfile = NULL;
    short int       *pcmaudio = NULL;
    int             samples_read = 0;
    unsigned int    frame_count = 0;
    unsigned char   *mp2buffer = NULL;
    int             mp2fill_size = 0;
    int             audioReadSize = 0;


    // Allocate memory for the PCM audio data
    if ((pcmaudio = (short int *) calloc(AUDIOBUFSIZE, sizeof(short int))) == NULL) {
        fprintf(stderr, "Error: pcmaudio memory allocation failed\n");
        exit(ERR_MEM_ALLOC);
    }
    
    // Allocate memory for the encoded MP2 audio data
    if ((mp2buffer = (unsigned char *) calloc(MP2BUFSIZE, sizeof(unsigned char))) == NULL) {
        fprintf(stderr, "Error: mp2buffer memory allocation failed\n");
        exit(ERR_MEM_ALLOC);
    }
    
    // Initialise Encoder Options Structure 
    encopts = twolame_init();
    if (encopts == NULL) {
        fprintf(stderr, "Error: initializing libtwolame encoder failed.\n");
        exit(ERR_MEM_ALLOC);
    }

    
    // Get options and parameters from the command line
    parse_args(argc, argv, encopts);


    // Open the input file
    inputfile = open_input_file( inputfilename );
    twolame_set_num_channels( encopts, sfinfo.channels );
    twolame_set_in_samplerate( encopts, sfinfo.samplerate );
        
    // Open the output file
    outputfile = open_output_file( outputfilename );
    
    // display file settings
	print_file_config( inputfile, twolame_get_verbosity(encopts) );

    // initialise twolame with this set of options
    if (twolame_init_params( encopts ) != 0) {
        fprintf(stderr, "Error: configuring libtwolame encoder failed.\n");
        exit(ERR_INVALID_PARAM);
    }

    // display encoder settings
	twolame_print_config( encopts );


	// Only encode a single frame of mpeg audio ?
    if (single_frame_mode) audioReadSize = 1152;
    else audioReadSize = AUDIOBUFSIZE;


    // Now do the reading/encoding/writing
    while ((samples_read = sf_read_short( inputfile, pcmaudio, audioReadSize )) > 0) {
        int bytes_out = 0;
        
        // Force byte swapping if requested
        if (byteswap) {
			int i;
			for (i = 0; i<samples_read; i++) {
				short tmp = pcmaudio[i];
				char *src = (char*)&tmp;
				char *dst = (char*)&pcmaudio[i];
				dst[0] = src[1];
				dst[1] = src[0];
			}
        }

		// Calculate the number of samples we have (per channel)
        samples_read /= sfinfo.channels;

        // Do swapping of left and right channels if requested
        if (channelswap && sfinfo.channels == 2) {
        	int i;
        	for(i=0; i<samples_read; i++) {
        		short tmp = pcmaudio[(2*i)];
        		pcmaudio[(2*i)] = pcmaudio[(2*i)+1];
        		pcmaudio[(2*i)+1] = tmp;
        	}
        }
        
        // Encode the audio to MP2
        mp2fill_size = twolame_encode_buffer_interleaved( encopts, pcmaudio, samples_read, mp2buffer, MP2BUFSIZE);
        
        // Stop if we don't have any bytes (probably don't have enough audio for a full frame of mpeg audio)
        if (mp2fill_size==0) break;
        if (mp2fill_size<0) {
            fprintf(stderr,"error while encoding audio: %d\n", mp2fill_size);
            exit(ERR_WRITING_OUTPUT);
        }

        // Write the encoded audio out
        bytes_out = fwrite(mp2buffer, sizeof(unsigned char), mp2fill_size, outputfile);
        if (bytes_out<=0) {
            perror("error while writing to output file");
            exit(ERR_WRITING_OUTPUT);
        }
        
        // Only single frame ?
        if (single_frame_mode) break;
        
        // Display Progress
        frame_count += (samples_read / 1152);
        if (twolame_get_verbosity(encopts)>0) {
        	fprintf(stderr, "[%04i", frame_count);
        	//fprintf(stderr, "/%04i", (int)(sfinfo.frames / sfinfo.channels  / 1152));
        	fprintf(stderr, "]\r");
        	fflush(stderr);
        }
    }

