bool SoundFileWriterFlac::open(const std::string& filename, unsigned int sampleRate, unsigned int channelCount)
{
// Create the encoder
m_encoder = FLAC__stream_encoder_new();
if (!m_encoder)
{
err() << "Failed to write flac file \"" << filename << "\" (failed to allocate encoder)" << std::endl;
return false;
}
// Setup the encoder
FLAC__stream_encoder_set_channels(m_encoder, channelCount);
FLAC__stream_encoder_set_bits_per_sample(m_encoder, 16);
FLAC__stream_encoder_set_sample_rate(m_encoder, sampleRate);
// Initialize the output stream
if (FLAC__stream_encoder_init_file(m_encoder, filename.c_str(), NULL, NULL) != FLAC__STREAM_ENCODER_INIT_STATUS_OK)
{
err() << "Failed to write flac file \"" << filename << "\" (failed to open the file)" << std::endl;
close();
return false;
}
// Store the channel count
m_channelCount = channelCount;
return true;
}
int convertWavToFlac(const char *wave_file, const char *flac_file, int split_interval_seconds, char** out_flac_files) {
FILE *fin;
if((fin = fopen(wave_file, "rb")) == NULL) {
fprintf(stderr, "ERROR: opening %s for output\n", wave_file);
return 1;
}
// read wav header and validate it, note this will most likely fail for WAVE files not created by Apple
if(fread(buffer, 1, 44, fin) != 44 ||
memcmp(buffer, "RIFF", 4) ||
memcmp(buffer+36, "FLLR", 4)) {
fprintf(stderr, "ERROR: invalid/unsupported WAVE file\n");
fclose(fin);
return 1;
}
unsigned num_channels = ((unsigned)buffer[23] << 8) | buffer[22];;
unsigned sample_rate = ((((((unsigned)buffer[27] << 8) | buffer[26]) << 8) | buffer[25]) << 8) | buffer[24];
//unsigned byte_rate = ((((((unsigned)buffer[31] << 8) | buffer[30]) << 8) | buffer[29]) << 8) | buffer[28];
//unsigned block_align = ((unsigned)buffer[33] << 8) | buffer[32];
unsigned bps = ((unsigned)buffer[35] << 8) | buffer[34];
//Apple puts the number of filler bytes in the 2 bytes following FLLR in the filler chunk
//get the int value of the hex
unsigned filler_byte_count = ((unsigned)buffer[41] << 8) | buffer[40];
//swallow the filler bytes, exiting if there were not enough
if(fread(buffer, 1, filler_byte_count, fin) != filler_byte_count) {
fprintf(stderr, "ERROR: invalid number of filler bytes\n");
return 1;
}
//swallow the beginning of the data chunk, i.e. the word 'data'
unsigned data_subchunk_size = 0;
if(fread(buffer, 1, 8, fin) != 8 || memcmp(buffer, "data", 4)) {
fprintf(stderr, "ERROR: bad data start section\n");
return 1;
}
else {
//Subchunk2Size == NumSamples * NumChannels * BitsPerSample/8
data_subchunk_size = ((((((unsigned)buffer[7] << 8) | buffer[6]) << 8) | buffer[5]) << 8) | buffer[4];
}
//create the flac encoder
FLAC__StreamEncoder *encoder = FLAC__stream_encoder_new();
FLAC__stream_encoder_set_verify(encoder, true);
FLAC__stream_encoder_set_compression_level(encoder, 5);
FLAC__stream_encoder_set_channels(encoder, num_channels);
FLAC__stream_encoder_set_bits_per_sample(encoder, bps);
FLAC__stream_encoder_set_sample_rate(encoder, sample_rate);
//unknown total samples
FLAC__stream_encoder_set_total_samples_estimate(encoder, 0);
char* next_flac_file = malloc(sizeof(char) * 1024);
sprintf(next_flac_file, "%s.flac", flac_file);
//fprintf(stderr, "writing to new flac file %s\n", next_flac_file);
FLAC__stream_encoder_init_file(encoder, next_flac_file, progress_callback, NULL);
long total_bytes_read = 0;
int did_split_at_interval[1024];
for(int i = 0; i < 1024; i++) {
did_split_at_interval[i] = 0;
}
//read the wav file data chunk until we reach the end of the file.
