예제 #1
0
static int flac_session_close()
{
  FLAC_ctx *ctx = flac_ctx;

  if (dpm.fd > 0) {
    close(dpm.fd);
  }
  dpm.fd = -1;

  if (ctx != NULL) {
    if (flac_options.isogg) {
      if (ctx->encoder.ogg.stream) {
	FLAC__stream_encoder_finish(ctx->encoder.ogg.stream);
	FLAC__stream_encoder_delete(ctx->encoder.ogg.stream);
      }
    }
    else
    if (flac_options.seekable) {
      if (ctx->encoder.flac.s_stream) {
	FLAC__stream_encoder_finish(ctx->encoder.flac.s_stream);
	FLAC__stream_encoder_delete(ctx->encoder.flac.s_stream);
      }
    }
    else
    {
      if (ctx->encoder.flac.stream) {
	FLAC__stream_encoder_finish(ctx->encoder.flac.stream);
	FLAC__stream_encoder_delete(ctx->encoder.flac.stream);
      }
    }
    free(ctx);
    flac_ctx = NULL;
  }
  return 0;
}
예제 #2
0
	void FinishStream()
	{
		if(inited)
		{
			if(!started)
			{
				StartStream();
			}
			ASSERT(inited && started);

			FLAC__stream_encoder_finish(encoder);
			FLAC__stream_encoder_delete(encoder);
			encoder = nullptr;

			if(flac_metadata[0])
			{
				FLAC__metadata_object_delete(flac_metadata[0]);
				flac_metadata[0] = nullptr;
			}

			started = false;
			inited = false;
		}
		ASSERT(!inited && !started);
	}
예제 #3
0
ACFLACEncoder::~ACFLACEncoder()
{
	if (mEncoder != NULL)
	{
		FLAC__stream_encoder_delete(mEncoder);
		mEncoder = NULL;
	}
}
SoftFlacEncoder::~SoftFlacEncoder() {
    ALOGV("SoftFlacEncoder::~SoftFlacEncoder()");
    if (mFlacStreamEncoder != NULL) {
        FLAC__stream_encoder_delete(mFlacStreamEncoder);
        mFlacStreamEncoder = NULL;
    }
    free(mInputBufferPcm32);
    mInputBufferPcm32 = NULL;
}
예제 #5
0
파일: flac_a.c 프로젝트: ranvis/tina
static void flac_session_close()
{
  FLAC_ctx *ctx = flac_ctx;

  if (dpm.fd > 0) {
    close(dpm.fd);
  }
  dpm.fd = -1;

  if (ctx != NULL) {
#ifdef LEGACY_FLAC
#ifdef AU_OGGFLAC
    if (flac_options.isogg) {
      if (ctx->encoder.ogg.stream) {
	OggFLAC__stream_encoder_finish(ctx->encoder.ogg.stream);
	OggFLAC__stream_encoder_delete(ctx->encoder.ogg.stream);
      }
    }
    else
#endif /* AU_OGGFLAC */
    if (flac_options.seekable) {
      if (ctx->encoder.flac.s_stream) {
	FLAC__seekable_stream_encoder_finish(ctx->encoder.flac.s_stream);
	FLAC__seekable_stream_encoder_delete(ctx->encoder.flac.s_stream);
      }
    }
    else
    {
      if (ctx->encoder.flac.stream) {
	FLAC__stream_encoder_finish(ctx->encoder.flac.stream);
	FLAC__stream_encoder_delete(ctx->encoder.flac.stream);
      }
    }
#else
    if (ctx->encoder.flac.stream) {
      FLAC__stream_encoder_finish(ctx->encoder.flac.stream);
      FLAC__stream_encoder_delete(ctx->encoder.flac.stream);
    }
#endif
    free(ctx);
    flac_ctx = NULL;
  }
}
예제 #6
0
static void
flac_encoder_close(struct encoder *_encoder)
{
	struct flac_encoder *encoder = (struct flac_encoder *)_encoder;

	FLAC__stream_encoder_delete(encoder->fse);

	pcm_buffer_deinit(&encoder->buffer);
	pcm_buffer_deinit(&encoder->expand_buffer);
}
예제 #7
0
void SoundFileWriterFlac::close()
{
    if (m_encoder)
    {
        // Close the output stream
        FLAC__stream_encoder_finish(m_encoder);

        // Destroy the encoder
        FLAC__stream_encoder_delete(m_encoder);
        m_encoder = NULL;
    }
}
예제 #8
0
CEncoderFlac::~CEncoderFlac()
{
  // free the metadata
  if (m_metadata[0])
    FLAC__metadata_object_delete(m_metadata[0]);
  if (m_metadata[1])
    FLAC__metadata_object_delete(m_metadata[1]);

