예제 #1
0
int enclameInit( hb_work_object_t * w, hb_job_t * job )
{
    hb_work_private_t * pv = calloc( 1, sizeof( hb_work_private_t ) );
    hb_audio_t * audio = w->audio;

    w->private_data = pv;

    pv->job   = job;

    hb_log( "enclame: opening libmp3lame" );

    pv->lame = lame_init();
    // use ABR
    lame_set_scale( pv->lame, 32768.0 );
    if( audio->config.out.compression_level >= 0 )
    {
        lame_set_quality( pv->lame, audio->config.out.compression_level );
    }
    if( audio->config.out.bitrate > 0 )
    {
        lame_set_VBR( pv->lame, vbr_abr );
        lame_set_VBR_mean_bitrate_kbps( pv->lame, audio->config.out.bitrate );
    }
    else if( audio->config.out.quality >= 0 )
    {
        lame_set_brate( pv->lame, 0 );
        lame_set_VBR( pv->lame, vbr_default );
        lame_set_VBR_quality( pv->lame, audio->config.out.quality );
    }
    lame_set_in_samplerate( pv->lame, audio->config.out.samplerate );
    lame_set_out_samplerate( pv->lame, audio->config.out.samplerate );

    pv->out_discrete_channels = hb_mixdown_get_discrete_channel_count( audio->config.out.mixdown );
    // Lame's default encoding mode is JOINT_STEREO.  This subtracts signal
    // that is "common" to left and right (within some threshold) and encodes
    // it separately.  This improves quality at low bitrates, but hurts 
    // imaging (channel separation) at higher bitrates.  So if the bitrate
    // is suffeciently high, use regular STEREO mode.
    if ( pv->out_discrete_channels == 1 )
    {
        lame_set_mode( pv->lame, MONO );
        lame_set_num_channels( pv->lame, 1 );
    }
    else if ( audio->config.out.bitrate >= 128 )
    {
        lame_set_mode( pv->lame, STEREO );
    }
    lame_init_params( pv->lame );

    pv->input_samples = 1152 * pv->out_discrete_channels;
    pv->output_bytes = LAME_MAXMP3BUFFER;
    pv->buf  = malloc( pv->input_samples * sizeof( float ) );
    audio->config.out.samples_per_frame = 1152;

    pv->list = hb_list_init();
    pv->pts  = -1;

    return 0;
}
void AudioReader::exportAudioDataToMP3(const char* data)
{
    QSettings settings;
    emit TCMessage("Exporting MP3...");
    int read, write;
    FILE *pcm = fopen("TCTemp.wav", "wb");
    fwrite(data,1,format.chunkSize,pcm);
    fflush(pcm);
    pcm = fopen("TCTemp.wav", "rb");
    f = fopen(this->outfile.toLocal8Bit().constData(), "wb");

    const int PCM_SIZE = 8192;
    const int MP3_SIZE = 8192*2;

    short int pcm_buffer[PCM_SIZE*2];
    unsigned char mp3_buffer[MP3_SIZE];

    lame_t lame = lame_init();
    lame_set_in_samplerate(lame, format.SampleRate);
    lame_set_VBR(lame, (vbr_mode)settings.value("vbr",0).toInt());

    if(lame_get_VBR(lame)) {
        lame_set_VBR_min_bitrate_kbps(lame,settings.value("min",128).toInt());
        lame_set_VBR_max_bitrate_kbps(lame,settings.value("max",192).toInt());

    }
    else
    {
        lame_set_brate(lame,settings.value("min",128).toInt());
    }
    lame_set_num_channels(lame,(int)format.Channels);
    lame_set_quality(lame,0);
    if(format.Channels==1)lame_set_mode(lame,MONO);
    else                  lame_set_mode(lame,STEREO);

    lame_init_params(lame);
    do {
        if(!knock->running)break;
        read = fread(pcm_buffer, 2*byte_size, PCM_SIZE, pcm);
        fflush(f);
        if (read == 0)
            write = lame_encode_flush(lame, mp3_buffer, MP3_SIZE);
        else
            write = lame_encode_buffer_interleaved(lame, pcm_buffer, read, mp3_buffer, MP3_SIZE);
        fwrite(mp3_buffer, write, 1, f);
        fflush(f);
    } while (read != 0);
    lame_close(lame);
    fclose(pcm);
    if(unlink("TCTemp.wav")!=0) {
        emit TCColor("color: red;");
        emit TCMessage(QString("Error while removing TCTemp.wav"));
    }
}
예제 #3
0
int doInit(LameConf* conf) {
	int result;

	conf->gf=lame_init();
	if (conf->gf==NULL) {
		//throwRuntimeException(env, "out of memory");
		return org_tritonus_lowlevel_lame_Lame_OUT_OF_MEMORY;
	}

	lame_set_num_channels(conf->gf, conf->channels);
	lame_set_in_samplerate(conf->gf, conf->sampleRate);
	if (conf->mode!=org_tritonus_lowlevel_lame_Lame_CHANNEL_MODE_AUTO) {
		lame_set_mode(conf->gf, conf->mode);
	}
	if (conf->VBR) {
		lame_set_VBR(conf->gf, vbr_default);
		lame_set_VBR_q(conf->gf, conf->quality);
	} else {
		if (conf->bitrate!=org_tritonus_lowlevel_lame_Lame_BITRATE_AUTO) {
		    lame_set_brate(conf->gf, conf->bitrate);
		}
	}
	lame_set_quality(conf->gf, conf->quality);
	result=lame_init_params(conf->gf);

	// return effective values
	conf->sampleRate=lame_get_out_samplerate(conf->gf);
	conf->bitrate=lame_get_brate(conf->gf);
	conf->mode=lame_get_mode(conf->gf);
	conf->VBR=lame_get_VBR(conf->gf);
	conf->quality=(conf->VBR)?lame_get_VBR_q(conf->gf):lame_get_quality(conf->gf);
	conf->mpegVersion=lame_get_version(conf->gf);

	return result;
}
예제 #4
0
    MP3Encoder(UINT bitRate)
    {
        curBitRate = bitRate;

        lgf = lame_init();
        if(!lgf)
            CrashError(TEXT("Unable to open mp3 encoder"));

        lame_set_in_samplerate(lgf, App->GetSampleRateHz());
        lame_set_out_samplerate(lgf, App->GetSampleRateHz());
        lame_set_num_channels(lgf, 2);
        lame_set_mode(lgf, STEREO);
        lame_set_disable_reservoir(lgf, TRUE); //bit reservoir has to be disabled for seamless streaming
        lame_set_VBR(lgf, vbr_off);
        lame_set_brate(lgf, bitRate);
        lame_init_params(lgf);

        outputFrameSize = lame_get_framesize(lgf); //1152 usually
        dwMP3MaxSize = DWORD(1.25*double(outputFrameSize*audioBlockSize) + 7200.0);
        MP3OutputBuffer.SetSize(dwMP3MaxSize+1);
        MP3OutputBuffer[0] = 0x2f;

        bFirstPacket = true;

        Log(TEXT("------------------------------------------"));
        Log(TEXT("%s"), GetInfoString().Array());
    }
예제 #5
0
파일: main.c 프로젝트: alex-grigoras/pLAME
int main(int argc, char** argv)
{
	if(argc != 2)
	{
		printf("Usage: %s <wav file>\n", argv[0]);
		exit(0);
	}

	lame_global_flags *gfp;
	
	
	//set params
	gfp = lame_init();
	
	lame_set_num_channels(gfp,2);
	lame_set_in_samplerate(gfp,44100);
	lame_set_brate(gfp,128);
	lame_set_mode(gfp,1);
	lame_set_quality(gfp,2); 
	
	int ret_code = lame_init_params(gfp);
	if(ret_code < 0)
	{
		printf("ret_code < 0\n");
		exit(-1);
	}
	
	//read input file
	int read, write;
	FILE* pcm = fopen(argv[1], "rb");
	
	char* outPath;
	outPath = strdup(argv[1]);
	sprintf(outPath+strlen(argv[1])-3, "mp3");
	
	FILE* mp3 = fopen(outPath, "wb");
	

	
	short int pcm_buffer[PCM_SIZE * 2];
	unsigned char mp3_buffer[MP3_SIZE];
	
	do
	{
		read = fread(pcm_buffer, 2*sizeof(short int), PCM_SIZE, pcm);
		if(read == 0)
			write = lame_encode_flush(gfp, mp3_buffer, MP3_SIZE);
		else
			write = lame_encode_buffer_interleaved(gfp, pcm_buffer, read, mp3_buffer, MP3_SIZE);
		fwrite(mp3_buffer, write, sizeof (unsigned char), mp3);
	}
	while (read != 0);
	
	lame_close(gfp);
	fclose(mp3);
	fclose(pcm);
	
	exit(0);
}
예제 #6
0
bool LameAudioWriter::open_private()
{
	m_fid = fopen(m_fileName.toUtf8().data(), "w+");
	if (!m_fid) {
		return false;
	}
	
	m_lameInfo->flags = lame_init();
	
	if (m_lameInfo->flags == 0) {
		PERROR("lame_init failed.");
		return false;
	}
	
	if (m_channels == 1) {
		lame_set_mode(m_lameInfo->flags, MONO);
	}
	
	lame_set_num_channels(m_lameInfo->flags, m_channels);
	lame_set_in_samplerate(m_lameInfo->flags, m_rate);
	lame_set_out_samplerate(m_lameInfo->flags, m_rate);
	
	if(m_method == 0) {
		// Constant Bitrate
		lame_set_VBR(m_lameInfo->flags, vbr_off);
		lame_set_brate(m_lameInfo->flags, m_maxBitrate);
	}
	else if (m_method == 1) {
		// Average Bitrate
		lame_set_VBR(m_lameInfo->flags, vbr_abr);
		lame_set_VBR_mean_bitrate_kbps(m_lameInfo->flags, m_maxBitrate);
	}
	else if (m_method == 2) {
		// Variable Bitrate (old)
		lame_set_VBR(m_lameInfo->flags, vbr_default);
		lame_set_VBR_min_bitrate_kbps(m_lameInfo->flags, m_minBitrate);
		lame_set_VBR_max_bitrate_kbps(m_lameInfo->flags, m_maxBitrate);
	}
	else if (m_method == 3) {
		// Variable Bitrate (new)
		lame_set_VBR(m_lameInfo->flags, vbr_default);
		lame_set_VBR_min_bitrate_kbps(m_lameInfo->flags, m_minBitrate);
		lame_set_VBR_max_bitrate_kbps(m_lameInfo->flags, m_maxBitrate);
	}

	lame_set_quality(m_lameInfo->flags, m_quality);
	