size_t bytes_read = 0;
size_t need = (size_t)READSIZE;
int flac_file_index = 0;
while((bytes_read = fread(buffer, num_channels * (bps/8), need, fin)) != 0) {
/* convert the packed little-endian 16-bit PCM samples from WAVE into an interleaved FLAC__int32 buffer for libFLAC */
size_t i;
for(i = 0; i < bytes_read*num_channels; i++) {
/* inefficient but simple and works on big- or little-endian machines */
pcm[i] = (FLAC__int32)(((FLAC__int16)(FLAC__int8)buffer[2*i+1] << 8) | (FLAC__int16)buffer[2*i]);
}
/* feed samples to encoder */
FLAC__stream_encoder_process_interleaved(encoder, pcm, bytes_read);
total_bytes_read += bytes_read;
if(split_interval_seconds > 0) {
double elapsed_time_seconds = (total_bytes_read * 16) / (bps * sample_rate);
int interval = elapsed_time_seconds / split_interval_seconds;
if(interval > 0) {
if(!did_split_at_interval[interval-1]) {
//finish encoding the current flac file
FLAC__stream_encoder_finish(encoder);
FLAC__stream_encoder_delete(encoder);
//add the flac file to the out_flac_files output parameter
*(out_flac_files + flac_file_index) = next_flac_file;
flac_file_index += 1;
//get a new flac file name
//free(next_flac_file);
next_flac_file = malloc(sizeof(char) * 1024);
sprintf(next_flac_file, "%s_%d.flac", flac_file, interval);
//fprintf(stderr, "writing to new flac file %s\n", next_flac_file);
//create a new encoder
encoder = FLAC__stream_encoder_new();
FLAC__stream_encoder_set_verify(encoder, true);
FLAC__stream_encoder_set_compression_level(encoder, 5);
FLAC__stream_encoder_set_channels(encoder, num_channels);
FLAC__stream_encoder_set_bits_per_sample(encoder, bps);
FLAC__stream_encoder_set_sample_rate(encoder, sample_rate);
FLAC__stream_encoder_set_total_samples_estimate(encoder, 0);
FLAC__stream_encoder_init_file(encoder, next_flac_file, progress_callback, NULL);
//mark the interval as split
did_split_at_interval[interval-1] = 1;
}
}
}
}
//fprintf(stderr, "total bytes read: %ld\nbits per sample: %d\nsample rate: %d\n", total_bytes_read, bps, sample_rate);
*(out_flac_files + flac_file_index) = next_flac_file;
//cleanup
FLAC__stream_encoder_finish(encoder);
FLAC__stream_encoder_delete(encoder);
fclose(fin);
return 0;
}
int main(int argc, char *argv[])
{
FLAC__bool ok = true;
FLAC__StreamEncoder *encoder = 0;
FLAC__StreamEncoderInitStatus init_status;
FLAC__StreamMetadata *metadata[2];
FLAC__StreamMetadata_VorbisComment_Entry entry;
FILE *fin;
unsigned sample_rate = 0;
unsigned channels = 0;
unsigned bps = 0;
if(argc != 3) {
fprintf(stderr, "usage: %s infile.wav outfile.flac\n", argv[0]);
return 1;
}
if((fin = fopen(argv[1], "rb")) == NULL) {
fprintf(stderr, "ERROR: opening %s for output\n", argv[1]);
return 1;
}
/* read wav header and validate it */
if(
fread(buffer, 1, 44, fin) != 44 ||
memcmp(buffer, "RIFF", 4) ||
memcmp(buffer+8, "WAVEfmt \020\000\000\000\001\000\002\000", 16) ||
memcmp(buffer+32, "\004\000\020\000data", 8)
) {
fprintf(stderr, "ERROR: invalid/unsupported WAVE file, only 16bps stereo WAVE in canonical form allowed\n");