    //
    // flush any remaining audio. (don't send any new audio data) There
    // should only ever be a max of 1 frame on a flush. There may be zero
    // frames if the audio data was an exact multiple of 1152
    //
    mp2fill_size = twolame_encode_flush( encopts, mp2buffer, MP2BUFSIZE);
    if (mp2fill_size>0) {
        int bytes_out = fwrite(mp2buffer, sizeof(unsigned char), mp2fill_size, outputfile);
        if (bytes_out<=0) {
            perror("error while writing to output file");
            exit(ERR_WRITING_OUTPUT);
        }
    }
    
 	if (twolame_get_verbosity(encopts)>1) {
	    fprintf(stderr, "\nEncoding Finished.\n");
	}

    // Close input and output files
    sf_close( inputfile );
    fclose( outputfile );

	// Close the libtwolame encoder
    twolame_close(&encopts);
    
    
	// Free up memory
	free(pcmaudio);
	free(mp2buffer);


	
    return (ERR_NO_ERROR);
}
コード例 #10
0
ファイル: twolame.c プロジェクト: mstorsjo/vlc
static block_t *Encode( encoder_t *p_enc, block_t *p_aout_buf )
{
    encoder_sys_t *p_sys = p_enc->p_sys;
    block_t *p_chain = NULL;

    if( unlikely( !p_aout_buf ) ) {
        int i_used = 0;
        block_t *p_block;

        i_used = twolame_encode_flush( p_sys->p_twolame,
                                p_sys->p_out_buffer, MAX_CODED_FRAME_SIZE );
        if( i_used <= 0 )
            return NULL;

        p_block = block_Alloc( i_used );
        if( !p_block )
            return NULL;
        memcpy( p_block->p_buffer, p_sys->p_out_buffer, i_used );
        p_block->i_length = vlc_tick_from_samples( MPEG_FRAME_SIZE,
                                                   p_enc->fmt_out.audio.i_rate );
        p_block->i_dts = p_block->i_pts = p_sys->i_pts;
        p_sys->i_pts += p_block->i_length;

        return p_block;
    }

    int16_t *p_buffer = (int16_t *)p_aout_buf->p_buffer;
    int i_nb_samples = p_aout_buf->i_nb_samples;

    p_sys->i_pts = p_aout_buf->i_pts -
                vlc_tick_from_samples( p_sys->i_nb_samples,
                                       p_enc->fmt_out.audio.i_rate );

    while ( p_sys->i_nb_samples + i_nb_samples >= MPEG_FRAME_SIZE )
    {
        int i_used;
        block_t *p_block;

        Bufferize( p_enc, p_buffer, MPEG_FRAME_SIZE - p_sys->i_nb_samples );
        i_nb_samples -= MPEG_FRAME_SIZE - p_sys->i_nb_samples;
        p_buffer += (MPEG_FRAME_SIZE - p_sys->i_nb_samples) * 2;

        i_used = twolame_encode_buffer_interleaved( p_sys->p_twolame,
                               p_sys->p_buffer, MPEG_FRAME_SIZE,
                               p_sys->p_out_buffer, MAX_CODED_FRAME_SIZE );
        /* On error, buffer samples and return what was already encoded */
        if( i_used < 0 )
        {
            msg_Err( p_enc, "encoder error: %d", i_used );
            break;
        }

        p_sys->i_nb_samples = 0;
        p_block = block_Alloc( i_used );
        if( !p_block )
        {
            if( p_chain )
                block_ChainRelease( p_chain );
            return NULL;
        }
        memcpy( p_block->p_buffer, p_sys->p_out_buffer, i_used );
        p_block->i_length = vlc_tick_from_samples( MPEG_FRAME_SIZE,
                                                   p_enc->fmt_out.audio.i_rate );
        p_block->i_dts = p_block->i_pts = p_sys->i_pts;
        p_sys->i_pts += p_block->i_length;
        block_ChainAppend( &p_chain, p_block );
    }

    if ( i_nb_samples )
    {
        Bufferize( p_enc, p_buffer, i_nb_samples );
        p_sys->i_nb_samples += i_nb_samples;
    }

    return p_chain;
}
コード例 #11
0
static GstFlowReturn
gst_two_lame_handle_frame (GstAudioEncoder * enc, GstBuffer * buf)
{
  GstTwoLame *twolame;
  gint mp3_buffer_size, mp3_size;
  GstBuffer *mp3_buf;
  GstFlowReturn result;
  gint num_samples;
  GstMapInfo map, mp3_map;

  twolame = GST_TWO_LAME (enc);