  // free the encoder
  if (m_encoder)
    FLAC__stream_encoder_delete(m_encoder);
}
예제 #9
0
OggFLAC_API void OggFLAC__seekable_stream_encoder_delete(OggFLAC__SeekableStreamEncoder *encoder)
{
	FLAC__ASSERT(0 != encoder);
	FLAC__ASSERT(0 != encoder->protected_);
	FLAC__ASSERT(0 != encoder->private_);
	FLAC__ASSERT(0 != encoder->private_->FLAC_stream_encoder);

	(void)OggFLAC__seekable_stream_encoder_finish(encoder);

	FLAC__stream_encoder_delete(encoder->private_->FLAC_stream_encoder);

	free(encoder->private_);
	free(encoder->protected_);
	free(encoder);
}
예제 #10
0
void FlacAudioEncoder::tear_down()
{
   if (not FLAC__stream_encoder_finish(_encoder))
   {
      THROW_EXCEPTION(AudioEncoderException, "ERROR: could not finish encoding");
   }

   FLAC__metadata_object_delete(_metadata[0]);
   FLAC__metadata_object_delete(_metadata[1]);

   if (_encoder)
   {
      FLAC__stream_encoder_delete(_encoder);
      _encoder = nullptr;
   }
}
예제 #11
0
int __stdcall
WWFlacRW_EncodeEnd(int id)
{
    FlacEncodeInfo *fei = FlacTInfoFindById<FlacEncodeInfo>(g_flacEncodeInfoMap, id);
    if (NULL == fei) {
        return FRT_IdNotFound;
    }

    if (NULL != fei->encoder) {
        FLAC__stream_encoder_delete(fei->encoder);
        fei->encoder = NULL;
    }
    FlacTInfoDelete<FlacEncodeInfo>(g_flacEncodeInfoMap, fei);

    return FRT_Success;
}
예제 #12
0
파일: flac.c 프로젝트: Emisense/eTracks
static int stop_write(sox_format_t * const ft)
{
  priv_t * p = (priv_t *)ft->priv;
  FLAC__StreamEncoderState state = FLAC__stream_encoder_get_state(p->encoder);
  unsigned i;

  FLAC__stream_encoder_finish(p->encoder);
  FLAC__stream_encoder_delete(p->encoder);
  for (i = 0; i < p->num_metadata; ++i)
    FLAC__metadata_object_delete(p->metadata[i]);
  free(p->decoded_samples);
  if (state != FLAC__STREAM_ENCODER_OK) {
    lsx_fail_errno(ft, SOX_EINVAL, "FLAC ERROR: failed to encode to end of stream");
    return SOX_EOF;
  }
  return SOX_SUCCESS;
}
예제 #13
0
void Free(void *ctx)
{
  flac_context *context = (flac_context*)ctx;
  if (context)
  {
    // free the metadata
    if (context->metadata[0])
      FLAC__metadata_object_delete(context->metadata[0]);
    if (context->metadata[1])
      FLAC__metadata_object_delete(context->metadata[1]);

    // free the encoder
    if (context->encoder)
      FLAC__stream_encoder_delete(context->encoder);

    delete context;
  }
}
예제 #14
0
int WFLACEncoder::Uninitialize()
{
	if(c_fReady)
	{
		FLAC__stream_encoder_finish(c_encoder);
		FLAC__stream_encoder_delete(c_encoder);
		c_encoder = NULL;

	}

	if(c_pcm)
		free(c_pcm);

	c_pcm = NULL;
	c_pcm_size = 0;

	c_fReady = 0;
	return 1;
}
예제 #15
0
static bool DecodeFLAC(FILE *f, FILE *of)
{
	fseek(f, 0, SEEK_SET);
	FLAC__StreamDecoder *decoder = FLAC__stream_decoder_new();
	if(decoder == nullptr)
	{
		return false;
	}

	FLAC__stream_decoder_set_metadata_respond_all(decoder);

	FLACClientData client;
	client.of = of;
	client.started = false;
	client.encoder = FLAC__stream_encoder_new();

	FLAC__StreamDecoderInitStatus initStatus = FLAC__stream_decoder_init_FILE(decoder, f, write_cb, metadata_cb, error_cb, &client);

	if(initStatus != FLAC__STREAM_DECODER_INIT_STATUS_OK || client.encoder == nullptr)
	{
		FLAC__stream_decoder_delete(decoder);
		return false;
	}