fclose(fin);
return 1;
}
sample_rate = ((((((unsigned)buffer[27] << 8) | buffer[26]) << 8) | buffer[25]) << 8) | buffer[24];
channels = 2;
bps = 16;
total_samples = (((((((unsigned)buffer[43] << 8) | buffer[42]) << 8) | buffer[41]) << 8) | buffer[40]) / 4;
/* allocate the encoder */
if((encoder = FLAC__stream_encoder_new()) == NULL) {
fprintf(stderr, "ERROR: allocating encoder\n");
fclose(fin);
return 1;
}
ok &= FLAC__stream_encoder_set_verify(encoder, true);
ok &= FLAC__stream_encoder_set_compression_level(encoder, 5);
ok &= FLAC__stream_encoder_set_channels(encoder, channels);
ok &= FLAC__stream_encoder_set_bits_per_sample(encoder, bps);
ok &= FLAC__stream_encoder_set_sample_rate(encoder, sample_rate);
ok &= FLAC__stream_encoder_set_total_samples_estimate(encoder, total_samples);
/* now add some metadata; we'll add some tags and a padding block */
if(ok) {
if(
(metadata[0] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_VORBIS_COMMENT)) == NULL ||
(metadata[1] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_PADDING)) == NULL ||
/* there are many tag (vorbiscomment) functions but these are convenient for this particular use: */
!FLAC__metadata_object_vorbiscomment_entry_from_name_value_pair(&entry, "ARTIST", "Some Artist") ||
!FLAC__metadata_object_vorbiscomment_append_comment(metadata[0], entry, /*copy=*/false) || /* copy=false: let metadata object take control of entry's allocated string */
!FLAC__metadata_object_vorbiscomment_entry_from_name_value_pair(&entry, "YEAR", "1984") ||
!FLAC__metadata_object_vorbiscomment_append_comment(metadata[0], entry, /*copy=*/false)
) {
fprintf(stderr, "ERROR: out of memory or tag error\n");
ok = false;
}
metadata[1]->length = 1234; /* set the padding length */
ok = FLAC__stream_encoder_set_metadata(encoder, metadata, 2);
}
/* initialize encoder */
if(ok) {
init_status = FLAC__stream_encoder_init_file(encoder, argv[2], progress_callback, /*client_data=*/NULL);
if(init_status != FLAC__STREAM_ENCODER_INIT_STATUS_OK) {
fprintf(stderr, "ERROR: initializing encoder: %s\n", FLAC__StreamEncoderInitStatusString[init_status]);
ok = false;
}
}
/* read blocks of samples from WAVE file and feed to encoder */
if(ok) {
size_t left = (size_t)total_samples;
while(ok && left) {
size_t need = (left>READSIZE? (size_t)READSIZE : (size_t)left);
if(fread(buffer, channels*(bps/8), need, fin) != need) {
fprintf(stderr, "ERROR: reading from WAVE file\n");
ok = false;
}
else {
/* convert the packed little-endian 16-bit PCM samples from WAVE into an interleaved FLAC__int32 buffer for libFLAC */
size_t i;
for(i = 0; i < need*channels; i++) {
/* inefficient but simple and works on big- or little-endian machines */
pcm[i] = (FLAC__int32)(((FLAC__int16)(FLAC__int8)buffer[2*i+1] << 8) | (FLAC__int16)buffer[2*i]);
}
/* feed samples to encoder */
ok = FLAC__stream_encoder_process_interleaved(encoder, pcm, need);
}
left -= need;
}
}
ok &= FLAC__stream_encoder_finish(encoder);