  /* squeeze remaining and push */
  if (G_UNLIKELY (buf == NULL))
    return gst_two_lame_flush_full (twolame, TRUE);

  gst_buffer_map (buf, &map, GST_MAP_READ);

  if (twolame->float_input)
    num_samples = map.size / 4;
  else
    num_samples = map.size / 2;

  /* allocate space for output */
  mp3_buffer_size = 1.25 * num_samples + 16384;
  mp3_buf = gst_buffer_new_and_alloc (mp3_buffer_size);
  gst_buffer_map (mp3_buf, &mp3_map, GST_MAP_WRITE);

  if (twolame->num_channels == 1) {
    if (twolame->float_input)
      mp3_size = twolame_encode_buffer_float32 (twolame->glopts,
          (float *) map.data,
          (float *) map.data, num_samples, mp3_map.data, mp3_buffer_size);
    else
      mp3_size = twolame_encode_buffer (twolame->glopts,
          (short int *) map.data,
          (short int *) map.data, num_samples, mp3_map.data, mp3_buffer_size);
  } else {
    if (twolame->float_input)
      mp3_size = twolame_encode_buffer_float32_interleaved (twolame->glopts,
          (float *) map.data,
          num_samples / twolame->num_channels, mp3_map.data, mp3_buffer_size);
    else
      mp3_size = twolame_encode_buffer_interleaved (twolame->glopts,
          (short int *) map.data,
          num_samples / twolame->num_channels, mp3_map.data, mp3_buffer_size);
  }

  GST_LOG_OBJECT (twolame, "encoded %" G_GSIZE_FORMAT " bytes of audio "
      "to %d bytes of mp3", map.size, mp3_size);

  gst_buffer_unmap (buf, &map);
  gst_buffer_unmap (mp3_buf, &mp3_map);

  if (mp3_size > 0) {
    gst_buffer_set_size (mp3_buf, mp3_size);
    result = gst_audio_encoder_finish_frame (enc, mp3_buf, -1);
  } else {
    if (mp3_size < 0) {
      /* eat error ? */
      g_warning ("error %d", mp3_size);
    }
    gst_buffer_unref (mp3_buf);
    result = GST_FLOW_OK;
  }

  return result;
}
コード例 #12
0
int main(int argc, char **argv)
{
    twolame_options *encodeOptions;
    char *inputfilename = argv[1];
    char *outputfilename = argv[2];
    FILE *outfile, *fpSrc = NULL;
    short int *pcmaudio;
    unsigned char *mp2buffer;
    int num_samples = 0;
    int mp2fill_size = 0;
    int frames = 0;
    wave_info_t *wave_info = NULL;
		int nReadSize = 0;

    if (argc != 3)
        usage();


    /* Allocate some space for the PCM audio data */
    if ((pcmaudio = (short *) calloc(AUDIOBUFSIZE, sizeof(short))) == NULL) {
        fprintf(stderr, "pcmaudio alloc failed\n");
        exit(99);
    }

    /* Allocate some space for the encoded MP2 audio data */
    if ((mp2buffer = (unsigned char *) calloc(MP2BUFSIZE, sizeof(unsigned char))) == NULL) {
        fprintf(stderr, "mp2buffer alloc failed\n");
        exit(99);
    }




    /* grab a set of default encode options */
    encodeOptions = twolame_init();
    printf("Using libtwolame version %s.\n", get_twolame_version());


    /* Open the wave file */
    if ((wave_info = wave_init(inputfilename)) == NULL) {
        fprintf(stderr, "Not a recognised WAV file.\n");
        exit(99);
    }
    // Use sound file to over-ride preferences for
    // mono/stereo and sampling-frequency
    twolame_set_num_channels(encodeOptions, wave_info->channels);
    if (wave_info->channels == 1) {
        twolame_set_mode(encodeOptions, TWOLAME_MONO);
    } else {
        twolame_set_mode(encodeOptions, TWOLAME_STEREO);
    }


    /* Set the input and output sample rate to the same */
    twolame_set_in_samplerate(encodeOptions, wave_info->samplerate);
    twolame_set_out_samplerate(encodeOptions, wave_info->samplerate);

    /* Set the bitrate to 192 kbps */
    twolame_set_bitrate(encodeOptions, 64);

