int flac_enc_encode(flac_enc *flac, short *pcm_buf, int samples_per_chan, int channel)
{
int i;
int samples_left;
int chunk_size;
int pcm_32[8192];
int samples_written = 0;
chunk_size = 2048;
if (chunk_size > samples_per_chan)
chunk_size = samples_per_chan;
samples_left = samples_per_chan;
while(samples_left > 0)
{
// Convert 16bit samples to 32bit samples
for(i = 0; i < chunk_size * channel; i++)
pcm_32[i] = (int)pcm_buf[i+samples_written];
FLAC__stream_encoder_process_interleaved(flac->encoder, pcm_32, chunk_size);
samples_written += chunk_size*channel;
samples_left -= chunk_size;
if(samples_left < chunk_size)
chunk_size = samples_left;
}
return 0;
}
int Encode(void *ctx, int nNumBytesRead, uint8_t* pbtStream)
{
flac_context *context = (flac_context*)ctx;
if (!context || !context->encoder)
return 0;
int nLeftSamples = nNumBytesRead / 2; // each sample takes 2 bytes (16 bits per sample)
while (nLeftSamples > 0)
{
int nSamples = nLeftSamples > SAMPLES_BUF_SIZE ? SAMPLES_BUF_SIZE : nLeftSamples;
// convert the packed little-endian 16-bit PCM samples into an interleaved FLAC__int32 buffer for libFLAC
for (int i = 0; i < nSamples; i++)
{ // inefficient but simple and works on big- or little-endian machines.
context->samplesBuf[i] = (FLAC__int32)(((FLAC__int16)(FLAC__int8)pbtStream[2*i+1] << 8) | (FLAC__int16)pbtStream[2*i]);
}
// feed samples to encoder
if (!FLAC__stream_encoder_process_interleaved(context->encoder, context->samplesBuf, nSamples / 2))
{
return 0;
}
nLeftSamples -= nSamples;
pbtStream += nSamples * 2; // skip processed samples
}
return nNumBytesRead; // consumed everything
}
int CEncoderFlac::Encode(int numBytesRead, const uint8_t* stream)
{
if (!m_encoder)
return 0;
int nLeftSamples = numBytesRead / 2; // each sample takes 2 bytes (16 bits per sample)
while (nLeftSamples > 0)
{
int nSamples = nLeftSamples > SAMPLES_BUF_SIZE ? SAMPLES_BUF_SIZE : nLeftSamples;
// convert the packed little-endian 16-bit PCM samples into an interleaved FLAC__int32 buffer for libFLAC
for (int i = 0; i < nSamples; i++)
{ // inefficient but simple and works on big- or little-endian machines.
m_samplesBuf[i] = (FLAC__int32)(((FLAC__int16)(FLAC__int8)stream[2*i+1] << 8) | (FLAC__int16)stream[2*i]);
}
// feed samples to encoder
if (!FLAC__stream_encoder_process_interleaved(m_encoder, m_samplesBuf, nSamples / 2))
{
return 0;
}
nLeftSamples -= nSamples;
stream += nSamples * 2; // skip processed samples
}
return numBytesRead; // consumed everything
}
static size_t write_samples(sox_format_t * const ft, sox_sample_t const * const sampleBuffer, size_t const len)
{
priv_t * p = (priv_t *)ft->priv;
unsigned i;
for (i = 0; i < len; ++i) {
SOX_SAMPLE_LOCALS;
long pcm = SOX_SAMPLE_TO_SIGNED_32BIT(sampleBuffer[i], ft->clips);
p->decoded_samples[i] = pcm >> (32 - p->bits_per_sample);
switch (p->bits_per_sample) {
case 8: p->decoded_samples[i] =
SOX_SAMPLE_TO_SIGNED_8BIT(sampleBuffer[i], ft->clips);
break;
case 16: p->decoded_samples[i] =
SOX_SAMPLE_TO_SIGNED_16BIT(sampleBuffer[i], ft->clips);
break;
case 24: p->decoded_samples[i] = /* sign extension: */
SOX_SAMPLE_TO_SIGNED_24BIT(sampleBuffer[i],ft->clips) << 8;
p->decoded_samples[i] >>= 8;
break;
case 32: p->decoded_samples[i] =
SOX_SAMPLE_TO_SIGNED_32BIT(sampleBuffer[i],ft->clips);
break;
}
}
FLAC__stream_encoder_process_interleaved(p->encoder, p->decoded_samples, (unsigned) len / ft->signal.channels);
return FLAC__stream_encoder_get_state(p->encoder) == FLAC__STREAM_ENCODER_OK ? len : 0;
}
int32_t FlacAudioEncoder::encode(const int8_t* data, uint32_t len)
{
FLAC__int32 buffer[SAMPLES_BUF_SIZE];
int32_t channels = _profile->channels();
int32_t left_samples = len / channels;
while (left_samples > 0)
{
int32_t samples = left_samples > SAMPLES_BUF_SIZE ? SAMPLES_BUF_SIZE : left_samples;
// convert the packed little-endian 16-bit PCM samples into an
// interleaved FLAC__int32 buffer for libFLAC
const int16_t *src = reinterpret_cast<const int16_t*>(data);
for (uint32_t i = 0; i < samples; i++)
for (int c = 0; c < 2; c++)