fprintf(stderr, "encoding: %s\n", ok? "succeeded" : "FAILED");
fprintf(stderr, " state: %s\n", FLAC__StreamEncoderStateString[FLAC__stream_encoder_get_state(encoder)]);
/* now that encoding is finished, the metadata can be freed */
FLAC__metadata_object_delete(metadata[0]);
FLAC__metadata_object_delete(metadata[1]);
FLAC__stream_encoder_delete(encoder);
fclose(fin);
return 0;
}
void CompressionTool::encodeRaw(const char *rawData, int length, int samplerate, const char *outname, AudioFormat compmode) {
print(" - len=%ld, ch=%d, rate=%d, %dbits", length, (rawAudioType.isStereo ? 2 : 1), samplerate, rawAudioType.bitsPerSample);
#ifdef USE_VORBIS
if (compmode == AUDIO_VORBIS) {
char outputString[256] = "";
int numChannels = (rawAudioType.isStereo ? 2 : 1);
int totalSamples = length / ((rawAudioType.bitsPerSample / 8) * numChannels);
int samplesLeft = totalSamples;
int eos = 0;
int totalBytes = 0;
vorbis_info vi;
vorbis_comment vc;
vorbis_dsp_state vd;
vorbis_block vb;
ogg_stream_state os;
ogg_page og;
ogg_packet op;
ogg_packet header;
ogg_packet header_comm;
ogg_packet header_code;
Common::File outputOgg(outname, "wb");
vorbis_info_init(&vi);
if (oggparms.nominalBitr > 0) {
int result = 0;
/* Input is in kbps, function takes bps */
result = vorbis_encode_setup_managed(&vi, numChannels, samplerate, (oggparms.maxBitr > 0 ? 1000 * oggparms.maxBitr : -1), (1000 * oggparms.nominalBitr), (oggparms.minBitr > 0 ? 1000 * oggparms.minBitr : -1));
if (result == OV_EFAULT) {
vorbis_info_clear(&vi);
error("Error: Internal Logic Fault");
} else if ((result == OV_EINVAL) || (result == OV_EIMPL)) {
vorbis_info_clear(&vi);
error("Error: Invalid bitrate parameters");
}
if (!oggparms.silent) {
sprintf(outputString, "Encoding to\n \"%s\"\nat average bitrate %i kbps (", outname, oggparms.nominalBitr);
if (oggparms.minBitr > 0) {
sprintf(outputString + strlen(outputString), "min %i kbps, ", oggparms.minBitr);
} else {
sprintf(outputString + strlen(outputString), "no min, ");
}
if (oggparms.maxBitr > 0) {
sprintf(outputString + strlen(outputString), "max %i kbps),\nusing full bitrate management engine\nSet optional hard quality restrictions\n", oggparms.maxBitr);
} else {
sprintf(outputString + strlen(outputString), "no max),\nusing full bitrate management engine\nSet optional hard quality restrictions\n");
}
}
} else {
int result = 0;
/* Quality input is -1 - 10, function takes -0.1 through 1.0 */
result = vorbis_encode_setup_vbr(&vi, numChannels, samplerate, oggparms.quality * 0.1f);
if (result == OV_EFAULT) {
vorbis_info_clear(&vi);
error("Internal Logic Fault");
} else if ((result == OV_EINVAL) || (result == OV_EIMPL)) {
vorbis_info_clear(&vi);
error("Invalid bitrate parameters");
}
if (!oggparms.silent) {
sprintf(outputString, "Encoding to\n \"%s\"\nat quality %2.2f", outname, oggparms.quality);
}
if ((oggparms.minBitr > 0) || (oggparms.maxBitr > 0)) {
struct ovectl_ratemanage_arg extraParam;
vorbis_encode_ctl(&vi, OV_ECTL_RATEMANAGE_GET, &extraParam);