	FLAC__stream_decoder_process_until_end_of_stream(decoder);
	FLAC__stream_decoder_finish(decoder);
	FLAC__stream_decoder_delete(decoder);
	FLAC__stream_encoder_finish(client.encoder);
	FLAC__stream_encoder_delete(client.encoder);

	for(auto m = client.metadata.begin(); m != client.metadata.end(); m++)
	{
		FLAC__metadata_object_delete(*m);
	}

	return client.started;
}
예제 #16
0
파일: main.c 프로젝트: kode54/flac
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;
}
예제 #17
0
int WFLACEncoder::Initialize(unsigned int nSampleRate, unsigned int nNumberOfChannels, unsigned int nBitPerSample,
			unsigned int custom_value,
			TEncoderReadCallback read_callback,
			TEncoderWriteCallback write_callback,
			TEncoderSeekCallback seek_callback,
			TEncoderTellCallback tell_callback)
{


	c_nSampleRate = nSampleRate;
	c_nNumberOfChannels = nNumberOfChannels;
	c_nBitBerSample = nBitPerSample;


	c_read_calllback = read_callback;
	c_write_callback = write_callback;
	c_seek_callback = seek_callback;
	c_tell_callback = tell_callback;

	c_user_data = (void*) custom_value;


	if((c_encoder = FLAC__stream_encoder_new()) == NULL)
	{
		err(ENCODER_INITIALIZATION_ERROR);
		return 0;
	}


	FLAC__bool ok = true;

	ok &= FLAC__stream_encoder_set_verify(c_encoder, false);
	ok &= FLAC__stream_encoder_set_compression_level(c_encoder, 5);
	ok &= FLAC__stream_encoder_set_channels(c_encoder, c_nNumberOfChannels);
	ok &= FLAC__stream_encoder_set_bits_per_sample(c_encoder, c_nBitBerSample);
	ok &= FLAC__stream_encoder_set_sample_rate(c_encoder, c_nSampleRate);
	ok &= FLAC__stream_encoder_set_total_samples_estimate(c_encoder, 0);

	if(!ok)
	{
		FLAC__stream_encoder_delete(c_encoder);
		c_encoder = NULL;
		err(ENCODER_INITIALIZATION_ERROR);
		return 0;
	}


	if(c_fOgg)
	{
		if(FLAC__stream_encoder_init_ogg_stream(c_encoder, f_read_callback, f_write_callback, f_seek_callback, f_tell_callback, NULL, this) != FLAC__STREAM_ENCODER_INIT_STATUS_OK)
		{
			FLAC__stream_encoder_delete(c_encoder);
			c_encoder = NULL;
			err(ENCODER_FILEOPEN_ERROR);
			return 0;			
		}

	}
	else
	{
		if(FLAC__stream_encoder_init_stream(c_encoder, f_write_callback, f_seek_callback, f_tell_callback, NULL, this) != FLAC__STREAM_ENCODER_INIT_STATUS_OK)
		{
			FLAC__stream_encoder_delete(c_encoder);
			c_encoder = NULL;
			err(ENCODER_FILEOPEN_ERROR);
			return 0;			
		}
	}

	c_fReady = 1;
	return 1;
}
예제 #18
0
파일: SampleBuffer.cpp 프로젝트: floft/lmms
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;
}
예제 #19
0
int __stdcall
WWFlacRW_EncodeInit(const WWFlacMetadata &meta)
{
    FLAC__bool                    ok = true;
    FLAC__StreamMetadata_VorbisComment_Entry entry;

    FlacEncodeInfo *fei = FlacTInfoNew<FlacEncodeInfo>(g_flacEncodeInfoMap);
    if (NULL == fei) {
        return FRT_OtherError;
    }
    
    fei->errorCode = FRT_Success;

    fei->sampleRate = meta.sampleRate;
    fei->channels = meta.channels;
    fei->bitsPerSample = meta.bitsPerSample;
    fei->totalSamples = meta.totalSamples;
    fei->totalBytesPerChannel = meta.totalSamples * fei->bitsPerSample/8;
    fei->pictureBytes = meta.pictureBytes;

    assert(NULL == fei->buffPerChannel);
    fei->buffPerChannel = new uint8_t*[fei->channels];
    if (NULL == fei->buffPerChannel) {
        return FRT_MemoryExhausted;
    }
    memset(fei->buffPerChannel, 0, sizeof(uint8_t*)*fei->channels);
    
    WCTOUTF8(titleStr);
    WCTOUTF8(artistStr);
    WCTOUTF8(albumStr);
    WCTOUTF8(albumArtistStr);
    WCTOUTF8(genreStr);