buffer[2 * i + c] = src[2 * i + c];
// feed samples to encoder
if (not FLAC__stream_encoder_process_interleaved(_encoder, buffer, samples / 2))
{
THROW_EXCEPTION(AudioEncoderException, "ERROR: FLAC__stream_encoder_process_interleaved error");
}
left_samples -= samples;
data += samples * 2; // skip processed samples
}
return 1;
}
int WFLACEncoder::EncodeSamples(void *pSamples, unsigned int nNumberOfSamples)
{
if(c_fReady == 0)
{
err(ENCODER_NOT_READY);
return 0;
}
if(c_pcm_size < nNumberOfSamples)
{
FLAC__int32 *tmp = (FLAC__int32*) malloc(nNumberOfSamples * c_nNumberOfChannels * sizeof(FLAC__int32));
if(tmp == 0)
{
err(ENCODER_MEMORY_ALLOCATION_FAIL);
return 0;
}
if(c_pcm)
free(c_pcm);
c_pcm = tmp;
c_pcm_size = nNumberOfSamples;
}
unsigned int i;
switch(c_nBitBerSample)
{
case 16:
{
short *samples = (short*) pSamples;
for(i = 0; i < nNumberOfSamples * c_nNumberOfChannels; i++)
c_pcm[i] = samples[i];
}
break;
default:
return 1;
}
FLAC__stream_encoder_process_interleaved(c_encoder, c_pcm, nNumberOfSamples);
return 1;
}
virtual void WriteInterleavedConverted(size_t frameCount, const char *data)
{
ASSERT(inited);
if(!started)
{
StartStream();
}
ASSERT(inited && started);
sampleBuf.resize(frameCount * formatInfo.Channels);
switch(formatInfo.Sampleformat.GetBitsPerSample()/8)
{
case 1:
{
const uint8 *p = reinterpret_cast<const uint8*>(data);
for(std::size_t frame = 0; frame < frameCount; ++frame)
{
for(int channel = 0; channel < formatInfo.Channels; ++channel)
{
sampleBuf[frame * formatInfo.Channels + channel] = *p - 0x80;
p++;
}
}
}
break;
case 2:
{
const int16 *p = reinterpret_cast<const int16*>(data);
for(std::size_t frame = 0; frame < frameCount; ++frame)
{
for(int channel = 0; channel < formatInfo.Channels; ++channel)
{
sampleBuf[frame * formatInfo.Channels + channel] = *p;
p++;
}
}
}
break;
case 3:
{
const int24 *p = reinterpret_cast<const int24*>(data);
for(std::size_t frame = 0; frame < frameCount; ++frame)
{
for(int channel = 0; channel < formatInfo.Channels; ++channel)
{
sampleBuf[frame * formatInfo.Channels + channel] = *p;
p++;
}
}
}
break;
}
FLAC__stream_encoder_process_interleaved(encoder, &sampleBuf[0], frameCount);
}
OggFLAC_API FLAC__bool OggFLAC__seekable_stream_encoder_process_interleaved(OggFLAC__SeekableStreamEncoder *encoder, const FLAC__int32 buffer[], unsigned samples)
{
FLAC__ASSERT(0 != encoder);
FLAC__ASSERT(0 != encoder->private_);
FLAC__ASSERT(0 != encoder->protected_);
FLAC__ASSERT(0 != encoder->private_->FLAC_stream_encoder);
if(!FLAC__stream_encoder_process_interleaved(encoder->private_->FLAC_stream_encoder, buffer, samples)) {
encoder->protected_->state = OggFLAC__SEEKABLE_STREAM_ENCODER_FLAC_STREAM_ENCODER_ERROR;
return false;
}
else
return true;
}
static void flac_write(gpointer data, gint length)
{
#if 1
FLAC__int32 *encbuffer[2];
short int *tmpdata = data;
int i;
encbuffer[0] = g_new0(FLAC__int32, length / input.channels);
encbuffer[1] = g_new0(FLAC__int32, length / input.channels);
if (input.channels == 1)
{
for (i = 0; i < (length / 2); i++)
{
encbuffer[0][i] = tmpdata[i];
encbuffer[1][i] = tmpdata[i];
}
}
else
{
for (i = 0; i < (length / 4); i++)
{
encbuffer[0][i] = tmpdata[2 * i];
encbuffer[1][i] = tmpdata[2 * i + 1];
}
}
FLAC__stream_encoder_process(flac_encoder, (const FLAC__int32 **)encbuffer, length / (input.channels * 2));
g_free(encbuffer[0]);
g_free(encbuffer[1]);
#else
FLAC__int32 *encbuffer;
gint16 *tmpdata = data;
int i;
encbuffer = g_new0(FLAC__int32, length);
for (i=0; i < length; i++) {
encbuffer[i] = tmpdata[i];
}
FLAC__stream_encoder_process_interleaved(flac_encoder, encbuffer, length);
g_free(encbuffer);
#endif
}
static bool
flac_encoder_write(struct encoder *_encoder,
const void *data, size_t length,
G_GNUC_UNUSED GError **error)
{
struct flac_encoder *encoder = (struct flac_encoder *)_encoder;
unsigned num_frames, num_samples;
void *exbuffer;
const void *buffer = NULL;
/* format conversion */
num_frames = length / audio_format_frame_size(&encoder->audio_format);