extraParam.bitrate_hard_min = (oggparms.minBitr > 0 ? (1000 * oggparms.minBitr) : -1);
extraParam.bitrate_hard_max = (oggparms.maxBitr > 0 ? (1000 * oggparms.maxBitr) : -1);
extraParam.management_active = 1;
vorbis_encode_ctl(&vi, OV_ECTL_RATEMANAGE_SET, &extraParam);
if (!oggparms.silent) {
sprintf(outputString + strlen(outputString), " using constrained VBR (");
if (oggparms.minBitr != -1) {
sprintf(outputString + strlen(outputString), "min %i kbps, ", oggparms.minBitr);
} else {
sprintf(outputString + strlen(outputString), "no min, ");
}
if (oggparms.maxBitr != -1) {
sprintf(outputString + strlen(outputString), "max %i kbps)\nSet optional hard quality restrictions\n", oggparms.maxBitr);
} else {
sprintf(outputString + strlen(outputString), "no max)\nSet optional hard quality restrictions\n");
}
}
} else {
sprintf(outputString + strlen(outputString), "\n");
}
}
puts(outputString);
vorbis_encode_setup_init(&vi);
vorbis_comment_init(&vc);
vorbis_analysis_init(&vd, &vi);
vorbis_block_init(&vd, &vb);
ogg_stream_init(&os, 0);
vorbis_analysis_headerout(&vd, &vc, &header, &header_comm, &header_code);
ogg_stream_packetin(&os, &header);
ogg_stream_packetin(&os, &header_comm);
ogg_stream_packetin(&os, &header_code);
while (!eos) {
int result = ogg_stream_flush(&os,&og);
if (result == 0) {
break;
}
outputOgg.write(og.header, og.header_len);
outputOgg.write(og.body, og.body_len);
}
while (!eos) {
int numSamples = ((samplesLeft < 2048) ? samplesLeft : 2048);
float **buffer = vorbis_analysis_buffer(&vd, numSamples);
/* We must tell the encoder that we have reached the end of the stream */
if (numSamples == 0) {
vorbis_analysis_wrote(&vd, 0);
} else {
/* Adapted from oggenc 1.1.1 */
if (rawAudioType.bitsPerSample == 8) {
const byte *rawDataUnsigned = (const byte *)rawData;
for (int i = 0; i < numSamples; i++) {
for (int j = 0; j < numChannels; j++) {
buffer[j][i] = ((int)(rawDataUnsigned[i * numChannels + j]) - 128) / 128.0f;
}
}
} else if (rawAudioType.bitsPerSample == 16) {
if (rawAudioType.isLittleEndian) {
for (int i = 0; i < numSamples; i++) {
for (int j = 0; j < numChannels; j++) {
buffer[j][i] = ((rawData[(i * 2 * numChannels) + (2 * j) + 1] << 8) | (rawData[(i * 2 * numChannels) + (2 * j)] & 0xff)) / 32768.0f;
}
}
} else {
for (int i = 0; i < numSamples; i++) {
for (int j = 0; j < numChannels; j++) {
buffer[j][i] = ((rawData[(i * 2 * numChannels) + (2 * j)] << 8) | (rawData[(i * 2 * numChannels) + (2 * j) + 1] & 0xff)) / 32768.0f;
}
}
}
}
vorbis_analysis_wrote(&vd, numSamples);
}
while (vorbis_analysis_blockout(&vd, &vb) == 1) {
vorbis_analysis(&vb, NULL);
vorbis_bitrate_addblock(&vb);
while (vorbis_bitrate_flushpacket(&vd, &op)) {
ogg_stream_packetin(&os, &op);
while (!eos) {
int result = ogg_stream_pageout(&os, &og);
if (result == 0) {
break;
}
totalBytes += outputOgg.write(og.header, og.header_len);
totalBytes += outputOgg.write(og.body, og.body_len);
if (ogg_page_eos(&og)) {
eos = 1;
}
}
}
}
rawData += 2048 * (rawAudioType.bitsPerSample / 8) * numChannels;
samplesLeft -= 2048;