	
	//
	// file options
	//
	lame_set_copyright(m_lameInfo->flags, false);
	lame_set_original(m_lameInfo->flags, true);
	lame_set_strict_ISO(m_lameInfo->flags, false);
	lame_set_error_protection(m_lameInfo->flags, false);
	
	return (lame_init_params(m_lameInfo->flags ) != -1);
}
예제 #7
0
long MP3_create(const char* format_parameters, amci_codec_fmt_info_t* format_description) {
  mp3_coder_state* coder_state;
  int ret_code;
  
  coder_state = malloc(sizeof(mp3_coder_state));
  if (!coder_state) {
    ERROR("no memory for allocating mp3 coder state\n");
    return -1;
  }
  
  DBG("MP3: creating lame %s\n", get_lame_version());
  format_description[0].id = 0; 
  coder_state->gfp = lame_init(); 
    
  if (!coder_state->gfp) {
    ERROR("initialiting lame\n");
    free(coder_state);
    return -1;
  }

  lame_set_errorf(coder_state->gfp, &no_output);
  lame_set_debugf(coder_state->gfp, &no_output);
  lame_set_msgf(coder_state->gfp, &no_output);
  
  lame_set_num_channels(coder_state->gfp,1);
  lame_set_in_samplerate(coder_state->gfp,8000);
  lame_set_brate(coder_state->gfp,16);
  lame_set_mode(coder_state->gfp,3); // mono
  lame_set_quality(coder_state->gfp,2);   /* 2=high  5 = medium  7=low */ 
  
  id3tag_init(coder_state->gfp);
  id3tag_set_title(coder_state->gfp, "mp3 voicemail by iptel.org");
  ret_code = lame_init_params(coder_state->gfp);
  
  if (ret_code < 0) {
    ERROR("lame encoder init failed: return code is %d\n", ret_code);
    free(coder_state);
    return -1;
  }
  

#ifdef WITH_MPG123DECODER
  coder_state->mpg123_h = mpg123_new(NULL, NULL);
  if (!coder_state->mpg123_h) {
    ERROR("initializing mpg123 decoder instance\n");
    return -1;
  }

  /* suppress output */
  mpg123_param(coder_state->mpg123_h, MPG123_FLAGS, MPG123_QUIET /* | MPG123_FORCE_MONO */,0);

/*   mpg123_param(coder_state->mpg123_h, MPG123_VERBOSE, 0, 0); */

#endif

  return (long)coder_state;
}
예제 #8
0
/*
 * Class:     com_example_mp3encodedemo_JNIMp3Encode
 * Method:    init
 * Signature: (III)V
 */
JNIEXPORT void JNICALL Java_com_example_soundtouchdemo_JNIMp3Encode_init
  (JNIEnv *env, jobject obj, jint channel, jint sampleRate, jint brate)
{
	lame = lame_init();
	lame_set_num_channels(lame, channel);
	lame_set_in_samplerate(lame, sampleRate);
	lame_set_brate(lame, brate);
	lame_set_mode(lame, 1);
	lame_set_quality(lame, 2);
	lame_init_params(lame);
}
예제 #9
0
/* initialize the lame library */
static void mp3streamout_tilde_lame_init(t_mp3streamout *x)
{
    int    ret;
    x->lgfp = lame_init(); /* set default parameters for now */

#ifdef _WIN32
    /* load lame_enc.dll library */
    HINSTANCE dll;
    dll=LoadLibrary("lame_enc.dll");
    if(dll==NULL)
    {
        error("mp3streamout~: error loading lame_enc.dll");
        closesocket(x->x_fd);
        x->x_fd = -1;
        outlet_float(x->x_obj.ob_outlet, 0);
        post("mp3streamout~: connection closed");
        return;
    }
#endif
    {
        const char *lameVersion = get_lame_version();
        logpost(NULL, 4,  "mp3streamout~ : using lame version : %s", lameVersion );
    }

    /* setting lame parameters */
    lame_set_num_channels( x->lgfp, 2);
    lame_set_in_samplerate( x->lgfp, sys_getsr() );
    lame_set_out_samplerate( x->lgfp, x->x_samplerate );
    lame_set_brate( x->lgfp, x->x_bitrate );
    lame_set_mode( x->lgfp, x->x_mp3mode );
    lame_set_quality( x->lgfp, x->x_mp3quality );
    lame_set_emphasis( x->lgfp, 1 );
    lame_set_original( x->lgfp, 1 );
    lame_set_copyright( x->lgfp, 1 ); /* viva free music societies !!! */
    lame_set_disable_reservoir( x->lgfp, 0 );
    //lame_set_padding_type( x->lgfp, PAD_NO ); /* deprecated in LAME */
    ret = lame_init_params( x->lgfp );
    if ( ret<0 )
    {
        post( "mp3streamout~ : error : lame params initialization returned : %d", ret );
    }
    else
    {
        x->x_lame=1;
        /* magic formula copied from windows dll for MPEG-I */
        x->x_lamechunk = 2*1152;

        post( "mp3streamout~ : lame initialization done. (%d)", x->x_lame );
    }
    lame_init_bitstream( x->lgfp );
}
예제 #10
0
/* DEPRECATED */
int
lame_set_mode_automs( lame_global_flags*  gfp,
                      int                 mode_automs )
{
    /* default = 0 (disabled) */

    /* enforce disable/enable meaning, if we need more than two values
       we need to switch to an enum to have an apropriate representation
       of the possible meanings of the value */
    if ( 0 > mode_automs || 1 < mode_automs )
        return -1;

    lame_set_mode( gfp, JOINT_STEREO );

    return 0;
}
void Java_mobi_cangol_mobile_utils_LameUtils_initEncoder(JNIEnv *env,
														 jobject jobj, jint in_num_channels, jint in_samplerate, jint in_brate,
														 jint in_mode, jint in_quality) {
	lame = lame_init();

	LOGD("Init parameters:",NULL);
	lame_set_num_channels(lame, in_num_channels);
	LOGD("Number of channels: %d", in_num_channels);
	lame_set_in_samplerate(lame, in_samplerate);
	LOGD("Sample rate: %d", in_samplerate);
	lame_set_brate(lame, in_brate);
	LOGD("Bitrate: %d", in_brate);
	lame_set_mode(lame, in_mode);
	LOGD("Mode: %d", in_mode);
	lame_set_quality(lame, in_quality);
	LOGD("Quality: %d", in_quality);

	int res = lame_init_params(lame);
	LOGD("Init returned: %d", res);
}
예제 #12
0
void Java_com_iliayugai_zapp_utils_CommonUtils_initEncoder(JNIEnv *env,
		jobject jobj, jint in_num_channels, jint in_samplerate, jint in_brate,
		jint in_mode, jint in_quality) {
	lame = lame_init();

	LOGD("Init parameters:");
	lame_set_num_channels(lame, in_num_channels);
	LOGD("Number of channels: %d", in_num_channels);
	lame_set_in_samplerate(lame, in_samplerate);
	LOGD("Sample rate: %d", in_samplerate);
	lame_set_brate(lame, in_brate);
	LOGD("Bitrate: %d", in_brate);
	lame_set_mode(lame, in_mode);
	LOGD("Mode: %d", in_mode);
	lame_set_quality(lame, in_quality);
	LOGD("Quality: %d", in_quality);

	int res = lame_init_params(lame);
	LOGD("Init returned: %d", res);
}
예제 #13
0
bool Mp3Writer::open(const QString &fn, long sr, bool s) {
	bool b = AudioFileWriter::open(fn + ".mp3", sr, s);

	if (!b)
		return false;

	lame = lame_init();
	if (!lame)
		return false;

	bitRate = preferences.get(Pref::OutputFormatMp3Bitrate).toInt();

	lame_set_in_samplerate(lame, sampleRate);
	lame_set_num_channels(lame, stereo ? 2 : 1);
	lame_set_out_samplerate(lame, sampleRate);
	// TODO: do we need this?
	lame_set_bWriteVbrTag(lame, 0);
	lame_set_mode(lame, stereo ? STEREO : MONO);
	lame_set_brate(lame, bitRate);
	if (lame_init_params(lame) == -1)
		return false;

	return true;
}
예제 #14
0
int main (int argc, char **argv) {
	char *url;
	int brate = 32;
	int c;
	while ((c = getopt(argc, argv, "u:b:")) != -1)
		switch (c) {
			case 'u':
				url = strdup(optarg);
				break;
			case 'b':
				brate = atoi(optarg);
				break;
			default:
				abort();
		}
// libev
	loop = ev_default_loop(0);
	ev_signal_init(&sigint_sig, sigint_cb, SIGINT);
	ev_signal_start (loop, &sigint_sig);
	ev_async_init(&async, send_cb);
	ev_async_start(loop, &async);

// jack
	jack_status_t status;
	ringbuf = jack_ringbuffer_create(buffer_size);
	framebuf = malloc(buffer_samples * sizeof(jack_default_audio_sample_t));
	client = jack_client_open("jack2rtmp", JackNullOption, &status);
	if (client == NULL) {
		fprintf(stderr, "jack server not running?\n");
		return EXIT_FAILURE;
	}
	jack_on_shutdown(client, jack_shutdown, NULL);
	jack_set_process_callback(client, jack_callback, NULL);
	//jack_set_latency_callback(client, jack_latency_callback, NULL);
	output_port = jack_port_register(client, "1", JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0);
	//jack_port_register(client, "2", JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0);
	if (jack_activate(client)) {
		fprintf(stderr, "cannot activate client");
	}

// lame
	mp3 = lame_init();
	lame_set_num_channels(mp3, 1);
	lame_set_mode(mp3, MONO);
	lame_set_in_samplerate(mp3, 48000);
	lame_set_out_samplerate(mp3, 44100);
	lame_set_quality(mp3, 2); // 0=best (very slow), 9=worst.
	lame_set_brate(mp3, brate);
	lame_init_params(mp3);