num_samples = num_frames * encoder->audio_format.channels;
switch (encoder->audio_format.format) {
case SAMPLE_FORMAT_S8:
exbuffer = pcm_buffer_get(&encoder->expand_buffer, length*4);
pcm8_to_flac(exbuffer, data, num_samples);
buffer = exbuffer;
break;
case SAMPLE_FORMAT_S16:
exbuffer = pcm_buffer_get(&encoder->expand_buffer, length*2);
pcm16_to_flac(exbuffer, data, num_samples);
buffer = exbuffer;
break;
case SAMPLE_FORMAT_S24_P32:
case SAMPLE_FORMAT_S32:
/* nothing need to be done; format is the same for
both mpd and libFLAC */
buffer = data;
break;
}
/* feed samples to encoder */
if (!FLAC__stream_encoder_process_interleaved(encoder->fse, buffer,
num_frames)) {
g_set_error(error, flac_encoder_quark(), 0,
"flac encoder process failed");
return false;
}
return true;
}
void SoundFileWriterFlac::write(const Int16* samples, Uint64 count)
{
while (count > 0)
{
// Make sure that we don't process too many samples at once
unsigned int frames = std::min(static_cast<unsigned int>(count / m_channelCount), 10000u);
// Convert the samples to 32-bits
m_samples32.assign(samples, samples + frames * m_channelCount);
// Write them to the FLAC stream
FLAC__stream_encoder_process_interleaved(m_encoder, &m_samples32[0], frames);
// Next chunk
count -= m_samples32.size();
samples += m_samples32.size();
}
}
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
}
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 SoftFlacEncoder::onQueueFilled(OMX_U32 portIndex) {
UNUSED_UNLESS_VERBOSE(portIndex);
ALOGV("SoftFlacEncoder::onQueueFilled(portIndex=%d)", portIndex);
if (mSignalledError) {
return;
}
List<BufferInfo *> &inQueue = getPortQueue(0);
List<BufferInfo *> &outQueue = getPortQueue(1);
while (!inQueue.empty() && !outQueue.empty()) {
BufferInfo *inInfo = *inQueue.begin();
OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
BufferInfo *outInfo = *outQueue.begin();
OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
inQueue.erase(inQueue.begin());
inInfo->mOwnedByUs = false;
notifyEmptyBufferDone(inHeader);
outHeader->nFilledLen = 0;
outHeader->nFlags = OMX_BUFFERFLAG_EOS;
outQueue.erase(outQueue.begin());
outInfo->mOwnedByUs = false;
notifyFillBufferDone(outHeader);
return;
}
if (inHeader->nFilledLen > kMaxInputBufferSize) {
ALOGE("input buffer too large (%d).", inHeader->nFilledLen);
mSignalledError = true;
notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
return;
}
assert(mNumChannels != 0);
mEncoderWriteData = true;
mEncoderReturnedEncodedData = false;
mEncoderReturnedNbBytes = 0;
mCurrentInputTimeStamp = inHeader->nTimeStamp;
const unsigned nbInputFrames = inHeader->nFilledLen / (2 * mNumChannels);
const unsigned nbInputSamples = inHeader->nFilledLen / 2;
const OMX_S16 * const pcm16 = reinterpret_cast<OMX_S16 *>(inHeader->pBuffer);
CHECK_LE(nbInputSamples, 2 * kMaxNumSamplesPerFrame);
for (unsigned i=0 ; i < nbInputSamples ; i++) {
mInputBufferPcm32[i] = (FLAC__int32) pcm16[i];
}
ALOGV(" about to encode %u samples per channel", nbInputFrames);
FLAC__bool ok = FLAC__stream_encoder_process_interleaved(
mFlacStreamEncoder,
mInputBufferPcm32,
nbInputFrames /*samples per channel*/ );
if (ok) {
if (mEncoderReturnedEncodedData && (mEncoderReturnedNbBytes != 0)) {
ALOGV(" dequeueing buffer on output port after writing data");
outInfo->mOwnedByUs = false;
outQueue.erase(outQueue.begin());
outInfo = NULL;
notifyFillBufferDone(outHeader);
outHeader = NULL;
mEncoderReturnedEncodedData = false;
} else {
ALOGV(" encoder process_interleaved returned without data to write");
}
} else {
ALOGE(" error encountered during encoding");
mSignalledError = true;
notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
return;
}
inInfo->mOwnedByUs = false;
inQueue.erase(inQueue.begin());
inInfo = NULL;
notifyEmptyBufferDone(inHeader);
inHeader = NULL;
}
}
QString & SampleBuffer::toBase64( QString & _dst ) const
{
#ifdef LMMS_HAVE_FLAC_STREAM_ENCODER_H
const f_cnt_t FRAMES_PER_BUF = 1152;
FLAC__StreamEncoder * flac_enc = FLAC__stream_encoder_new();