}
ogg_stream_clear(&os);
vorbis_block_clear(&vb);
vorbis_dsp_clear(&vd);
vorbis_info_clear(&vi);
if (!oggparms.silent) {
print("\nDone encoding file \"%s\"", outname);
print("\n\tFile length: %dm %ds", (int)(totalSamples / samplerate / 60), (totalSamples / samplerate % 60));
print("\tAverage bitrate: %.1f kb/s\n", (8.0 * (double)totalBytes / 1000.0) / ((double)totalSamples / (double)samplerate));
}
}
#endif
#ifdef USE_FLAC
if (compmode == AUDIO_FLAC) {
int i;
int numChannels = (rawAudioType.isStereo ? 2 : 1);
int samplesPerChannel = length / ((rawAudioType.bitsPerSample / 8) * numChannels);
FLAC__StreamEncoder *encoder;
FLAC__StreamEncoderInitStatus initStatus;
FLAC__int32 *flacData;
flacData = (FLAC__int32 *)malloc(samplesPerChannel * numChannels * sizeof(FLAC__int32));
if (rawAudioType.bitsPerSample == 8) {
for (i = 0; i < samplesPerChannel * numChannels; i++) {
FLAC__uint8 *rawDataUnsigned;
rawDataUnsigned = (FLAC__uint8 *)rawData;
flacData[i] = (FLAC__int32)rawDataUnsigned[i] - 0x80;
}
} else if (rawAudioType.bitsPerSample == 16) {
/* The rawData pointer is an 8-bit char so we must create a new pointer to access 16-bit samples */
FLAC__int16 *rawData16;
rawData16 = (FLAC__int16 *)rawData;
for (i = 0; i < samplesPerChannel * numChannels; i++) {
flacData[i] = (FLAC__int32)rawData16[i];
}
}
if (!flacparms.silent) {
print("Encoding to\n \"%s\"\nat compression level %d using blocksize %d\n", outname, flacparms.compressionLevel, flacparms.blocksize);
}
encoder = FLAC__stream_encoder_new();
FLAC__stream_encoder_set_bits_per_sample(encoder, rawAudioType.bitsPerSample);
FLAC__stream_encoder_set_blocksize(encoder, flacparms.blocksize);
FLAC__stream_encoder_set_channels(encoder, numChannels);
FLAC__stream_encoder_set_compression_level(encoder, flacparms.compressionLevel);
FLAC__stream_encoder_set_sample_rate(encoder, samplerate);
FLAC__stream_encoder_set_streamable_subset(encoder, false);
FLAC__stream_encoder_set_total_samples_estimate(encoder, samplesPerChannel);
FLAC__stream_encoder_set_verify(encoder, flacparms.verify);
initStatus = FLAC__stream_encoder_init_file(encoder, outname, NULL, NULL);
if (initStatus != FLAC__STREAM_ENCODER_INIT_STATUS_OK) {
char buf[2048];
sprintf(buf, "Error in FLAC encoder. (check the parameters)\nExact error was:%s", FLAC__StreamEncoderInitStatusString[initStatus]);
free(flacData);
throw ToolException(buf);
} else {
FLAC__stream_encoder_process_interleaved(encoder, flacData, samplesPerChannel);
}
FLAC__stream_encoder_finish(encoder);
FLAC__stream_encoder_delete(encoder);
free(flacData);
if (!flacparms.silent) {
print("\nDone encoding file \"%s\"", outname);
print("\n\tFile length: %dm %ds\n", (int)(samplesPerChannel / samplerate / 60), (samplesPerChannel / samplerate % 60));
}
}
#endif
}
bool ofxFlacEncoder::encode(string wavInput, string flacOutput) {
//ofLog(OF_LOG_VERBOSE, "init encoding (device%d)",deviceId);
FLAC__bool ok = true;
FLAC__StreamEncoder *encoder = 0;
FLAC__StreamEncoderInitStatus init_status;