// rtmp
	rtmp_o = RTMP_Alloc();
	if (rtmp_o == NULL) {
		fprintf(stderr, "RTMP_Alloc\n");
		return EXIT_FAILURE;
	}
	RTMP_Init(rtmp_o);
	RTMP_SetupURL(rtmp_o, url);
	RTMP_EnableWrite(rtmp_o);
	if (!RTMP_Connect(rtmp_o, NULL) || !RTMP_ConnectStream(rtmp_o, 0)) {
		RTMP_Free(rtmp_o);
		fprintf(stderr, "Can not connect.\n");
		return EXIT_FAILURE;
	} else {
		fprintf(stderr, "Connected.\n");
	}

/*
	char *rtmp_buffer = malloc(4096);
	char *rtmp_top = rtmp_buffer + RTMP_MAX_HEADER_SIZE;
	char *outend = rtmp_buffer + 4096;
	*rtmp_top = (uint8_t) AMF_STRING;
	rtmp_top++;
	rtmp_top = put_amf_string(rtmp_top, "@setDataFrame");
	*rtmp_top = (uint8_t) AMF_STRING;
	rtmp_top++;
	rtmp_top = put_amf_string(rtmp_top, "onMetaData");

	// описание объекта
	*rtmp_top = (uint8_t) 8;
	rtmp_top++;
	rtmp_top = AMF_EncodeInt32(rtmp_top, outend, 8);
	// содержимое объекта
	rtmp_top = put_amf_string(rtmp_top, "duration");
	rtmp_top = AMF_EncodeNumber(rtmp_top, outend, 0);
	rtmp_top = put_amf_string(rtmp_top, "audiodatarate");
	rtmp_top = AMF_EncodeNumber(rtmp_top, outend, brate);
	rtmp_top = put_amf_string(rtmp_top, "audiosamplerate");
	rtmp_top = AMF_EncodeNumber(rtmp_top, outend, 44100);
	rtmp_top = put_amf_string(rtmp_top, "audiosamplesize");
	rtmp_top = AMF_EncodeNumber(rtmp_top, outend, 16);
	rtmp_top = put_amf_string(rtmp_top, "stereo");
	rtmp_top = AMF_EncodeBoolean(rtmp_top, outend, FALSE);
	rtmp_top = put_amf_string(rtmp_top, "audiocodecid");
	rtmp_top = AMF_EncodeNumber(rtmp_top, outend, 2);
	rtmp_top = put_amf_string(rtmp_top, "encoder");
	*rtmp_top = (uint8_t) AMF_STRING;
	rtmp_top++;
	rtmp_top = put_amf_string(rtmp_top, "Lavf53.21.1");
	rtmp_top = put_amf_string(rtmp_top, "filesize");
	rtmp_top = AMF_EncodeNumber(rtmp_top, outend, 0);
	// конец объекта
	rtmp_top = put_amf_string(rtmp_top, "");
	*rtmp_top = (uint8_t) AMF_OBJECT_END;
	rtmp_top++;

	rtmp_w.m_headerType = RTMP_PACKET_SIZE_LARGE;
	rtmp_w.m_nChannel = 0x04; // source channel
	rtmp_w.m_nBodySize = rtmp_top - rtmp_buffer - RTMP_MAX_HEADER_SIZE;
	rtmp_w.m_packetType = RTMP_PACKET_TYPE_INFO;
	rtmp_w.m_nInfoField2 = rtmp_o->m_stream_id;
	rtmp_w.m_body = rtmp_buffer + RTMP_MAX_HEADER_SIZE;
	if ( !RTMP_SendPacket(rtmp_o, &rtmp_w, TRUE) ) {
		fprintf(stderr, "RTMP_SendPacket error\n");
	}
	free(rtmp_buffer);
*/

	int off = 0;
	setsockopt(RTMP_Socket(rtmp_o), SOL_TCP, TCP_NODELAY, &off, sizeof(off));




	rtmp_i = TRUE;
	ev_loop(loop, 0);


	jack_ringbuffer_free(ringbuf);
	//jack_client_close(client);
	return EXIT_SUCCESS;
}
예제 #15
0
/*------------------------------------------------------------------------------
 *  Open an encoding session
 *----------------------------------------------------------------------------*/
bool
LameLibEncoder :: open ( void )
                                                            throw ( Exception )
{
    if ( isOpen() ) {
        close();
    }

    lameGlobalFlags = lame_init();

    // ugly lame returns -1 in a pointer on allocation errors
    if ( !lameGlobalFlags || ((int)lameGlobalFlags) == -1 ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib init error",
                         (int) lameGlobalFlags);
    }

    if ( 0 > lame_set_num_channels( lameGlobalFlags, getInChannel()) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib setting channels error",
                         getInChannel() );
    }

    if ( 0 > lame_set_mode( lameGlobalFlags,
                            getInChannel() == 1 ? MONO : JOINT_STEREO) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib setting mode error",
                         JOINT_STEREO );
    }

    reportEvent( 5, "set lame mode", lame_get_mode( lameGlobalFlags));
    
    reportEvent( 5,
                 "set lame channels",
                 lame_get_num_channels( lameGlobalFlags));
    
    if ( 0 > lame_set_in_samplerate( lameGlobalFlags, getInSampleRate()) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib setting input sample rate error",
                         getInSampleRate() );
    }

    reportEvent( 5,
                 "set lame in sample rate",
                 lame_get_in_samplerate( lameGlobalFlags));
    
    if ( 0 > lame_set_out_samplerate( lameGlobalFlags, getOutSampleRate()) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib setting output sample rate error",
                         getOutSampleRate() );
    }

    reportEvent( 5,
                 "set lame out sample rate",
                 lame_get_out_samplerate( lameGlobalFlags));
    
    if ( 0 > lame_set_brate( lameGlobalFlags, getOutBitrate()) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib setting output bit rate error",
                         getOutBitrate() );
    }

    reportEvent( 5, "set lame bit rate", lame_get_brate( lameGlobalFlags));
    
    if ( lowpass ) {
        if ( 0 > lame_set_lowpassfreq( lameGlobalFlags, lowpass) ) {
            throw Exception( __FILE__, __LINE__,
                             "lame lib setting lowpass frequency error",
                             lowpass );
        }

        reportEvent( 5,
                     "set lame lowpass frequency",
                     lame_get_lowpassfreq( lameGlobalFlags));
    }
    
    if ( highpass ) {
        if ( 0 > lame_set_highpassfreq( lameGlobalFlags, highpass) ) {
            throw Exception( __FILE__, __LINE__,
                             "lame lib setting highpass frequency error",
                             lowpass );
        }

        reportEvent( 5,
                     "set lame highpass frequency",
                     lame_get_highpassfreq( lameGlobalFlags));
    }

    // not configurable lame settings

    if ( 0 > lame_set_quality( lameGlobalFlags, 2) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib setting quality error",
                         2 );
    }

    reportEvent( 5, "set lame quality", lame_get_quality( lameGlobalFlags));
    
    if ( 0 > lame_set_exp_nspsytune( lameGlobalFlags, 1) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib setting  psycho acoustic model error");
    }

    reportEvent( 5,
                 "set lame psycho acoustic model",
                 lame_get_exp_nspsytune( lameGlobalFlags));
    
    if ( 0 > lame_set_error_protection( lameGlobalFlags, 1) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib setting error protection error",
                         1 );
    }

    reportEvent( 5,
                 "set lame error protection",
                 lame_get_error_protection( lameGlobalFlags));

    // let lame init its own params based on our settings
    if ( 0 > lame_init_params( lameGlobalFlags) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib initializing params error" );
    }

    lame_print_config( lameGlobalFlags);

    // open the underlying sink
    if ( !sink->open() ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib opening underlying sink error");
    }
 
    return true;
}
예제 #16
0
void do_broadcast(switch_stream_handle_t *stream)
{
	char *path_info = switch_event_get_header(stream->param_event, "http-path-info");
	char *file;
	lame_global_flags *gfp = NULL;
	switch_file_handle_t fh = { 0 };
	unsigned char mp3buf[TC_BUFFER_SIZE] = "";
	uint8_t buf[1024];
	int rlen;
	int is_local = 0;
	uint32_t interval = 20000;

	if (strstr(path_info + 7, "://")) {
		file = strdup(path_info + 7);
		is_local++;
	} else {
		file = switch_mprintf("%s/streamfiles/%s", SWITCH_GLOBAL_dirs.base_dir, path_info + 7);
	}
	assert(file);

	if (switch_core_file_open(&fh, file, 0, 0, SWITCH_FILE_FLAG_READ | SWITCH_FILE_DATA_SHORT, NULL) != SWITCH_STATUS_SUCCESS) {
		memset(&fh, 0, sizeof(fh));
		stream->write_function(stream, "Content-type: text/html\r\n\r\n<h2>File not found</h2>\n");
		goto end;
	}

	if (switch_test_flag((&fh), SWITCH_FILE_NATIVE)) {
		stream->write_function(stream, "Content-type: text/html\r\n\r\n<h2>File format not supported</h2>\n");
		goto end;
	}

	if (!(gfp = lame_init())) {
		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Could not allocate lame\n");
		goto end;
	}

	lame_set_num_channels(gfp, fh.channels);
	lame_set_in_samplerate(gfp, fh.samplerate);
	lame_set_brate(gfp, 16 * (fh.samplerate / 8000) * fh.channels);
	lame_set_mode(gfp, 3);
	lame_set_quality(gfp, 2);
	lame_set_errorf(gfp, log_error);
	lame_set_debugf(gfp, log_debug);
	lame_set_msgf(gfp, log_msg);
	lame_set_bWriteVbrTag(gfp, 0);
	lame_mp3_tags_fid(gfp, NULL);
	lame_init_params(gfp);
	lame_print_config(gfp);

	stream->write_function(stream, "Content-type: audio/mpeg\r\n" "Content-Disposition: inline; filename=\"%s.mp3\"\r\n\r\n", path_info + 7);

	if (fh.interval) {
		interval = fh.interval * 1000;
	}

	for (;;) {
		switch_size_t samples = sizeof(buf) / 2;

		switch_core_file_read(&fh, buf, &samples);

		if (is_local) {
			switch_yield(interval);
		}

		if (!samples) {
			break;
		}

		if ((rlen = lame_encode_buffer(gfp, (void *) buf, NULL, samples, mp3buf, sizeof(mp3buf))) < 0) {
			switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "MP3 encode error %d!\n", rlen);
			goto end;
		}

		if (rlen) {
			if (stream->raw_write_function(stream, mp3buf, rlen)) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Disconnected\n");
				goto end;
			}
		}
	}

	while ((rlen = lame_encode_flush(gfp, mp3buf, sizeof(mp3buf))) > 0) {
		if (stream->raw_write_function(stream, mp3buf, rlen)) {
			switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Disconnected\n");
			goto end;
		}
	}

  end:

	if (fh.channels) {
		switch_core_file_close(&fh);
	}

	switch_safe_free(file);

	if (gfp) {
		lame_close(gfp);
		gfp = NULL;
	}
}
예제 #17
0
//////////////////////////////////////////////////////////////////////
// Init - initialized or reiniyialized encoder SDK with given input 
// and output settings
//////////////////////////////////////////////////////////////////////
HRESULT CEncoder::Init()
{
    CAutoLock l(&m_lock);

    m_outOffset     = 0;
    m_outReadOffset = 0;

    m_bFinished     = FALSE;

    m_frameCount    = 0;

    if (!pgf)
    {
        if (!m_bInpuTypeSet || !m_bOutpuTypeSet)
            return E_UNEXPECTED;