    /* initialise twolame with this set of options */
    if (twolame_init_params(encodeOptions) != 0) {
        fprintf(stderr, "Error: configuring libtwolame encoder failed.\n");
        exit(99);
    }

    /* Open the output file for the encoded MP2 data */
    if ((outfile = fopen(outputfilename, "wb")) == 0) {
        fprintf(stderr, "error opening output file %s\n", outputfilename);
        exit(99);
    }

    // Read num_samples of audio data *per channel* from the input file
    //while ((num_samples = wave_get_samples(wave_info, pcmaudio, AUDIOBUFSIZE/2)) != 0) {
	



	fpSrc = fopen( "a2002011001-e02.wav", "rb" );
	while(1)
	{
		num_samples = fread( pcmaudio, sizeof( char ), AUDIOBUFSIZE/2, fpSrc );
        // Encode the audio!2024*4

		if( num_samples != AUDIOBUFSIZE/2 )
			break;

		num_samples /= 2;

		mp2fill_size =
            twolame_encode_buffer_interleaved(encodeOptions, pcmaudio, num_samples/2, mp2buffer,
                                              MP2BUFSIZE);

        // Write the MPEG bitstream to the file
        fwrite(mp2buffer, sizeof(unsigned char), mp2fill_size, outfile);

        // Display the number of MPEG audio frames we have encoded
        frames += (num_samples / TWOLAME_SAMPLES_PER_FRAME);
        printf("[%04i]\r", frames);
        fflush(stdout);
    }

    /* flush any remaining audio. (don't send any new audio data) There should only ever be a max
       of 1 frame on a flush. There may be zero frames if the audio data was an exact multiple of
       1152 */
    mp2fill_size = twolame_encode_flush(encodeOptions, mp2buffer, MP2BUFSIZE);
    fwrite(mp2buffer, sizeof(unsigned char), mp2fill_size, outfile);


    twolame_close(&encodeOptions);
    free(pcmaudio);

    printf("\nFinished nicely.\n");


    return (0);
}
コード例 #13
0
int main(int argc, char **argv)
{
    twolame_options *encopts = NULL;
    audioin_t *inputfile = NULL;
    FILE *outputfile = NULL;
    short int *pcmaudio = NULL;
    unsigned int frame_count = 0;
    unsigned int total_frames = 0;
    unsigned int frame_len = 0;
    unsigned int total_bytes = 0;
    unsigned char *mp2buffer = NULL;
    int samples_read = 0;
    int mp2fill_size = 0;
    int audioReadSize = 0;


    // Allocate memory for the PCM audio data
    if ((pcmaudio = (short int *) calloc(AUDIO_BUF_SIZE, sizeof(short int))) == NULL) {
        fprintf(stderr, "Error: pcmaudio memory allocation failed\n");
        exit(ERR_MEM_ALLOC);
    }
    // Allocate memory for the encoded MP2 audio data
    if ((mp2buffer = (unsigned char *) calloc(MP2_BUF_SIZE, sizeof(unsigned char))) == NULL) {
        fprintf(stderr, "Error: mp2buffer memory allocation failed\n");
        exit(ERR_MEM_ALLOC);
    }
    // Initialise Encoder Options Structure
    encopts = twolame_init();
    if (encopts == NULL) {
        fprintf(stderr, "Error: initializing libtwolame encoder failed.\n");
        exit(ERR_MEM_ALLOC);
    }
    // Get options and parameters from the command line
    parse_args(argc, argv, encopts);

    // Display the filenames
    print_filenames(twolame_get_verbosity(encopts));

    // Open the input file
    if (use_raw) {
        // use raw input handler
        inputfile = open_audioin_raw(inputfilename, &sfinfo, sample_size);
    } else {
        // use libsndfile
        inputfile = open_audioin_sndfile(inputfilename, &sfinfo);
    }

    // Display input information
    if (twolame_get_verbosity(encopts) > 1) {
        inputfile->print_info(inputfile);
    }
    // Use information from input file to configure libtwolame
    twolame_set_num_channels(encopts, sfinfo.channels);
    twolame_set_in_samplerate(encopts, sfinfo.samplerate);


    // Open the output file
    outputfile = open_output_file(outputfilename);

    // initialise twolame with this set of options
    if (twolame_init_params(encopts) != 0) {
        fprintf(stderr, "Error: configuring libtwolame encoder failed.\n");
        exit(ERR_INVALID_PARAM);
    }
    // display encoder settings
    twolame_print_config(encopts);