FILE *fin;
unsigned sample_rate = 0;
unsigned channels = 0;
unsigned bps = 0;
if((fin = fopen(ofToDataPath(wavInput).c_str(), "rb")) == NULL){
//ofLog(OF_LOG_ERROR, "ERROR: opening %s for output\n", wavFile);
return false;
}
// read and validate wav header
if(fread(buffer, 1, 44, fin) != 44 || memcmp(buffer, "RIFF", 4)
|| memcmp(buffer + 8, "WAVEfmt \020\000\000\000\001\000\002\000", 16)
|| memcmp(buffer + 32, "\004\000\020\000data", 8)){
ofLog(OF_LOG_ERROR,
"invalid/unsupported WAVE file, only 16bps stereo WAVE in canonical form allowed");
//fclose(fin);
//return false;
}
sample_rate = ((((((unsigned) buffer[27] << 8) | buffer[26]) << 8) | buffer[25]) << 8)
| buffer[24];
channels = 2;
bps = 16;
total_samples = (((((((unsigned) buffer[43] << 8) | buffer[42]) << 8) | buffer[41]) << 8)
| buffer[40]) / 4;
// allocate the encoder
if((encoder = FLAC__stream_encoder_new()) == NULL){
ofLog(OF_LOG_ERROR, " allocating encoder\n");
fclose(fin);
return false;
}
ok &= FLAC__stream_encoder_set_verify(encoder, true);
ok &= FLAC__stream_encoder_set_compression_level(encoder, 5);
ok &= FLAC__stream_encoder_set_channels(encoder, channels);
ok &= FLAC__stream_encoder_set_bits_per_sample(encoder, bps);
ok &= FLAC__stream_encoder_set_sample_rate(encoder, sample_rate);
ok &= FLAC__stream_encoder_set_total_samples_estimate(encoder, total_samples);
// initialize encoder
if(ok){
init_status = FLAC__stream_encoder_init_file(encoder, ofToDataPath(flacOutput).c_str(), NULL, NULL);
if(init_status != FLAC__STREAM_ENCODER_INIT_STATUS_OK){
ofLog(OF_LOG_ERROR, "initializing encoder: ");
ofLog(OF_LOG_ERROR, FLAC__StreamEncoderInitStatusString[init_status]);
ok = false;
}
}
//ofLog(OF_LOG_VERBOSE, "start encoding (device%d)",deviceId);
/* read blocks of samples from WAVE file and feed to encoder */
if(ok){
size_t left = (size_t) total_samples;
while(ok && left){
size_t need = (left > READSIZE ? (size_t) READSIZE : (size_t) left);
if(fread(buffer, channels * (bps / 8), need, fin) != need){
ofLog(OF_LOG_ERROR, "reading from WAVE file");
ok = false;
}else{
/* convert the packed little-endian 16-bit PCM samples from WAVE into an interleaved FLAC__int32 buffer for libFLAC */
size_t i;
for(i = 0; i < need * channels; i++){
/* inefficient but simple and works on big- or little-endian machines */
pcm[i] = (FLAC__int32) (((FLAC__int16) (FLAC__int8) buffer[2 * i + 1] << 8)
| (FLAC__int16) buffer[2 * i]);
}
/* feed samples to encoder */
ok = FLAC__stream_encoder_process_interleaved(encoder, pcm, need);
}
left -= need;
}
}
ok &= FLAC__stream_encoder_finish(encoder);
// fprintf(stderr, "encoding: %s\n", ok ? "succeeded" : "FAILED");
// fprintf(stderr,
// " state: %s\n",
// FLAC__StreamEncoderStateString[FLAC__stream_encoder_get_state(encoder)]);
FLAC__stream_encoder_delete(encoder);
fclose(fin);
return ok;
}
const char*
_edje_multisense_encode_to_flac(char *snd_path, SF_INFO sfinfo)
{
unsigned int total_samples = 0; /* can use a 32-bit number due to WAVE size limitations */
FLAC__bool ok = 1;