FLAC__stream_encoder_set_channels( flac_enc, DEFAULT_CHANNELS );
FLAC__stream_encoder_set_blocksize( flac_enc, FRAMES_PER_BUF );
/* FLAC__stream_encoder_set_do_exhaustive_model_search( flac_enc, true );
FLAC__stream_encoder_set_do_mid_side_stereo( flac_enc, true );*/
FLAC__stream_encoder_set_sample_rate( flac_enc,
engine::mixer()->sampleRate() );
QBuffer ba_writer;
ba_writer.open( QBuffer::WriteOnly );
FLAC__stream_encoder_set_write_callback( flac_enc,
flacStreamEncoderWriteCallback );
FLAC__stream_encoder_set_metadata_callback( flac_enc,
flacStreamEncoderMetadataCallback );
FLAC__stream_encoder_set_client_data( flac_enc, &ba_writer );
if( FLAC__stream_encoder_init( flac_enc ) != FLAC__STREAM_ENCODER_OK )
{
printf( "error within FLAC__stream_encoder_init()!\n" );
}
f_cnt_t frame_cnt = 0;
while( frame_cnt < m_frames )
{
f_cnt_t remaining = qMin<f_cnt_t>( FRAMES_PER_BUF,
m_frames - frame_cnt );
FLAC__int32 buf[FRAMES_PER_BUF * DEFAULT_CHANNELS];
for( f_cnt_t f = 0; f < remaining; ++f )
{
for( ch_cnt_t ch = 0; ch < DEFAULT_CHANNELS; ++ch )
{
buf[f*DEFAULT_CHANNELS+ch] = (FLAC__int32)(
Mixer::clip( m_data[f+frame_cnt][ch] ) *
OUTPUT_SAMPLE_MULTIPLIER );
}
}
FLAC__stream_encoder_process_interleaved( flac_enc, buf,
remaining );
frame_cnt += remaining;
}
FLAC__stream_encoder_finish( flac_enc );
FLAC__stream_encoder_delete( flac_enc );
printf("%d %d\n", frame_cnt, (int)ba_writer.size() );
ba_writer.close();
base64::encode( ba_writer.buffer().data(), ba_writer.buffer().size(),
_dst );
#else /* LMMS_HAVE_FLAC_STREAM_ENCODER_H */
base64::encode( (const char *) m_data,
m_frames * sizeof( sampleFrame ), _dst );
#endif /* LMMS_HAVE_FLAC_STREAM_ENCODER_H */
return _dst;
}
static int output_data(char *buf, int32 nbytes)
{
FLAC__int32 *oggbuf;
FLAC__int16 *s;
int i;
int nch = (dpm.encoding & PE_MONO) ? 1 : 2;
FLAC_ctx *ctx = flac_ctx;
if (dpm.fd < 0)
return 0;
if (ctx == NULL) {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "FLAC stream is not initialized");
return -1;
}
oggbuf = (FLAC__int32 *)safe_malloc(nbytes * sizeof(FLAC__int32) / nch);
/*
packing 16 -> 32 bit sample
*/
s = (FLAC__int16 *)buf;
for (i = 0; i < nbytes / nch; i++) {
oggbuf[i] = *s++;
}
#ifdef LEGACY_FLAC
#ifdef AU_OGGFLAC
if (flac_options.isogg) {
ctx->state.ogg = OggFLAC__stream_encoder_get_state(ctx->encoder.ogg.stream);
if (ctx->state.ogg != OggFLAC__STREAM_ENCODER_OK) {
if (ctx->state.ogg == OggFLAC__STREAM_ENCODER_FLAC_STREAM_ENCODER_ERROR) {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "FLAC stream verify error (%s)",
FLAC__StreamDecoderStateString[OggFLAC__stream_encoder_get_verify_decoder_state(ctx->encoder.ogg.stream)]);
}
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "cannot encode OggFLAC stream (%s)",
OggFLAC__StreamEncoderStateString[ctx->state.ogg]);
flac_session_close();
return -1;
}
if (!OggFLAC__stream_encoder_process_interleaved(ctx->encoder.ogg.stream, oggbuf,
nbytes / nch / 2)) {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "cannot encode OggFLAC stream");
flac_session_close();
return -1;
}
}
else
#endif /* AU_OGGFLAC */
if (flac_options.seekable) {
ctx->state.s_flac = FLAC__seekable_stream_encoder_get_state(ctx->encoder.flac.s_stream);
if (ctx->state.s_flac != FLAC__STREAM_ENCODER_OK) {
if (ctx->state.s_flac == FLAC__STREAM_ENCODER_VERIFY_DECODER_ERROR ||
ctx->state.s_flac == FLAC__STREAM_ENCODER_VERIFY_MISMATCH_IN_AUDIO_DATA) {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "FLAC stream verify error (%s)",
FLAC__SeekableStreamDecoderStateString[FLAC__seekable_stream_encoder_get_verify_decoder_state(ctx->encoder.flac.s_stream)]);
}
else {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "cannot encode FLAC stream (%s)",
FLAC__SeekableStreamEncoderStateString[ctx->state.s_flac]);
}
flac_session_close();
return -1;
}
if (!FLAC__seekable_stream_encoder_process_interleaved(ctx->encoder.flac.s_stream, oggbuf,
nbytes / nch / 2 )) {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "cannot encode FLAC stream");
flac_session_close();
return -1;
}
}