    // Only encode a single frame of mpeg audio ?
    if (single_frame_mode)
        audioReadSize = TWOLAME_SAMPLES_PER_FRAME;
    else
        audioReadSize = AUDIO_BUF_SIZE;

    // Calculate the size and number of frames we are going to encode
    frame_len = twolame_get_framelength(encopts);
    if (sfinfo.frames)
        total_frames = sfinfo.frames / TWOLAME_SAMPLES_PER_FRAME;


    // Now do the reading/encoding/writing
    while ((samples_read = inputfile->read(inputfile, pcmaudio, audioReadSize)) > 0) {
        int bytes_out = 0;

        // Force byte swapping if requested
        if (byteswap) {
            int i;
            for (i = 0; i < samples_read; i++) {
                short tmp = pcmaudio[i];
                char *src = (char *) &tmp;
                char *dst = (char *) &pcmaudio[i];
                dst[0] = src[1];
                dst[1] = src[0];
            }
        }
        // Calculate the number of samples we have (per channel)
        samples_read /= sfinfo.channels;

        // Do swapping of left and right channels if requested
        if (channelswap && sfinfo.channels == 2) {
            int i;
            for (i = 0; i < samples_read; i++) {
                short tmp = pcmaudio[(2 * i)];
                pcmaudio[(2 * i)] = pcmaudio[(2 * i) + 1];
                pcmaudio[(2 * i) + 1] = tmp;
            }
        }
        // Encode the audio to MP2
        mp2fill_size =
            twolame_encode_buffer_interleaved(encopts, pcmaudio, samples_read, mp2buffer,
                                              MP2_BUF_SIZE);

        // Stop if we don't have any bytes (probably don't have enough audio for a full frame of
        // mpeg audio)
        if (mp2fill_size == 0)
            break;
        if (mp2fill_size < 0) {
            fprintf(stderr, "error while encoding audio: %d\n", mp2fill_size);
            exit(ERR_ENCODING);
        }
        // Check that a whole number of frame was written
        // if (mp2fill_size % frame_len != 0) {
        // fprintf(stderr,"error while encoding audio: non-whole number of frames written\n");
        // exit(ERR_ENCODING);
        // }

        // Write the encoded audio out
        bytes_out = fwrite(mp2buffer, sizeof(unsigned char), mp2fill_size, outputfile);
        if (bytes_out != mp2fill_size) {
            perror("error while writing to output file");
            exit(ERR_WRITING_OUTPUT);
        }
        total_bytes += bytes_out;

        // Only single frame ?
        if (single_frame_mode)
            break;


        // Display Progress
        frame_count += (mp2fill_size / frame_len);
        if (twolame_get_verbosity(encopts) > 0) {
            fprintf(stderr, "\rEncoding frame: %i", frame_count);
            if (total_frames) {
                fprintf(stderr, "/%i (%i%%)", total_frames, (frame_count * 100) / total_frames);
            }
            fflush(stderr);
        }
    }

    // Was there an error reading the audio?
    if (inputfile->error_str(inputfile)) {
        fprintf(stderr, "Error reading from input file: %s\n", inputfile->error_str(inputfile));
    }
    //
    // flush any remaining audio. (don't send any new audio data) There
    // should only ever be a max of 1 frame on a flush. There may be zero
    // frames if the audio data was an exact multiple of 1152
    //
    mp2fill_size = twolame_encode_flush(encopts, mp2buffer, MP2_BUF_SIZE);
    if (mp2fill_size > 0) {
        int bytes_out = fwrite(mp2buffer, sizeof(unsigned char), mp2fill_size, outputfile);
        frame_count++;
        if (bytes_out <= 0) {
            perror("error while writing to output file");
            exit(ERR_WRITING_OUTPUT);
        }
        total_bytes += bytes_out;
    }

    if (twolame_get_verbosity(encopts) > 1) {
        char *filesize = format_filesize_string(total_bytes);
        fprintf(stderr, "\nEncoding Finished.\n");
        fprintf(stderr, "Total bytes written: %s.\n", filesize);
        free(filesize);
    }
    // Close input and output streams
    inputfile->close(inputfile);
    fclose(outputfile);

    // Close the libtwolame encoder
    twolame_close(&encopts);


    // Free up memory
    free(pcmaudio);
    free(mp2buffer);

    return (ERR_NO_ERROR);
}