FLAC__StreamEncoder *encoder = 0;
FLAC__StreamEncoderInitStatus init_status;
FLAC__StreamMetadata *metadata[2];
FLAC__StreamMetadata_VorbisComment_Entry entry;
SNDFILE *sfile;
sf_count_t size;
char *tmp;
sfile = sf_open(snd_path, SFM_READ, &sfinfo);
if (!sfile) return NULL;
if (!sf_format_check(&sfinfo))
{
sf_close(sfile);
return NULL;
}
size = sf_seek(sfile, 0, SEEK_END);
sf_seek(sfile, 0, SEEK_SET);
tmp = malloc(strlen(snd_path) + 1 + 5);
if (!tmp)
{
sf_close(sfile);
return NULL;
}
strcpy(tmp, snd_path);
snd_path = tmp;
strcat(snd_path, ".flac");
total_samples = size;
/* allocate the encoder */
if ((encoder = FLAC__stream_encoder_new()) == NULL)
{
ERR("ERROR: Creating FLAC encoder\n");
free(snd_path);
sf_close(sfile);
return NULL;
}
/* Verify it's own encoded output. This will slow the encoding process. */
ok &= FLAC__stream_encoder_set_verify(encoder, 1);
//Levels range from 0 (fastest, least compression) to 8 (slowest, most compression).
//A value larger than 8 will be treated as 8.
//5 is used for good compression and moderate compression/decompression speed.
ok &= FLAC__stream_encoder_set_compression_level(encoder, 5);
ok &= FLAC__stream_encoder_set_channels(encoder, sfinfo.channels);
ok &= FLAC__stream_encoder_set_bits_per_sample(encoder, 16);
ok &= FLAC__stream_encoder_set_sample_rate(encoder, sfinfo.samplerate);
ok &= FLAC__stream_encoder_set_total_samples_estimate(encoder, total_samples);
/* now add some metadata; we'll add some tags and a padding block */
if (ok)
{
if ((metadata[0] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_VORBIS_COMMENT)) == NULL
|| (metadata[1] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_PADDING)) == NULL
|| !FLAC__metadata_object_vorbiscomment_entry_from_name_value_pair(&entry, "Encoder", "flac")
|| !FLAC__metadata_object_vorbiscomment_append_comment(metadata[0], entry, 0))
{
ERR("ERROR: out of memory error or tag error\n");
ok = 0;
}
metadata[1]->length = 16; /* set the padding length */
ok = FLAC__stream_encoder_set_metadata(encoder, metadata, 2);
}
/* initialize encoder */
if (ok)
{
init_status = FLAC__stream_encoder_init_file(encoder, snd_path, NULL,
(void *)(long)(total_samples));
if (init_status != FLAC__STREAM_ENCODER_INIT_STATUS_OK)
{
ERR("ERROR: unable to initialize FLAC encoder: %s\n",
FLAC__StreamEncoderInitStatusString[init_status]);
ok = 0;
}
}
/* read blocks of samples from WAVE file and feed to encoder */
while (ok)
{
FLAC__int32 readbuffer[READBUF * 2];
sf_count_t count;
int i;
count = sf_readf_int(sfile, readbuffer, READBUF);
if (count <= 0) break;
for (i = 0; i < (count * sfinfo.channels); i++)
readbuffer[i] = readbuffer[i] >> 16;
ok = FLAC__stream_encoder_process_interleaved(encoder, readbuffer,
count);
}
FLAC__stream_encoder_finish(encoder);
/* now that encoding is finished, the metadata can be freed */
FLAC__metadata_object_delete(metadata[0]);
FLAC__metadata_object_delete(metadata[1]);
FLAC__stream_encoder_delete(encoder);
sf_close(sfile);
return (snd_path);
}