else
{
ctx->state.flac = FLAC__stream_encoder_get_state(ctx->encoder.flac.stream);
if (ctx->state.flac != FLAC__STREAM_ENCODER_OK) {
if (ctx->state.flac == FLAC__STREAM_ENCODER_VERIFY_DECODER_ERROR ||
ctx->state.flac == FLAC__STREAM_ENCODER_VERIFY_MISMATCH_IN_AUDIO_DATA) {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "FLAC stream verify error (%s)",
FLAC__StreamDecoderStateString[FLAC__stream_encoder_get_verify_decoder_state(ctx->encoder.flac.stream)]);
}
else {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "cannot encode FLAC stream (%s)",
FLAC__StreamEncoderStateString[ctx->state.flac]);
}
flac_session_close();
return -1;
}
if (!FLAC__stream_encoder_process_interleaved(ctx->encoder.flac.stream, oggbuf,
nbytes / nch / 2 )) {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "cannot encode FLAC stream");
flac_session_close();
return -1;
}
}
#else /* !LEGACY_FLAC */
ctx->state.flac = FLAC__stream_encoder_get_state(ctx->encoder.flac.stream);
if (ctx->state.flac != FLAC__STREAM_ENCODER_OK) {
if (ctx->state.flac == FLAC__STREAM_ENCODER_VERIFY_DECODER_ERROR |
FLAC__STREAM_ENCODER_VERIFY_MISMATCH_IN_AUDIO_DATA) {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "FLAC stream verify error (%s)",
FLAC__StreamDecoderStateString[FLAC__stream_encoder_get_verify_decoder_state(ctx->encoder.flac.stream)]);
}
else {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "cannot encode FLAC stream (%s)",
FLAC__StreamEncoderStateString[ctx->state.flac]);
}
flac_session_close();
return -1;
}
if (!FLAC__stream_encoder_process_interleaved(ctx->encoder.flac.stream, oggbuf,
nbytes / nch / 2 )) {
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "cannot encode FLAC stream");
flac_session_close();
return -1;
}
#endif
ctx->in_bytes += nbytes;
free(oggbuf);
return 0;
}
int __stdcall
WWFlacRW_EncodeRun(int id, const wchar_t *path)
{
FILE *fp = NULL;
errno_t ercd;
int64_t left;
int64_t readPos;
int64_t writePos;
FLAC__bool ok = true;
FLAC__int32 *pcm = NULL;
if (NULL == path || wcslen(path) == 0) {
return FRT_BadParams;
}
FLAC__StreamEncoderInitStatus initStatus = FLAC__STREAM_ENCODER_INIT_STATUS_ENCODER_ERROR;
FlacEncodeInfo *fei = FlacTInfoFindById<FlacEncodeInfo>(g_flacEncodeInfoMap, id);
if (NULL == fei) {
return FRT_IdNotFound;
}
if (0 < fei->pictureBytes && fei->pictureData == NULL) {
dprintf("%s picture data is not set yet.\n", __FUNCTION__);
return FRT_DataNotReady;
}
assert(fei->buffPerChannel);
ok = FLAC__stream_encoder_set_metadata(fei->encoder, &fei->flacMetaArray[0], fei->flacMetaCount);
if(!ok) {
dprintf("FLAC__stream_encoder_set_metadata failed\n");
fei->errorCode = FRT_OtherError;
goto end;
}
for (int ch=0; ch<fei->channels; ++ch) {
if (fei->buffPerChannel[ch] == NULL){
dprintf("%s pcm buffer is not set yet.\n", __FUNCTION__);
return FRT_DataNotReady;
}
}
if (fei->bitsPerSample != 16 && fei->bitsPerSample != 24) {
return FRT_InvalidBitsPerSample;
}
pcm = new FLAC__int32[FLACENCODE_READFRAMES * fei->channels];
if (pcm == NULL) {
return FRT_MemoryExhausted;
}
// Windowsでは、この方法でファイルを開かなければならぬ。
wcsncpy_s(fei->path, path, (sizeof fei->path)/2-1);
ercd = _wfopen_s(&fp, fei->path, L"wb");
if (ercd != 0 || NULL == fp) {
fei->errorCode = FRT_FileOpenError;
goto end;
}
initStatus = FLAC__stream_encoder_init_FILE(fei->encoder, fp, ProgressCallback, fei);
if(initStatus != FLAC__STREAM_ENCODER_INIT_STATUS_OK) {
dprintf("FLAC__stream_encoder_init_FILE failed %s\n", FLAC__StreamEncoderInitStatusString[initStatus]);
switch (initStatus) {
case FLAC__STREAM_ENCODER_INIT_STATUS_ENCODER_ERROR:
{
FLAC__StreamDecoderState state = FLAC__stream_encoder_get_verify_decoder_state(fei->encoder);
dprintf("decoderState=%d\n", state);
}
fei->errorCode = FRT_EncoderError;
goto end;
case FLAC__STREAM_ENCODER_INIT_STATUS_INVALID_NUMBER_OF_CHANNELS:
fei->errorCode = FRT_InvalidNumberOfChannels;
goto end;
case FLAC__STREAM_ENCODER_INIT_STATUS_INVALID_BITS_PER_SAMPLE:
fei->errorCode = FRT_InvalidBitsPerSample;
goto end;
case FLAC__STREAM_ENCODER_INIT_STATUS_INVALID_SAMPLE_RATE:
fei->errorCode = FRT_InvalidSampleRate;
goto end;
case FLAC__STREAM_ENCODER_INIT_STATUS_INVALID_METADATA:
fei->errorCode = FRT_InvalidMetadata;
goto end;
case FLAC__STREAM_ENCODER_INIT_STATUS_INVALID_CALLBACKS:
case FLAC__STREAM_ENCODER_INIT_STATUS_ALREADY_INITIALIZED:
case FLAC__STREAM_ENCODER_INIT_STATUS_UNSUPPORTED_CONTAINER:
case FLAC__STREAM_ENCODER_INIT_STATUS_INVALID_BLOCK_SIZE:
case FLAC__STREAM_ENCODER_INIT_STATUS_INVALID_MAX_LPC_ORDER:
case FLAC__STREAM_ENCODER_INIT_STATUS_INVALID_QLP_COEFF_PRECISION:
case FLAC__STREAM_ENCODER_INIT_STATUS_BLOCK_SIZE_TOO_SMALL_FOR_LPC_ORDER:
case FLAC__STREAM_ENCODER_INIT_STATUS_NOT_STREAMABLE:
default:
fei->errorCode = FRT_OtherError;
goto end;
}
}
fp = NULL;
readPos = 0;
left = fei->totalSamples;
while(ok && left) {
uint32_t need = left>FLACENCODE_READFRAMES ? FLACENCODE_READFRAMES : (unsigned int)left;
// create interleaved PCM samples to pcm[]
writePos = 0;
switch (fei->bitsPerSample) {
case 16:
for (uint32_t i=0; i<need; ++i) {
for (int ch=0; ch<fei->channels;++ch) {
uint8_t *p = &fei->buffPerChannel[ch][readPos];
int v = (p[0]<<16) + (p[1]<<24);
pcm[writePos] = v>>16;
++writePos;
}
readPos += 2;
}
break;
case 24:
for (uint32_t i=0; i<need; ++i) {
for (int ch=0; ch<fei->channels;++ch) {
uint8_t *p = &fei->buffPerChannel[ch][readPos];
int v = (p[0]<<8) + (p[1]<<16) + (p[2]<<24);
pcm[writePos] = v >> 8;
++writePos;
}
readPos += 3;
}
break;
default:
assert(0);
break;
}
ok = FLAC__stream_encoder_process_interleaved(fei->encoder, pcm, need);
left -= need;
}
if (!ok) {
dprintf("FLAC__stream_encoder_process_interleaved failed");
fei->errorCode = FRT_EncoderProcessFailed;
}
end:
delete [] pcm;
pcm = NULL;
if (NULL != fp) {
fclose(fp);
fp = NULL;
}
if (NULL != fei->encoder) {
if (initStatus == FLAC__STREAM_ENCODER_INIT_STATUS_OK) {
FLAC__stream_encoder_finish(fei->encoder);
}
DeleteFlacMetaArray(fei);
FLAC__stream_encoder_delete(fei->encoder);
fei->encoder = NULL;
}
if (fei->errorCode < 0) {
int result = fei->errorCode;
FlacTInfoDelete<FlacEncodeInfo>(g_flacEncodeInfoMap, fei);
fei = NULL;
return result;
}
return fei->id;
}
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);
}
FLAC__bool file_utils__generate_flacfile(FLAC__bool is_ogg, const char *output_filename, off_t *output_filesize, unsigned length, const FLAC__StreamMetadata *streaminfo, FLAC__StreamMetadata **metadata, unsigned num_metadata)
{
FLAC__int32 samples[1024];
FLAC__StreamEncoder *encoder;
FLAC__StreamEncoderInitStatus init_status;
encoder_client_struct encoder_client_data;
unsigned i, n;
FLAC__ASSERT(0 != output_filename);
FLAC__ASSERT(0 != streaminfo);
FLAC__ASSERT(streaminfo->type == FLAC__METADATA_TYPE_STREAMINFO);
FLAC__ASSERT((streaminfo->is_last && num_metadata == 0) || (!streaminfo->is_last && num_metadata > 0));
if(0 == (encoder_client_data.file = fopen(output_filename, "wb")))
return false;
encoder = FLAC__stream_encoder_new();
if(0 == encoder) {
fclose(encoder_client_data.file);
return false;
}
FLAC__stream_encoder_set_ogg_serial_number(encoder, file_utils__ogg_serial_number);
FLAC__stream_encoder_set_verify(encoder, true);
FLAC__stream_encoder_set_streamable_subset(encoder, true);
FLAC__stream_encoder_set_do_mid_side_stereo(encoder, false);
FLAC__stream_encoder_set_loose_mid_side_stereo(encoder, false);
FLAC__stream_encoder_set_channels(encoder, streaminfo->data.stream_info.channels);
FLAC__stream_encoder_set_bits_per_sample(encoder, streaminfo->data.stream_info.bits_per_sample);
FLAC__stream_encoder_set_sample_rate(encoder, streaminfo->data.stream_info.sample_rate);
FLAC__stream_encoder_set_blocksize(encoder, streaminfo->data.stream_info.min_blocksize);
FLAC__stream_encoder_set_max_lpc_order(encoder, 0);
FLAC__stream_encoder_set_qlp_coeff_precision(encoder, 0);
FLAC__stream_encoder_set_do_qlp_coeff_prec_search(encoder, false);
FLAC__stream_encoder_set_do_escape_coding(encoder, false);
FLAC__stream_encoder_set_do_exhaustive_model_search(encoder, false);
FLAC__stream_encoder_set_min_residual_partition_order(encoder, 0);