        // Init Lame library
        // note: newer, safer interface which doesn't 
        // allow or require direct access to 'gf' struct is being written
        // see the file 'API' included with LAME.
        if (pgf = lame_init())
        {
            lame_set_num_channels(pgf, m_wfex.nChannels);
            lame_set_in_samplerate(pgf, m_wfex.nSamplesPerSec);
            lame_set_out_samplerate(pgf, m_mabsi.dwSampleRate);
            if ((lame_get_out_samplerate(pgf) >= 32000) && (m_mabsi.dwBitrate < 32))
                lame_set_brate(pgf, 32);
            else
                lame_set_brate(pgf, m_mabsi.dwBitrate);
            lame_set_VBR(pgf, m_mabsi.vmVariable);
            lame_set_VBR_min_bitrate_kbps(pgf, m_mabsi.dwVariableMin);
            lame_set_VBR_max_bitrate_kbps(pgf, m_mabsi.dwVariableMax);

            lame_set_copyright(pgf, m_mabsi.bCopyright);
            lame_set_original(pgf, m_mabsi.bOriginal);
            lame_set_error_protection(pgf, m_mabsi.bCRCProtect);

            lame_set_bWriteVbrTag(pgf, m_mabsi.dwXingTag);
            lame_set_strict_ISO(pgf, m_mabsi.dwStrictISO);
            lame_set_VBR_hard_min(pgf, m_mabsi.dwEnforceVBRmin);

            if (lame_get_num_channels(pgf) == 2 && !m_mabsi.bForceMono)
            {
                //int act_br = pgf->VBR ? pgf->VBR_min_bitrate_kbps + pgf->VBR_max_bitrate_kbps / 2 : pgf->brate;

                // Disabled. It's for user's consideration now
                //int rel = pgf->out_samplerate / (act_br + 1);
                //pgf->mode = rel < 200 ? m_mabsi.ChMode : JOINT_STEREO;

                lame_set_mode(pgf, m_mabsi.ChMode);
            }
            else
                lame_set_mode(pgf, MONO);

            if (lame_get_mode(pgf) == JOINT_STEREO)
                lame_set_force_ms(pgf, m_mabsi.dwForceMS);
            else
                lame_set_force_ms(pgf, 0);

//            pgf->mode_fixed = m_mabsi.dwModeFixed;

            if (m_mabsi.dwVoiceMode != 0)
            {
                lame_set_lowpassfreq(pgf,12000);
                ///pgf->VBR_max_bitrate_kbps = 160;
            }

            if (m_mabsi.dwKeepAllFreq != 0)
            {
                ///pgf->lowpassfreq = -1;
                ///pgf->highpassfreq = -1;
                /// not available anymore
            }

            lame_set_quality(pgf, m_mabsi.dwQuality);
            lame_set_VBR_q(pgf, m_mabsi.dwVBRq);

            lame_init_params(pgf);

            // encoder delay compensation
            {
                int const nch = lame_get_num_channels(pgf);
                short * start_padd = (short *)calloc(48, nch * sizeof(short));

				int out_bytes = 0;

                if (nch == 2)
                    out_bytes = lame_encode_buffer_interleaved(pgf, start_padd, 48, m_outFrameBuf, OUT_BUFFER_SIZE);
                else
                    out_bytes = lame_encode_buffer(pgf, start_padd, start_padd, 48, m_outFrameBuf, OUT_BUFFER_SIZE);

				if (out_bytes > 0)
					m_outOffset += out_bytes;

                free(start_padd);
            }

            return S_OK;
        }

        return E_FAIL;
    }

    return S_OK;
}
예제 #18
0
/*------------------------------------------------------------------------------
 *  Open an encoding session
 *----------------------------------------------------------------------------*/
bool
LameLibEncoder :: open ( void )
                                                            throw ( Exception )
{
    if ( isOpen() ) {
        close();
    }

    lameGlobalFlags = lame_init();

    // ugly lame returns -1 in a pointer on allocation errors
    if ( !lameGlobalFlags || ((int)lameGlobalFlags) == -1 ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib init error",
                         (int) lameGlobalFlags);
    }

    if ( 0 > lame_set_num_channels( lameGlobalFlags, getInChannel()) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib setting channels error",
                         getInChannel() );
    }

    if ( 0 > lame_set_mode( lameGlobalFlags,
                            getOutChannel() == 1 ? MONO : JOINT_STEREO) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib setting mode error",
                         JOINT_STEREO );
    }

    reportEvent( 5, "set lame mode", lame_get_mode( lameGlobalFlags));
    
    reportEvent( 5,
                 "set lame channels",
                 lame_get_num_channels( lameGlobalFlags));
    
    if ( 0 > lame_set_in_samplerate( lameGlobalFlags, getInSampleRate()) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib setting input sample rate error",
                         getInSampleRate() );
    }

    reportEvent( 5,
                 "set lame in sample rate",
                 lame_get_in_samplerate( lameGlobalFlags));
    
    if ( 0 > lame_set_out_samplerate( lameGlobalFlags, getOutSampleRate()) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib setting output sample rate error",
                         getOutSampleRate() );
    }

    reportEvent( 5,
                 "set lame out sample rate",
                 lame_get_out_samplerate( lameGlobalFlags));
    
    switch ( getOutBitrateMode() ) {
        
        case cbr: {

            if ( 0 > lame_set_brate( lameGlobalFlags, getOutBitrate()) ) {
                throw Exception( __FILE__, __LINE__,
                                "lame lib setting output bit rate error",
                                getOutBitrate() );
            }
            
            reportEvent( 5,
                         "set lame bit rate",
                         lame_get_brate( lameGlobalFlags));
            
            double  d = (1.0 - getOutQuality()) * 10.0;
            int     q = int (d + 0.499999);

            if ( 0 > lame_set_quality( lameGlobalFlags, q) ) {
                throw Exception( __FILE__, __LINE__,
                                "lame lib setting quality error", q);
            }
            
            reportEvent( 5,
                         "set lame quality",
                         lame_get_quality( lameGlobalFlags));
            } break;

        case abr:

            if ( 0 > lame_set_VBR( lameGlobalFlags,vbr_abr)) {
                throw Exception( __FILE__, __LINE__,
                                 "lame lib setting abr error", vbr_abr);
            }
            
            reportEvent( 5,
                         "set lame abr bitrate",
                         lame_get_VBR( lameGlobalFlags));
            
            if ( 0 > lame_set_VBR_mean_bitrate_kbps( lameGlobalFlags,
                                                     getOutBitrate())) {
                throw Exception( __FILE__, __LINE__,
                                 "lame lib setting abr mean bitrate error",
                                 getOutBitrate());
            }
            
            reportEvent( 5,
                         "set lame abr mean bitrate", 
                         lame_get_VBR_mean_bitrate_kbps( lameGlobalFlags));
            break;

        case vbr: {
        
            if ( 0 > lame_set_VBR( lameGlobalFlags, vbr_mtrh)) {
                throw Exception( __FILE__, __LINE__,
                                 "lame lib setting vbr error", vbr_mtrh );
            }
        
            reportEvent( 5,
                         "set lame vbr bitrate",
                         lame_get_VBR( lameGlobalFlags));

            double  d = (1.0 - getOutQuality()) * 10.0;
            int     q = int (d + 0.499999);

            if ( 0 > lame_set_VBR_q( lameGlobalFlags, q) ) {
                throw Exception( __FILE__, __LINE__,
                                 "lame lib setting vbr quality error", q);
            }
        
            reportEvent( 5,
                         "set lame vbr quality",
                         lame_get_VBR_q( lameGlobalFlags));
            } break;
    }


    if ( 0 > lame_set_lowpassfreq( lameGlobalFlags, lowpass) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib setting lowpass frequency error",
                         lowpass );
    }

    reportEvent( 5,
                 "set lame lowpass frequency",
                 lame_get_lowpassfreq( lameGlobalFlags));
    
    if ( 0 > lame_set_highpassfreq( lameGlobalFlags, highpass) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib setting highpass frequency error",
                         lowpass );
    }

    reportEvent( 5,
                 "set lame highpass frequency",
                 lame_get_highpassfreq( lameGlobalFlags));


    
    
    // not configurable lame settings
    
    if ( 0 > lame_set_exp_nspsytune( lameGlobalFlags, 1) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib setting  psycho acoustic model error");
    }

    reportEvent( 5,
                 "set lame psycho acoustic model",
                 lame_get_exp_nspsytune( lameGlobalFlags));
    
    if ( 0 > lame_set_error_protection( lameGlobalFlags, 1) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib setting error protection error",
                         1 );
    }

    reportEvent( 5,
                 "set lame error protection",
                 lame_get_error_protection( lameGlobalFlags));

    // let lame init its own params based on our settings
    if ( 0 > lame_init_params( lameGlobalFlags) ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib initializing params error" );
    }

    lame_print_config( lameGlobalFlags);

    // open the underlying sink
    if ( !sink->open() ) {
        throw Exception( __FILE__, __LINE__,
                         "lame lib opening underlying sink error");
    }
 
    return true;
}
//core function to encide WAV to MP3, relying on LAME library
void encode(char* wavFilePath, char* mp3FilePath) {