    WCTOUTF8(dateStr);
    WCTOUTF8(trackNumberStr);
    WCTOUTF8(discNumberStr);
    WCTOUTF8(pictureMimeTypeStr);
    WCTOUTF8(pictureDescriptionStr);

    if((fei->encoder = FLAC__stream_encoder_new()) == NULL) {
        dprintf("FLAC__stream_encoder_new failed\n");
        fei->errorCode = FRT_OtherError;
        goto end;
    }

    ok &= FLAC__stream_encoder_set_verify(fei->encoder, true);
    ok &= FLAC__stream_encoder_set_compression_level(fei->encoder, 5);
    ok &= FLAC__stream_encoder_set_channels(fei->encoder, fei->channels);
    ok &= FLAC__stream_encoder_set_bits_per_sample(fei->encoder, fei->bitsPerSample);
    ok &= FLAC__stream_encoder_set_sample_rate(fei->encoder, fei->sampleRate);
    ok &= FLAC__stream_encoder_set_total_samples_estimate(fei->encoder, fei->totalSamples);

    if(!ok) {
        dprintf("FLAC__stream_encoder_set_??? failed\n");
        fei->errorCode = FRT_OtherError;
        goto end;
    }

    if((fei->flacMetaArray[FMT_VorbisComment] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_VORBIS_COMMENT)) == NULL) {
        dprintf("FLAC__metadata_object_new vorbis comment failed\n");
        fei->errorCode = FRT_OtherError;
        goto end;
    }
    if((fei->flacMetaArray[FMT_Picture] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_PICTURE)) == NULL) {
        dprintf("FLAC__metadata_object_new picture failed\n");
        fei->errorCode = FRT_OtherError;
        goto end;
    }

    fei->flacMetaCount = 1;

    ADD_TAG(titleStr,       "TITLE");
    ADD_TAG(artistStr,      "ARTIST");
    ADD_TAG(albumStr,       "ALBUM");
    ADD_TAG(albumArtistStr, "ALBUMARTIST");
    ADD_TAG(genreStr,       "GENRE");

    ADD_TAG(dateStr,        "DATE");
    ADD_TAG(trackNumberStr, "TRACKNUMBER");
    ADD_TAG(discNumberStr,  "DISCNUMBER");

end:
    if (fei->errorCode < 0) {
        if (NULL != fei->encoder) {
            FLAC__stream_encoder_delete(fei->encoder);
            fei->encoder = NULL;
        }

        DeleteFlacMetaArray(fei);

        int result = fei->errorCode;
        FlacTInfoDelete<FlacEncodeInfo>(g_flacEncodeInfoMap, fei);
        fei = NULL;

        return result;
    }

    return fei->id;
}
예제 #20
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;
}
예제 #21
0
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);
}
예제 #22
0
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;
}
예제 #23
0
static bool
flac_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
		     GError **error)
{
	struct flac_encoder *encoder = (struct flac_encoder *)_encoder;
	unsigned bits_per_sample;

	encoder->audio_format = *audio_format;

	/* FIXME: flac should support 32bit as well */
	switch (audio_format->format) {
	case SAMPLE_FORMAT_S8:
		bits_per_sample = 8;
		break;

	case SAMPLE_FORMAT_S16:
		bits_per_sample = 16;
		break;

	case SAMPLE_FORMAT_S24_P32:
		bits_per_sample = 24;
		break;

	default:
		bits_per_sample = 24;
		audio_format->format = SAMPLE_FORMAT_S24_P32;
	}

	/* allocate the encoder */
	encoder->fse = FLAC__stream_encoder_new();
	if (encoder->fse == NULL) {
		g_set_error(error, flac_encoder_quark(), 0,
			    "flac_new() failed");
		return false;
	}

	if (!flac_encoder_setup(encoder, bits_per_sample, error)) {
		FLAC__stream_encoder_delete(encoder->fse);
		return false;
	}

	encoder->buffer_length = 0;
	pcm_buffer_init(&encoder->buffer);
	pcm_buffer_init(&encoder->expand_buffer);

	/* this immediately outputs data through callback */

#if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7
	{
		FLAC__StreamEncoderState init_status;

		FLAC__stream_encoder_set_write_callback(encoder->fse,
					    flac_write_callback);

		init_status = FLAC__stream_encoder_init(encoder->fse);

		if (init_status != FLAC__STREAM_ENCODER_OK) {
			g_set_error(error, flac_encoder_quark(), 0,
			    "failed to initialize encoder: %s\n",
			    FLAC__StreamEncoderStateString[init_status]);
			flac_encoder_close(_encoder);
			return false;
		}
	}
#else
	{
		FLAC__StreamEncoderInitStatus init_status;

		init_status = FLAC__stream_encoder_init_stream(encoder->fse,
			    flac_write_callback,
			    NULL, NULL, NULL, encoder);

		if(init_status != FLAC__STREAM_ENCODER_INIT_STATUS_OK) {
			g_set_error(error, flac_encoder_quark(), 0,
			    "failed to initialize encoder: %s\n",
			    FLAC__StreamEncoderInitStatusString[init_status]);
			flac_encoder_close(_encoder);
			return false;
		}
	}
#endif

	return true;
}
예제 #24
0
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;
}
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;


    
}
예제 #26
0
static void flac_close(void)
{
    FLAC__stream_encoder_finish(flac_encoder);
    FLAC__stream_encoder_delete(flac_encoder);
}
예제 #27
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
}