FLAC__stream_encoder_set_max_residual_partition_order(encoder, 0);
FLAC__stream_encoder_set_rice_parameter_search_dist(encoder, 0);
FLAC__stream_encoder_set_total_samples_estimate(encoder, streaminfo->data.stream_info.total_samples);
FLAC__stream_encoder_set_metadata(encoder, metadata, num_metadata);
if(is_ogg)
init_status = FLAC__stream_encoder_init_ogg_stream(encoder, /*read_callback=*/0, encoder_write_callback_, /*seek_callback=*/0, /*tell_callback=*/0, encoder_metadata_callback_, &encoder_client_data);
else
init_status = FLAC__stream_encoder_init_stream(encoder, encoder_write_callback_, /*seek_callback=*/0, /*tell_callback=*/0, encoder_metadata_callback_, &encoder_client_data);
if(init_status != FLAC__STREAM_ENCODER_INIT_STATUS_OK) {
fclose(encoder_client_data.file);
return false;
}
/* init the dummy sample buffer */
for(i = 0; i < sizeof(samples) / sizeof(FLAC__int32); i++)
samples[i] = i & 7;
while(length > 0) {
n = min(length, sizeof(samples) / sizeof(FLAC__int32));
if(!FLAC__stream_encoder_process_interleaved(encoder, samples, n)) {
fclose(encoder_client_data.file);
return false;
}
length -= n;
}
(void)FLAC__stream_encoder_finish(encoder);
fclose(encoder_client_data.file);
FLAC__stream_encoder_delete(encoder);
if(0 != output_filesize) {
struct stat filestats;
if(stat(output_filename, &filestats) != 0)
return false;
else
*output_filesize = filestats.st_size;
}
return true;
}
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;
}
static int start_write(sox_format_t * const ft)
{
priv_t * p = (priv_t *)ft->priv;
FLAC__StreamEncoderState status;
unsigned compression_level = MAX_COMPRESSION; /* Default to "best" */
if (ft->encoding.compression != HUGE_VAL) {
compression_level = ft->encoding.compression;
if (compression_level != ft->encoding.compression ||
compression_level > MAX_COMPRESSION) {
lsx_fail_errno(ft, SOX_EINVAL,
"FLAC compression level must be a whole number from 0 to %i",
MAX_COMPRESSION);
return SOX_EOF;
}
}
p->encoder = FLAC__stream_encoder_new();
if (p->encoder == NULL) {
lsx_fail_errno(ft, SOX_ENOMEM, "FLAC ERROR creating the encoder instance");
return SOX_EOF;
}
p->decoded_samples = lsx_malloc(sox_globals.bufsiz * sizeof(FLAC__int32));
p->bits_per_sample = ft->encoding.bits_per_sample;
lsx_report("encoding at %i bits per sample", p->bits_per_sample);
FLAC__stream_encoder_set_channels(p->encoder, ft->signal.channels);
FLAC__stream_encoder_set_bits_per_sample(p->encoder, p->bits_per_sample);
FLAC__stream_encoder_set_sample_rate(p->encoder, (unsigned)(ft->signal.rate + .5));
{ /* Check if rate is streamable: */
static const unsigned streamable_rates[] =
{8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000};
size_t i;
sox_bool streamable = sox_false;
for (i = 0; !streamable && i < array_length(streamable_rates); ++i)
streamable = (streamable_rates[i] == ft->signal.rate);
if (!streamable) {
lsx_report("non-standard rate; output may not be streamable");
FLAC__stream_encoder_set_streamable_subset(p->encoder, sox_false);
}
}
#if FLAC_API_VERSION_CURRENT >= 10
FLAC__stream_encoder_set_compression_level(p->encoder, compression_level);
#else
{
static struct {
unsigned blocksize;
FLAC__bool do_exhaustive_model_search;
FLAC__bool do_mid_side_stereo;
FLAC__bool loose_mid_side_stereo;
unsigned max_lpc_order;
unsigned max_residual_partition_order;
unsigned min_residual_partition_order;
} const options[MAX_COMPRESSION + 1] = {
{1152, sox_false, sox_false, sox_false, 0, 2, 2},
{1152, sox_false, sox_true, sox_true, 0, 2, 2},
{1152, sox_false, sox_true, sox_false, 0, 3, 0},
{4608, sox_false, sox_false, sox_false, 6, 3, 3},
{4608, sox_false, sox_true, sox_true, 8, 3, 3},
{4608, sox_false, sox_true, sox_false, 8, 3, 3},
{4608, sox_false, sox_true, sox_false, 8, 4, 0},
{4608, sox_true, sox_true, sox_false, 8, 6, 0},
{4608, sox_true, sox_true, sox_false, 12, 6, 0},
};
#define SET_OPTION(x) do {\
lsx_report(#x" = %i", options[compression_level].x); \
FLAC__stream_encoder_set_##x(p->encoder, options[compression_level].x);\