	FILE* wavFile = fopen(wavFilePath, "rb");
	FILE* mp3File = fopen(mp3FilePath, "wb+");
	int readPCMSampleCount;
	
	short int pcmBuffer[PCM_BUF_SIZE];
	unsigned char mp3Buffer[MP3_BUF_SIZE];

	unsigned short channel;
	unsigned long sampleRate;
	unsigned short blockAlign;

	lame_t lame = lame_init();

	if((wavFile == NULL) || (mp3File == NULL)) {
		printErrorMessage(FILE_IO_EXCEPTION);
		return;
	}
	
	if(lame == NULL) {
		printErrorMessage(LAME_API_ERROR);
		return;
	}

	//Set number of channels
	else if(fseek(wavFile, 22, SEEK_SET) != 0) {
		printErrorMessage(FILE_IO_EXCEPTION);
		return;
	}
	else if(fread(&channel, 2, 1, wavFile) < 1) {
		printErrorMessage(FILE_IO_EXCEPTION);
		return;
	}
	else {
		lame_set_num_channels(lame, channel);
		lame_set_mode(lame, (channel == 1) ? MONO : JOINT_STEREO);
	}

	//Set number of sample rate
	if(fseek(wavFile, 24, SEEK_SET) != 0) {
		printErrorMessage(FILE_IO_EXCEPTION);
		return;
	}
	else if(fread(&sampleRate, 4, 1, wavFile) < 1) {
		printErrorMessage(FILE_IO_EXCEPTION);
		return;
	}
	else {
		lame_set_in_samplerate(lame, sampleRate);
		lame_set_out_samplerate(lame, sampleRate);
	}
	
	//Set block align (byte/sample)
	if(fseek(wavFile, 32, SEEK_SET) != 0) {
		printErrorMessage(FILE_IO_EXCEPTION);
		return;
	}
	else if(fread(&blockAlign, 4, 1, wavFile) < 1) {
		printErrorMessage(FILE_IO_EXCEPTION);
		return;
	}

	//Look for data chank in the WAV file
	if(fseek(wavFile, 36, SEEK_SET) != 0) {
		printErrorMessage(FILE_IO_EXCEPTION);
		return;
	}
	else {
		char data[5];
		data[4] = '\0';
		
		while(fread(data, 4, 1, wavFile) == 1) {
			
			if(strcmp(data, "data") != 0) {
				//Search data chank by 1 byte
				fseek(wavFile, -3, SEEK_CUR);
				continue;
			}
			else if(fseek(wavFile, 4, SEEK_CUR)) {
				//data chank was found, then skip next 4 byte(indicating data size) to proceed
				printErrorMessage(FILE_IO_EXCEPTION);
				return;
			}
			else {
				//Now we are pointing the head of the PCM data, let's proceed
				break;
			}
		}
		
		//If data chank cannot be detected
		if(feof(wavFile)) {
			printErrorMessage(NOT_WAV_HEADER);
			return;
		}
	}
	
	
	//Encoding conficuration with "good" level quality
	lame_set_quality(lame, 5);
	lame_set_VBR(lame, vbr_default);
	lame_set_VBR_q(lame, 5);

	if(lame_init_params(lame) == -1) {
		printErrorMessage(LAME_API_ERROR);
		return;
	}
	
	//Perfoem encoding
	do {
		int encodedSampleCount;

		readPCMSampleCount = fread(pcmBuffer, blockAlign, PCM_BUF_SIZE / blockAlign, wavFile);
		
		if(readPCMSampleCount != 0) {

			if(channel == 1) { //For monoral case
				
				short int pcmBufferRight[PCM_BUF_SIZE];
				short int pcmBufferLeft[PCM_BUF_SIZE];
				int i;
				
				for(i = 0; i < PCM_BUF_SIZE; i++) {
					pcmBufferRight[i] = pcmBuffer[i];
					pcmBufferLeft[i] = pcmBuffer[i];
				}
				encodedSampleCount = lame_encode_buffer(lame, pcmBufferLeft, pcmBufferRight, readPCMSampleCount, mp3Buffer, MP3_BUF_SIZE);
			}
			
			else { //For stereo case
				encodedSampleCount = lame_encode_buffer_interleaved(lame, pcmBuffer, readPCMSampleCount, mp3Buffer, MP3_BUF_SIZE);
			}
		}
		else {
			encodedSampleCount = lame_encode_flush(lame, mp3Buffer, MP3_BUF_SIZE);
		}
		
		if(encodedSampleCount < 0) {
			printErrorMessage(LAME_API_ERROR);
		}
		else {
			fwrite(mp3Buffer, encodedSampleCount, 1, mp3File);
		}
		
	} while(readPCMSampleCount != 0);
	
	lame_close(lame);

	fclose(wavFile);
	fclose(mp3File);
}
void lame_encoder_impl_internal::set_stereo_mode(lame_encoder_impl_internal::StereoMode _Mode)
{
    int lameRet = lame_set_mode(m_Lame.get(), (MPEG_mode)_Mode);
    DT_LAME_CHECK_ERR(lameRet);
}
예제 #21
0
Mp3OutputStream::Mp3OutputStream() : global_fags(), bytes_per_sample(), bytes_per_second()
#ifdef Mp3OutputStream_Buffering
	m_inBuffer(), m_inBufferLenght(), m_inBufferPos(),
#endif
	,m_outBuffer(), m_outBufferLenght()
{
}

Mp3OutputStream::~Mp3OutputStream()
{
	Close();

#ifdef Mp3OutputStream_Buffering
	delete [] m_inBuffer;
#endif	

	delete [] m_outBuffer;
}
	
bool Mp3OutputStream::Initialize(int samplerate, int numchannels, int bitspersamp)
{
	global_fags = lame_init();
	if(global_fags == nullptr)
		return false;
	
	//lame_set_errorf(global_fags, lame_debugger);
	lame_set_analysis(global_fags, 0);

	lame_set_in_samplerate(global_fags, samplerate);
	lame_set_bWriteVbrTag( global_fags, 0 );
	lame_set_error_protection( global_fags, 0 );
	lame_set_extension( global_fags, 0 );
	lame_set_original( global_fags, 0 );
	lame_set_VBR(global_fags, vbr_off);

	lame_set_num_channels(global_fags, numchannels);
	lame_set_mode( global_fags, STEREO);

	lame_set_out_samplerate(global_fags, 44100);
	// 9 = LQP_LOW_QUALITY
	// 2 = LQP_HIGH_QUALITY:
	// 0 = LQP_VERYHIGH_QUALITY
	//lame_set_quality(global_fags, 0 );
	//lame_set_preset( global_fags, STANDARD_FAST);

	Settings& s = Global::Settings;
	
#ifdef STREAMER_PAYD_VERSION
	lame_set_brate(global_fags, std::min(Mp3OutputStreamConstants::Presets[s.encoder_preset], 320));
#else
	lame_set_brate(global_fags, 64);
#endif
	
	if(lame_init_params(global_fags) != 0)
	{
		OutputDebugString(_T("lame_init_params failed"));
		lame_close(global_fags);
		global_fags = nullptr;
		return false;
	}
	
	bytes_per_second = lame_get_brate(global_fags) / 8;

#ifdef Mp3OutputStream_Buffering
	if ( 0 == lame_get_version( global_fags ) )
	{
		// For MPEG-II, only 576 samples per frame per channel
		m_inBufferLenght = 576 * lame_get_num_channels( global_fags );
	}
	else
	{
		// For MPEG-I, 1152 samples per frame per channel
		m_inBufferLenght = 1152 * lame_get_num_channels( global_fags );
	}


	// delete buffers
	if(m_inBuffer != nullptr)
		delete [] m_inBuffer;
	m_inBuffer = new BYTE[m_inBufferLenght * 2];
#endif

	m_outBufferLenght = static_cast<int>( ( 1.25 * ( samplerate / numchannels ) + 7200 ) * 2.0);
	
	if(m_outBuffer != nullptr)
		delete [] m_outBuffer;
	m_outBuffer = new BYTE[m_outBufferLenght];

	bytes_per_sample = bitspersamp / 8;
	return true;
}
예제 #22
0
bool AUDMEncoder_Lame::initialize (void)
{

  int ret;
  MPEG_mode_e mmode;
  uint32_t frequence;



  lameFlags = lame_init ();
  if (lameFlags == NULL)
    return false;

  if (_incoming->getInfo ()->channels > 2)
    {
      printf ("[Lame]Too many channels\n");
      return false;
    }

  // recompute output length


  ret = lame_set_in_samplerate (MYFLAGS, wavheader.frequency);
  ret = lame_set_num_channels (MYFLAGS,  wavheader.channels);


  frequence = wavheader.frequency;
  printf ("[Lame] output frequency : %"PRIu32"\n", frequence);
  ret = lame_set_out_samplerate (MYFLAGS, frequence);

  ret = lame_set_quality (MYFLAGS, 2);

  if (wavheader.channels == 2)
    {

          mmode = STEREO;
    }
  else
    {
      mmode = MONO;
      printf ("[Lame] mono audio mp3");
    }

  ret = lame_set_brate (MYFLAGS, _config.bitrate);
  ret = lame_set_mode (MYFLAGS, mmode);	// 0 stereo 1 jstero
  ret = lame_set_quality (MYFLAGS, _config.quality);	// 0 stereo 1 jstero
  ret = lame_set_disable_reservoir (MYFLAGS, _config.disableBitReservoir);
  printf ("[Lame]Using quality of %d\n", lame_get_quality (MYFLAGS));
  // update bitrate in header
  wavheader.byterate = (_config.bitrate >> 3) * 1000;
#define BLOCK_SIZE 1152
  // configure CBR/ABR/...

  switch (_config.preset)
    {
    default:
    case ADM_LAME_PRESET_CBR:
        ADM_info("Lame : CBR Mode\n");
      break;
    case ADM_LAME_PRESET_ABR:
        ADM_info("Lame : ABR Mode\n");

      lame_set_preset (MYFLAGS, _config.bitrate);
      wavheader.blockalign = BLOCK_SIZE;
      break;
    case ADM_LAME_PRESET_EXTREME:
        ADM_info("Lame : Extreme Mode\n");
      wavheader.blockalign = BLOCK_SIZE;
      lame_set_preset (MYFLAGS, EXTREME);
      break;