} while (0)
SET_OPTION(blocksize);
SET_OPTION(do_exhaustive_model_search);
SET_OPTION(max_lpc_order);
SET_OPTION(max_residual_partition_order);
SET_OPTION(min_residual_partition_order);
if (ft->signal.channels == 2) {
SET_OPTION(do_mid_side_stereo);
SET_OPTION(loose_mid_side_stereo);
}
#undef SET_OPTION
}
#endif
if (ft->signal.length != 0) {
FLAC__stream_encoder_set_total_samples_estimate(p->encoder, (FLAC__uint64)(ft->signal.length / ft->signal.channels));
p->metadata[p->num_metadata] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_SEEKTABLE);
if (p->metadata[p->num_metadata] == NULL) {
lsx_fail_errno(ft, SOX_ENOMEM, "FLAC ERROR creating the encoder seek table template");
return SOX_EOF;
}
{
#if FLAC_API_VERSION_CURRENT >= 8
if (!FLAC__metadata_object_seektable_template_append_spaced_points_by_samples(p->metadata[p->num_metadata], (unsigned)(10 * ft->signal.rate + .5), (FLAC__uint64)(ft->signal.length/ft->signal.channels))) {
#else
size_t samples = 10 * ft->signal.rate;
size_t total_samples = ft->signal.length/ft->signal.channels;
if (!FLAC__metadata_object_seektable_template_append_spaced_points(p->metadata[p->num_metadata], total_samples / samples + (total_samples % samples != 0), (FLAC__uint64)total_samples)) {
#endif
lsx_fail_errno(ft, SOX_ENOMEM, "FLAC ERROR creating the encoder seek table points");
return SOX_EOF;
}
}
p->metadata[p->num_metadata]->is_last = sox_false; /* the encoder will set this for us */
++p->num_metadata;
}
if (ft->oob.comments) { /* Make the comment structure */
FLAC__StreamMetadata_VorbisComment_Entry entry;
int i;
p->metadata[p->num_metadata] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_VORBIS_COMMENT);
for (i = 0; ft->oob.comments[i]; ++i) {
static const char prepend[] = "Comment=";
char * text = lsx_calloc(strlen(prepend) + strlen(ft->oob.comments[i]) + 1, sizeof(*text));
/* Prepend `Comment=' if no field-name already in the comment */
if (!strchr(ft->oob.comments[i], '='))
strcpy(text, prepend);
entry.entry = (FLAC__byte *) strcat(text, ft->oob.comments[i]);
entry.length = strlen(text);
FLAC__metadata_object_vorbiscomment_append_comment(p->metadata[p->num_metadata], entry, /*copy= */ sox_true);
free(text);
}
++p->num_metadata;
}
if (p->num_metadata)
FLAC__stream_encoder_set_metadata(p->encoder, p->metadata, p->num_metadata);
#if FLAC_API_VERSION_CURRENT <= 7
FLAC__stream_encoder_set_write_callback(p->encoder, flac_stream_encoder_write_callback);
FLAC__stream_encoder_set_metadata_callback(p->encoder, flac_stream_encoder_metadata_callback);
FLAC__stream_encoder_set_client_data(p->encoder, ft);
status = FLAC__stream_encoder_init(p->encoder);
#else
status = FLAC__stream_encoder_init_stream(p->encoder, flac_stream_encoder_write_callback,
flac_stream_encoder_seek_callback, flac_stream_encoder_tell_callback, flac_stream_encoder_metadata_callback, ft);
#endif
if (status != FLAC__STREAM_ENCODER_OK) {
lsx_fail_errno(ft, SOX_EINVAL, "%s", FLAC__StreamEncoderStateString[status]);
return SOX_EOF;
}
return SOX_SUCCESS;
}
static size_t write_samples(sox_format_t * const ft, sox_sample_t const * const sampleBuffer, size_t const len)
{
priv_t * p = (priv_t *)ft->priv;
unsigned i;
for (i = 0; i < len; ++i) {
long pcm = SOX_SAMPLE_TO_SIGNED_32BIT(sampleBuffer[i], ft->clips);
p->decoded_samples[i] = pcm >> (32 - p->bits_per_sample);
switch (p->bits_per_sample) {
case 8: p->decoded_samples[i] =
SOX_SAMPLE_TO_SIGNED_8BIT(sampleBuffer[i], ft->clips);
break;
case 16: p->decoded_samples[i] =
SOX_SAMPLE_TO_SIGNED_16BIT(sampleBuffer[i], ft->clips);
break;
case 24: p->decoded_samples[i] = /* sign extension: */
SOX_SAMPLE_TO_SIGNED_24BIT(sampleBuffer[i],ft->clips) << 8;
p->decoded_samples[i] >>= 8;
break;
case 32: p->decoded_samples[i] =
SOX_SAMPLE_TO_SIGNED_32BIT(sampleBuffer[i],ft->clips);
break;
}
}
FLAC__stream_encoder_process_interleaved(p->encoder, p->decoded_samples, (unsigned) len / ft->signal.channels);
return FLAC__stream_encoder_get_state(p->encoder) == FLAC__STREAM_ENCODER_OK ? len : 0;
}