    }
  ret = lame_init_params (MYFLAGS);
  if (ret == -1)
    {
        printf("[Lame] Init params failes %d\n",ret);
        return false;
    }

  lame_print_config (MYFLAGS);
  lame_print_internals (MYFLAGS);
  _chunk = BLOCK_SIZE * wavheader.channels;

  return true;
}
예제 #23
0
		bool WaveEncoder::ToMP3(const std::string &destFileName)
		{
			FILE *pcm = fopen(m_source.c_str(), "rb");
			FILE *mp3 = fopen(destFileName.c_str(), "wb");

			fseek(pcm, 0, SEEK_END);
			float pcm_length = ftell(pcm);
			float pcm_read = 0;
			int percentage = 0;
			fseek(pcm, 0, SEEK_SET);

			if(pcm == 0 || mp3 == 0)
			{
				m_lastErrorMessage = "Failed to open output stream.";
				return false;
			}

			int read, write;

			const int PCM_SIZE = 8192;
			const int MP3_SIZE = 8192;

			short int pcm_buffer[PCM_SIZE*2];
			unsigned char mp3_buffer[MP3_SIZE];

			lame_t lame = lame_init();
			lame_set_num_channels(lame, 2);

			if(m_quality == 3)
			{
				lame_set_in_samplerate(lame, 48000);
				lame_set_brate(lame, 320);
				lame_set_quality(lame, 2);
			}
			else if(m_quality == 2)
			{
				lame_set_in_samplerate(lame, 48000);
				lame_set_brate(lame, 192);
				lame_set_quality(lame, 5);
			}
			else if(m_quality == 1)
			{
				lame_set_in_samplerate(lame, 48000);
				lame_set_brate(lame, 128);
				lame_set_quality(lame, 7);
			}

			lame_set_mode(lame, JOINT_STEREO);
			lame_init_params(lame);

			printf("0%%");

			do
			{
				read = fread(pcm_buffer, 2*sizeof(short int), PCM_SIZE, pcm);
				if (read == 0)
					write = lame_encode_flush(lame, mp3_buffer, MP3_SIZE);
				else
					write = lame_encode_buffer_interleaved(lame, pcm_buffer, read, mp3_buffer, MP3_SIZE);
				fwrite(mp3_buffer, write, 1, mp3);

				pcm_read += PCM_SIZE * 2 * sizeof(short int);

				if(percentage < 10)
					printf("\b\b");
				else if(percentage < 100)
					printf("\b\b\b");
				else printf("\b\b\b\b");

				percentage = (int)(pcm_read / pcm_length * 100.f);
				printf("%d%%", percentage);
			} while (read != 0);

			lame_close(lame);
			fclose(mp3);
			fclose(pcm);

			return true;
		}
예제 #24
0
파일: mp3.cpp 프로젝트: zylthinking/s2m
int64_t mp3_open(const char* name, speex_info* info){
    if(info == NULL){
        return -1;
    }
    assert(info->sps == 8000 && info->bps == 16);
    assert(info->mode >= 0 && info->mode <= 10);
    assert(info->nr_chans == 1 || info->nr_chans == 2);

    lame_global_flags* gfp = lame_init();
    if(gfp == NULL){
        return -1;
    }

    lame_set_num_channels(gfp, info->nr_chans);
    lame_set_in_samplerate(gfp, info->sps);
    lame_set_brate(gfp, 128);
    
    if(info->nr_chans == 1){
        lame_set_mode(gfp, MONO);
    }else if(info->nr_chans == 2){
        lame_set_mode(gfp, STEREO);
    }
    lame_set_quality(gfp, 5);

    int32_t frame_len;
    FILE* file = NULL;
    char* obj_addr;
    unsigned int bytes;
    mp3_file* ptr;

    int n = lame_init_params(gfp);
    if(n == -1){
        goto LABEL1;
    }

    frame_len = 1.25 * speex_frame_samples + 7200;
    n = sizeof(mp3_file) + sizeof(CSpeexDecoder) + frame_len;
    ptr = (mp3_file *) malloc(n);
    if(ptr == NULL){
        goto LABEL1;
    }

    bytes = speex_bytes[info->nr_chans - 1][info->mode];
    obj_addr = (char *) (ptr + 1);
    ptr->dec = new (obj_addr) CSpeexDecoder(bytes, info->sps, info->nr_chans, info->bps);
    if(!ptr->dec->Start()){
        goto LABEL0;
    }

    ptr->gfp = gfp;
    ptr->length = frame_len;
    ptr->frame = (char *) (ptr->dec + 1);

    if(name){
        file = fopen(name, "wb");
        if(file == NULL){
            goto LABEL0;
        }
    }

    ptr->file = file;
    info->bytes = bytes;
    return (int64_t) ptr;

LABEL0:
    ptr->dec->~CSpeexDecoder();
    free(ptr);
LABEL1:
    lame_close(gfp);
    return -1;
}
예제 #25
0
파일: voix_enc.c 프로젝트: redstock/rvoix
int mp3_encode(int fd_in, int fd_out) {

    lame_global_flags *gfp;
    uint32_t i, j, k, count;
    int16_t *b0 = 0, *b1 = 0;
    unsigned char *c;
    int ret;
	
	/* calculate data size*/
	k = (uint32_t) lseek(fd_in, 0, SEEK_CUR);
	i = (uint32_t) lseek(fd_in, 0, SEEK_END);
	lseek(fd_in, k, SEEK_SET);
	i -= k;
	k = 0;

	gfp = lame_init();
	lame_set_errorf(gfp,lame_error_handler);
	lame_set_debugf(gfp,lame_error_handler);
	lame_set_msgf(gfp,lame_error_handler);
	lame_set_num_channels(gfp,2);
	lame_set_in_samplerate(gfp,8000);
#if 0
	lame_set_brate(gfp,64); /* compress 1:4 */
	lame_set_mode(gfp,0);   /* mode = stereo */
	lame_set_quality(gfp,2);   /* 2=high  5 = medium  7=low */
#else
	lame_set_quality(gfp,5);   /* 2=high  5 = medium  7=low */
	lame_set_mode(gfp,3);   /* mode = mono */
	if(lame_get_VBR(gfp) == vbr_off) lame_set_VBR(gfp, vbr_default);
	lame_set_VBR_quality(gfp, 7.0);
	lame_set_findReplayGain(gfp, 0);
	lame_set_bWriteVbrTag(gfp, 1);
	lame_set_out_samplerate(gfp,11025);
	/*   lame_set_num_samples(gfp,i/2); */
#endif
	if(lame_init_params(gfp) < 0) {
	     log_err("encode: failed to init lame");
	     return 0;
	}

#define CHSZ (512*1024)
#define OCHSZ (5*(CHSZ/8)+7200)

	b0 = (int16_t *) malloc(OCHSZ);
	b1 = (int16_t *) malloc(CHSZ);
	if(!b0 || !b1) {
	     log_err("encode: out of memory");
	     if(b0) free(b0);
	     if(b1) free(b1);
             return 0;
	}

	do {
	    ret = 0; j = 0; count = 0;
	    for(c = (unsigned char*) b1; count < CHSZ; c += ret) {
		j = (k + RSZ > i) ? i - k : RSZ;
		ret = read(fd_in,c,j);
		if(ret < 0) break;
		k += ret; count += ret;
		if(k == i) break;
	    }
	    if(ret < 0) break;
	    if(count) {
		ret = lame_encode_buffer(gfp,b1,b1,count/2,(unsigned char *)b0,OCHSZ);
		if(ret < 0) break;
		c = (unsigned char *) b0;
		count = ret;
		for(j = 0; j < count; j += RSZ, c += ret) {
		     ret = (j + RSZ > count) ? count - j : RSZ;
		     write(fd_out,c,ret);
		}
	    }
	} while(k < i);

	k = (uint32_t) lame_encode_flush(gfp,(unsigned char *)b0,OCHSZ);
	if(k) write(fd_out,b0,k);

	k = lame_get_lametag_frame(gfp,(unsigned char *)b0,OCHSZ);
	if(k > 0) write(fd_out,b0,k);

	lame_close(gfp);
	free(b0);
	free(b1);

     return 1;	
}
예제 #26
0
파일: mp3write~.c 프로젝트: Angeldude/pd
    /* initialize the lame library */
static int mp3write_tilde_lame_init(t_mp3write *x)
{
  time_t now;

    int    ret;
    x->lgfp = lame_init(); /* set default parameters for now */

#ifndef UNIX
    /* load lame_enc.dll library */
    HINSTANCE dll;
    dll=LoadLibrary("lame_enc.dll");
    if(dll==NULL)
    {
        error("mp3write~: error loading lame_enc.dll");
        closesocket(x->x_fd);
        x->x_fd = -1;
        post("mp3write~: connection closed");
        return -1;
    }
#endif
    {
       const char *lameVersion = get_lame_version();
       post( "mp3write~ : using lame version : %s", lameVersion );
    }

        /* setting lame parameters */
    lame_set_num_channels( x->lgfp, 2);
    lame_set_in_samplerate( x->lgfp, sys_getsr() );
    lame_set_out_samplerate( x->lgfp, x->x_samplerate );
    lame_set_brate( x->lgfp, x->x_bitrate );
    lame_set_mode( x->lgfp, x->x_mp3mode );
    lame_set_quality( x->lgfp, x->x_mp3quality );
    lame_set_emphasis( x->lgfp, 1 );
    lame_set_original( x->lgfp, 1 );
    lame_set_copyright( x->lgfp, 1 ); /* viva free music societies !!! */
    lame_set_disable_reservoir( x->lgfp, 0 );
    lame_set_padding_type( x->lgfp, PAD_NO );
    ret = lame_init_params( x->lgfp );
    if ( ret<0 ) {
       post( "mp3write~ : error : lame params initialization returned : %d", ret );
       return -1;
    } else {
       x->x_lame=1;
       /* magic formula copied from windows dll for MPEG-I */
       x->x_lamechunk = 2*1152;

       post( "mp3write~ : lame initialization done. (%d)", x->x_lame );
    }
    lame_init_bitstream( x->lgfp );

    /* setting tag information */
    id3tag_init(x->lgfp);
    id3tag_v1_only(x->lgfp);
    id3tag_space_v1(x->lgfp);
    id3tag_set_artist(x->lgfp, "Pd Session");
    now=time(NULL);
    sprintf( x->x_title, "Started at %s", ctime(&now) );
    id3tag_set_title(x->lgfp, x->x_title );
    
    return 0;

}
예제 #27
0
void do_telecast(switch_stream_handle_t *stream)
{
	char *path_info = switch_event_get_header(stream->param_event, "http-path-info");
	char *uuid = strdup(path_info + 4);
	switch_core_session_t *tsession;
	char *fname = "stream.mp3";

	if ((fname = strchr(uuid, '/'))) {
		*fname++ = '\0';
	}

	if (!(tsession = switch_core_session_locate(uuid))) {
		char *ref = switch_event_get_header(stream->param_event, "http-referer");
		stream->write_function(stream, "Content-type: text/html\r\n\r\n<h2>Not Found!</h2>\n" "<META http-equiv=\"refresh\" content=\"1;URL=%s\">", ref);
	} else {
		switch_media_bug_t *bug = NULL;
		switch_buffer_t *buffer = NULL;
		switch_mutex_t *mutex;
		switch_channel_t *channel = switch_core_session_get_channel(tsession);
		lame_global_flags *gfp = NULL;
		switch_codec_implementation_t read_impl = { 0 };
		switch_core_session_get_read_impl(tsession, &read_impl);

		if (switch_channel_test_flag(channel, CF_PROXY_MODE)) {
			switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Stepping into media path so this will work!\n");
			switch_ivr_media(uuid, SMF_REBRIDGE);
		}

		if (!(gfp = lame_init())) {
			switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Could not allocate lame\n");
			goto end;
		}
		lame_set_num_channels(gfp, read_impl.number_of_channels);
		lame_set_in_samplerate(gfp, read_impl.actual_samples_per_second);
		lame_set_brate(gfp, 16 * (read_impl.actual_samples_per_second / 8000) * read_impl.number_of_channels);
		lame_set_mode(gfp, 3);
		lame_set_quality(gfp, 2);
		lame_set_errorf(gfp, log_error);
		lame_set_debugf(gfp, log_debug);
		lame_set_msgf(gfp, log_msg);
		lame_set_bWriteVbrTag(gfp, 0);
		lame_mp3_tags_fid(gfp, NULL);
		lame_init_params(gfp);
		lame_print_config(gfp);

		switch_mutex_init(&mutex, SWITCH_MUTEX_NESTED, switch_core_session_get_pool(tsession));
		switch_buffer_create_dynamic(&buffer, 1024, 2048, 0);
		switch_buffer_add_mutex(buffer, mutex);

		if (switch_core_media_bug_add(tsession, "telecast", NULL,
									  telecast_callback, buffer, 0,
									  SMBF_READ_STREAM | SMBF_WRITE_STREAM | SMBF_READ_PING, &bug) != SWITCH_STATUS_SUCCESS) {
			goto end;
		}

		stream->write_function(stream, "Content-type: audio/mpeg\r\n" "Content-Disposition: inline; filename=\"%s\"\r\n\r\n", fname);

		while (switch_channel_ready(channel)) {
			unsigned char mp3buf[TC_BUFFER_SIZE] = "";
			int rlen;
			uint8_t buf[1024];
			switch_size_t bytes = 0;

			if (switch_buffer_inuse(buffer) >= 1024) {
				switch_buffer_lock(buffer);
				bytes = switch_buffer_read(buffer, buf, sizeof(buf));
				switch_buffer_unlock(buffer);
			} else {
				if (!bytes) {
					switch_cond_next();
					continue;
				}
				memset(buf, 0, bytes);
			}

			if ((rlen = lame_encode_buffer(gfp, (void *) buf, NULL, bytes / 2, mp3buf, sizeof(mp3buf))) < 0) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "MP3 encode error %d!\n", rlen);
				goto end;
			}

			if (rlen) {
				if (stream->raw_write_function(stream, mp3buf, rlen)) {
					switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Disconnected\n");
					goto end;
				}
			}
		}

	  end:

		switch_safe_free(uuid);

		if (gfp) {
			lame_close(gfp);
			gfp = NULL;
		}

		if (bug) {
			switch_core_media_bug_remove(tsession, &bug);
		}

		if (buffer) {
			switch_buffer_destroy(&buffer);
		}

		switch_core_session_rwunlock(tsession);
	}
}
//________________________________________________
//   Init lame encoder
// frequence    : Impose frequency , 0 means reuse the incoming fq
// mode                         : ADM_STEREO etc...
// bitrate              : Bitrate in kbps (96,192...)
// return 0 : init failed
//                              1 : init succeeded
//_______________________________________________
uint8_t AVDMProcessAudio_Lame::initLame(uint32_t frequence,
					uint32_t mode, 
					uint32_t bitrate,
					ADM_LAME_PRESET preset)
{

    int ret, ratio;
    MPEG_mode_e mmode;
    double dratio;

    myflags = lame_init();
    if (myflags == NULL)
	return 0;

	// recompute output length
	// since we compress recompute the length in bytes
    double comp;
    
    comp=_instream->getLength();
    comp/=2; // 16 bits sample
    comp/=_wavheader->channels; // We got sample
    comp/=_wavheader->frequency; // and no we got seconds
    comp*=bitrate*1000;
    comp/=8;			// now we got bytes
    _length=(uint32_t) floor(comp+1);
    printf("Incoming : %lu bytes, fq : %lu, channel : %lu bitrate: %lu outgoing : %lu\n",
    			_instream->getLength(),_wavheader->frequency,_wavheader->channels,bitrate,_length);
    //initLame(frequence,mode,bitrate);	
	
    ret = lame_set_in_samplerate(myflags, _instream->getInfo()->frequency);
    ret = lame_set_num_channels(myflags, _instream->getInfo()->channels);

    if (frequence == 0)		// keep same as instream
	frequence = _instream->getInfo()->frequency;
    printf("\n output frequency : %lu\n", frequence);
    ret = lame_set_out_samplerate(myflags, frequence);

    ret = lame_set_quality(myflags, 2);	//max quality);
    if (_instream->getInfo()->channels == 2)
      {
	  switch (mode)
	    {
	    case ADM_STEREO:
		mmode = STEREO;
		break;
	    case ADM_JSTEREO:
		mmode = JOINT_STEREO;
		break;
	    default:
		printf("\n **** unknown mode ***\n");
		mmode = STEREO;
		break;

	    }
    } else
    {
		mmode = MONO;
     	printf("\n mono audio mp3");
  	}

   _mode=mmode;

   

    ret = lame_set_brate(myflags, bitrate);
    ret = lame_set_mode(myflags, mmode);	// 0 stereo 1 jstero
    ret = lame_init_params(myflags);
    if (ret == -1)
	return 0;
    // update bitrate in header
    _wavheader->byterate = (bitrate >> 3) * 1000;
   
    // configure CBR/ABR/...
    _preset=preset;
    switch(preset)
    {
    	default:
    	case ADM_LAME_PRESET_CBR: break;
	case ADM_LAME_PRESET_ABR:
	  
	  lame_set_preset( myflags, bitrate);
	  _wavheader->blockalign=1152;
	 break;
	case ADM_LAME_PRESET_EXTREME: 
	  _wavheader->blockalign=1152;
	  lame_set_preset( myflags, EXTREME);	
	break;
    
    
    }
    
    lame_print_config(myflags);
    lame_print_internals(myflags);

    return 1;
}
예제 #29
0
static switch_status_t shout_file_open(switch_file_handle_t *handle, const char *path)
{
	shout_context_t *context;
	char *host, *file;
	char *username, *password, *port;
	char *err = NULL;
	const char *mpg123err = NULL;
	int portno = 0;

	if ((context = switch_core_alloc(handle->memory_pool, sizeof(*context))) == 0) {
		return SWITCH_STATUS_MEMERR;
	}

	if (!handle->samplerate) {
		handle->samplerate = 8000;
	}

	context->memory_pool = handle->memory_pool;
	context->samplerate = handle->samplerate;
	context->handle = handle;

	switch_thread_rwlock_create(&(context->rwlock), context->memory_pool);

	switch_thread_rwlock_rdlock(context->rwlock);

	switch_mutex_init(&context->audio_mutex, SWITCH_MUTEX_NESTED, context->memory_pool);

	if (switch_test_flag(handle, SWITCH_FILE_FLAG_READ)) {
		if (switch_buffer_create_dynamic(&context->audio_buffer, TC_BUFFER_SIZE, TC_BUFFER_SIZE * 2, 0) != SWITCH_STATUS_SUCCESS) {
			switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Memory Error!\n");
			goto error;
		}

		context->mh = our_mpg123_new(NULL, NULL);
		if (mpg123_format_all(context->mh) != MPG123_OK) {
			MPGERROR();
		}
		if (mpg123_param(context->mh, MPG123_FORCE_RATE, context->samplerate, 0) != MPG123_OK) {
			MPGERROR();
		}

		if (handle->handler) {
			if (mpg123_param(context->mh, MPG123_FLAGS, MPG123_SEEKBUFFER | MPG123_MONO_MIX, 0) != MPG123_OK) {
				MPGERROR();
			}
			if (mpg123_open_feed(context->mh) != MPG123_OK) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error opening mpg feed\n");
				mpg123err = mpg123_strerror(context->mh);
				goto error;
			}
			context->stream_url = switch_core_sprintf(context->memory_pool, "http://%s", path);
			context->prebuf = handle->prebuf;
			launch_read_stream_thread(context);
		} else {
			handle->seekable = 1;
			if (mpg123_param(context->mh, MPG123_FLAGS, MPG123_MONO_MIX, 0) != MPG123_OK) {
				MPGERROR();
			}
			if (mpg123_open(context->mh, path) != MPG123_OK) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error opening %s\n", path);
				mpg123err = mpg123_strerror(context->mh);
				goto error;
			}

		}
	} else if (switch_test_flag(handle, SWITCH_FILE_FLAG_WRITE)) {
		if (switch_test_flag(handle, SWITCH_FILE_WRITE_APPEND)) {
			switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Appending to MP3 not supported.\n");
		}
		if (!(context->gfp = lame_init())) {
			switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Could not allocate lame\n");
			goto error;
		}

		if (!handle->handler) {
			id3tag_init(context->gfp);
			id3tag_v2_only(context->gfp);
			id3tag_pad_v2(context->gfp);

		}
		context->channels = handle->channels;
		lame_set_brate(context->gfp, 16 * (handle->samplerate / 8000) * handle->channels);
		lame_set_num_channels(context->gfp, handle->channels);
		lame_set_in_samplerate(context->gfp, handle->samplerate);
		lame_set_out_samplerate(context->gfp, handle->samplerate);

		if (handle->channels == 2) {
			lame_set_mode(context->gfp, STEREO);
		} else {
			lame_set_mode(context->gfp, MONO);
		}
		lame_set_quality(context->gfp, 2);	/* 2=high  5 = medium  7=low */

		lame_set_errorf(context->gfp, log_error);
		lame_set_debugf(context->gfp, log_debug);
		lame_set_msgf(context->gfp, log_msg);

		if (handle->handler) {
			if (switch_buffer_create_dynamic(&context->audio_buffer, MY_BLOCK_SIZE, MY_BUF_LEN, 0) != SWITCH_STATUS_SUCCESS) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Memory Error!\n");
				goto error;
			}
			lame_set_bWriteVbrTag(context->gfp, 0);
			lame_mp3_tags_fid(context->gfp, NULL);

			username = switch_core_strdup(handle->memory_pool, path);
			if (!(password = strchr(username, ':'))) {
				err = "invalid url";
				goto error;
			}
			*password++ = '\0';

			if (!(host = strchr(password, '@'))) {
				err = "invalid url";
				goto error;
			}
			*host++ = '\0';

			if ((file = strchr(host, '/'))) {
				*file++ = '\0';
			} else {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Invalid URL: %s\n", path);
				goto error;
			}

			if ((port = strchr(host, ':'))) {
				*port++ = '\0';
				if (port) {
					portno = atoi(port);
				}
			}

			if (!portno) {
				portno = 8000;
			}

			if (!(context->shout = shout_new())) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Could not allocate shout_t\n");
				goto error;
			}

			if (shout_set_host(context->shout, host) != SHOUTERR_SUCCESS) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error setting hostname: %s\n", shout_get_error(context->shout));
				goto error;
			}

			if (shout_set_protocol(context->shout, SHOUT_PROTOCOL_HTTP) != SHOUTERR_SUCCESS) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error setting protocol: %s\n", shout_get_error(context->shout));
				goto error;
			}

			if (shout_set_port(context->shout, portno) != SHOUTERR_SUCCESS) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error setting port: %s\n", shout_get_error(context->shout));
				goto error;
			}

			if (shout_set_password(context->shout, password) != SHOUTERR_SUCCESS) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error setting password: %s\n", shout_get_error(context->shout));
				goto error;
			}

			if (shout_set_mount(context->shout, file) != SHOUTERR_SUCCESS) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error setting mount: %s\n", shout_get_error(context->shout));
				goto error;
			}

			if (shout_set_user(context->shout, username) != SHOUTERR_SUCCESS) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error setting user: %s\n", shout_get_error(context->shout));
				goto error;
			}

			if (shout_set_url(context->shout, "http://www.freeswitch.org") != SHOUTERR_SUCCESS) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error setting name: %s\n", shout_get_error(context->shout));
				goto error;
			}

			if (shout_set_description(context->shout, "FreeSWITCH mod_shout Broadcasting Module") != SHOUTERR_SUCCESS) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error setting description: %s\n", shout_get_error(context->shout));
				goto error;
			}

			if (shout_set_audio_info(context->shout, "bitrate", "24000") != SHOUTERR_SUCCESS) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error setting bitrate: %s\n", shout_get_error(context->shout));
				goto error;
			}

			if (shout_set_format(context->shout, SHOUT_FORMAT_MP3) != SHOUTERR_SUCCESS) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error setting format: %s\n", shout_get_error(context->shout));
				goto error;
			}

		} else {
			/* lame being lame and all has FILE * coded into it's API for some functions so we gotta use it */
			if (!(context->fp = fopen(path, "wb+"))) {
				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error opening %s\n", path);
				goto error;
			}
		}
	}

	handle->samples = 0;
	handle->format = 0;
	handle->sections = 0;
	handle->speed = 0;
	handle->private_info = context;
	switch_thread_rwlock_unlock(context->rwlock);

	return SWITCH_STATUS_SUCCESS;

  error:
	switch_thread_rwlock_unlock(context->rwlock);
	if (err) {
		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error: %s\n", err);
	}
	if (mpg123err) {
		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error from mpg123: %s\n", mpg123err);
	}
	free_context(context);
	return SWITCH_STATUS_GENERR;

}
예제 #30
0
bool ACMStream::open(const AEncodeProperties & the_Properties)
{
	bool bResult = false;

	// Init the MP3 Stream
	// Init the global flags structure
	gfp = lame_init();

	// Set input sample frequency
	lame_set_in_samplerate( gfp, my_SamplesPerSec );

	// Set output sample frequency
	lame_set_out_samplerate( gfp, my_OutBytesPerSec );

	lame_set_num_channels( gfp, my_Channels );
	if (my_Channels == 1)
		lame_set_mode( gfp, MONO );
	else
		lame_set_mode( gfp, (MPEG_mode_e)the_Properties.GetChannelModeValue()) ; /// \todo Get the mode from the default configuration

//	lame_set_VBR( gfp, vbr_off ); /// \note VBR not supported for the moment
	lame_set_VBR( gfp, my_VBRMode ); /// \note VBR not supported for the moment
	
	if (my_VBRMode == vbr_abr)
	{
		lame_set_VBR_q( gfp, 1 );

		lame_set_VBR_mean_bitrate_kbps( gfp, (my_AvgBytesPerSec * 8 + 500) / 1000 );

		if (24000 > lame_get_in_samplerate( gfp ))
		{
			// For MPEG-II
			lame_set_VBR_min_bitrate_kbps( gfp, 8);

			lame_set_VBR_max_bitrate_kbps( gfp, 160);
		}
		else
		{
			// For MPEG-I
			lame_set_VBR_min_bitrate_kbps( gfp, 32);

			lame_set_VBR_max_bitrate_kbps( gfp, 320);
		}
	}

	// Set bitrate
	lame_set_brate( gfp, my_AvgBytesPerSec * 8 / 1000 );

	/// \todo Get the mode from the default configuration
	// Set copyright flag?
	lame_set_copyright( gfp, the_Properties.GetCopyrightMode()?1:0 );
	// Do we have to tag  it as non original 
	lame_set_original( gfp, the_Properties.GetOriginalMode()?1:0 );
	// Add CRC?
	lame_set_error_protection( gfp, the_Properties.GetCRCMode()?1:0 );
	// Set private bit?
	lame_set_extension( gfp, the_Properties.GetPrivateMode()?1:0 );
	// INFO tag support not possible in ACM - it requires rewinding 
        // output stream to the beginning after encoding is finished.   
	lame_set_bWriteVbrTag( gfp, 0 );

	if (0 == lame_init_params( gfp ))
	{
		//LAME encoding call will accept any number of samples.  
		if ( 0 == lame_get_version( gfp ) )
		{
			// For MPEG-II, only 576 samples per frame per channel
			my_SamplesPerBlock = 576 * lame_get_num_channels( gfp );
		}
		else
		{
			// For MPEG-I, 1152 samples per frame per channel
			my_SamplesPerBlock = 1152 * lame_get_num_channels( gfp );
		}
	}

	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "version                =%d",lame_get_version( gfp ) );
	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "Layer                  =3");
	switch ( lame_get_mode( gfp ) )
	{
		case STEREO:       my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG,  "mode                   =Stereo" ); break;
		case JOINT_STEREO: my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG,  "mode                   =Joint-Stereo" ); break;
		case DUAL_CHANNEL: my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG,  "mode                   =Forced Stereo" ); break;
		case MONO:         my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG,  "mode                   =Mono" ); break;
		case NOT_SET:      /* FALLTROUGH */
		default:           my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG,  "mode                   =Error (unknown)" ); break;
	}

	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "sampling frequency     =%.1f kHz", lame_get_in_samplerate( gfp ) /1000.0 );
	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "bitrate                =%d kbps", lame_get_brate( gfp ) );
	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "Vbr Min bitrate        =%d kbps", lame_get_VBR_min_bitrate_kbps( gfp ) );
	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "Vbr Max bitrate        =%d kbps", lame_get_VBR_max_bitrate_kbps( gfp ) );
	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "Quality Setting        =%d", lame_get_quality( gfp ) );

	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "Low pass frequency     =%d", lame_get_lowpassfreq( gfp ) );
	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "Low pass width         =%d", lame_get_lowpasswidth( gfp ) );

	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "High pass frequency    =%d", lame_get_highpassfreq( gfp ) );
	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "High pass width        =%d", lame_get_highpasswidth( gfp ) );

	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "No Short Blocks        =%d", lame_get_no_short_blocks( gfp ) );

	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "de-emphasis            =%d", lame_get_emphasis( gfp ) );
	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "private flag           =%d", lame_get_extension( gfp ) );

	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "copyright flag         =%d", lame_get_copyright( gfp ) );
	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "original flag          =%d",	lame_get_original( gfp ) );
	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "CRC                    =%s", lame_get_error_protection( gfp ) ? "on" : "off" );
	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "Fast mode              =%s", ( lame_get_quality( gfp ) )? "enabled" : "disabled" );
	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "Force mid/side stereo  =%s", ( lame_get_force_ms( gfp ) )?"enabled":"disabled" );
	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "Padding Type           =%d", lame_get_padding_type( gfp ) );
	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "Disable Resorvoir      =%d", lame_get_disable_reservoir( gfp ) );
	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "VBR                    =%s, VBR_q =%d, VBR method =",
					( lame_get_VBR( gfp ) !=vbr_off ) ? "enabled": "disabled",
		            lame_get_VBR_q( gfp ) );

	switch ( lame_get_VBR( gfp ) )
	{
		case vbr_off:	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG,  "vbr_off" );	break;
		case vbr_mt :	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG,  "vbr_mt" );	break;
		case vbr_rh :	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG,  "vbr_rh" );	break;
		case vbr_mtrh:	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG,  "vbr_mtrh" );	break;
		case vbr_abr: 
			my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG,  "vbr_abr (average bitrate %d kbps)", lame_get_VBR_mean_bitrate_kbps( gfp ) );
		break;
		default:
			my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "error, unknown VBR setting");
		break;
	}

	my_debug->OutPut(DEBUG_LEVEL_FUNC_DEBUG, "Write VBR Header       =%s\n", ( lame_get_bWriteVbrTag( gfp ) ) ?"Yes":"No");

#ifdef FROM_DLL
beConfig.format.LHV1.dwReSampleRate		= my_OutBytesPerSec;	  // force the user resampling
#endif // FROM_DLL

	bResult = true;

	return bResult;
}