예제 #1
0
static Tuple *
wv_probe_for_tuple(const char * filename, VFSFile * fd)
{
    WavpackContext *ctx;
    Tuple *tu;
    char error[1024];

    ctx = WavpackOpenFileInputEx(&wv_readers, fd, NULL, error, OPEN_TAGS, 0);

    if (ctx == NULL)
        return NULL;

    AUDDBG("starting probe of %p\n", (void *) fd);

    vfs_rewind(fd);
    tu = tuple_new_from_filename(filename);

    vfs_rewind(fd);
    tag_tuple_read(tu, fd);

    tuple_set_int(tu, FIELD_LENGTH, NULL,
        ((uint64_t) WavpackGetNumSamples(ctx) * 1000) / (uint64_t) WavpackGetSampleRate(ctx));
    tuple_set_str(tu, FIELD_CODEC, NULL, "WavPack");

    char * quality = wv_get_quality (ctx);
    tuple_set_str (tu, FIELD_QUALITY, NULL, quality);
    str_unref (quality);

    WavpackCloseFile(ctx);

    AUDDBG("returning tuple %p for file %p\n", (void *) tu, (void *) fd);
    return tu;
}
예제 #2
0
Result SoundSourceWV::tryOpen(const AudioSourceConfig& audioSrcCfg) {
    DEBUG_ASSERT(!m_wpc);
    char msg[80]; // hold possible error message
    int openFlags = OPEN_WVC | OPEN_NORMALIZE;
    if ((kChannelCountMono == audioSrcCfg.channelCountHint) ||
            (kChannelCountStereo == audioSrcCfg.channelCountHint)) {
        openFlags |= OPEN_2CH_MAX;
    }
    m_wpc = WavpackOpenFileInput(
            getLocalFileNameBytes().constData(), msg, openFlags, 0);
    if (!m_wpc) {
        qDebug() << "SSWV::open: failed to open file : " << msg;
        return ERR;
    }

    setChannelCount(WavpackGetReducedChannels(m_wpc));
    setFrameRate(WavpackGetSampleRate(m_wpc));
    setFrameCount(WavpackGetNumSamples(m_wpc));

    if (WavpackGetMode(m_wpc) & MODE_FLOAT) {
        m_sampleScaleFactor = CSAMPLE_PEAK;
    } else {
        const int bitsPerSample = WavpackGetBitsPerSample(m_wpc);
        const uint32_t wavpackPeakSampleValue = uint32_t(1)
                << (bitsPerSample - 1);
        m_sampleScaleFactor = CSAMPLE_PEAK / CSAMPLE(wavpackPeakSampleValue);
    }

    return OK;
}
예제 #3
0
soundfile_t *
soundfile_open_read(const char *path) {
	dp(30, "path=%s \n", path);
	soundfile_t *s = salloc(sizeof *s);
	s->m = sft_read;
	if 	(g_regex_match_simple ("\\.wv$", path, 0, 0)) {
		char error[80] = {0};
	    int flags = 0;
	    int norm_offset = 0;
		s->t = sft_wavpack;
	    s->p = WavpackOpenFileInput(path, error, flags, norm_offset);
		if (!s->p)
			die("can not open input file '%s'", path);
		s->bits_per_sample = WavpackGetBitsPerSample(s->p);
		s->channels = WavpackGetNumChannels(s->p);
		s->samplerate = WavpackGetSampleRate(s->p);
		s->frames = WavpackGetNumSamples(s->p);
	} else {
		SF_INFO	 	infile_info = {0};
		if (strcmp(path, "-"))
			s->p = sf_open(path, SFM_READ, &infile_info);
		else
			s->p = sf_open_fd(0, SFM_READ, &infile_info, 0);
		if (!s->p)
			die("can not open input file '%s'", path);
		s->t = sft_libsndfile;
		s->channels = infile_info.channels;
		s->samplerate = infile_info.samplerate;
		s->frames = infile_info.frames;
	}
	return s;
}
예제 #4
0
SoundSource::OpenResult SoundSourceWV::tryOpen(
        OpenMode /*mode*/,
        const OpenParams& params) {
    DEBUG_ASSERT(!m_wpc);
    char msg[80]; // hold possible error message
    int openFlags = OPEN_WVC | OPEN_NORMALIZE;
    if ((params.channelCount() == 1) ||
            (params.channelCount() == 2)) {
        openFlags |= OPEN_2CH_MAX;
    }

    // We use WavpackOpenFileInputEx to support Unicode paths on windows
    // http://www.wavpack.com/lib_use.txt
    QString wavPackFileName = getLocalFileName();
    m_pWVFile = new QFile(wavPackFileName);
    m_pWVFile->open(QFile::ReadOnly);
    QString correctionFileName(wavPackFileName + "c");
    if (QFile::exists(correctionFileName)) {
        // If there is a correction file, open it as well
        m_pWVCFile = new QFile(correctionFileName);
        m_pWVCFile->open(QFile::ReadOnly);
    }
    m_wpc = WavpackOpenFileInputEx(&s_streamReader, m_pWVFile, m_pWVCFile,
            msg, openFlags, 0);
    if (!m_wpc) {
        kLogger.warning() << "failed to open file : " << msg;
        return OpenResult::Failed;
    }

    setChannelCount(WavpackGetReducedChannels(m_wpc));
    setSampleRate(WavpackGetSampleRate(m_wpc));
    initFrameIndexRangeOnce(
            mixxx::IndexRange::forward(
                    0,
                    WavpackGetNumSamples(m_wpc)));

    if (WavpackGetMode(m_wpc) & MODE_FLOAT) {
        m_sampleScaleFactor = CSAMPLE_PEAK;
    } else {
        const int bitsPerSample = WavpackGetBitsPerSample(m_wpc);
        if ((bitsPerSample >= 8) && (bitsPerSample <= 32)) {
            // Range of signed sample values: [-2 ^ (bitsPerSample - 1), 2 ^ (bitsPerSample - 1) - 1]
            const uint32_t absSamplePeak = 1u << (bitsPerSample - 1);
            DEBUG_ASSERT(absSamplePeak > 0);
            // Scaled range of sample values: [-CSAMPLE_PEAK, CSAMPLE_PEAK)
            m_sampleScaleFactor = CSAMPLE_PEAK / absSamplePeak;
        } else {
            kLogger.warning()
                    << "Invalid bits per sample:"
                    << bitsPerSample;
            return OpenResult::Aborted;
        }
    }

    m_curFrameIndex = frameIndexMin();

    return OpenResult::Succeeded;
}
예제 #5
0
static void wav_info (const char *file_name, struct file_tags *info,
		const int tags_sel)
{
	char wv_error[100];
	char *tag;
	int tag_len;

	WavpackContext *wpc;

	wpc = WavpackOpenFileInput (file_name, wv_error, OPEN_TAGS, 0);

	if (wpc == NULL) {
		logit ("wv_open error: %s", wv_error);
		return;
	}

	int duration = WavpackGetNumSamples (wpc) / WavpackGetSampleRate (wpc);

	if(tags_sel & TAGS_TIME) {
		info->time = duration;
		info->filled |= TAGS_TIME;
	}

	if(tags_sel & TAGS_COMMENTS) {
		if ((tag_len = WavpackGetTagItem (wpc, "title", NULL, 0)) > 0) {
			info->title = (char *)xmalloc (++tag_len);
			WavpackGetTagItem (wpc, "title", info->title, tag_len);
		}

		if ((tag_len = WavpackGetTagItem (wpc, "artist", NULL, 0)) > 0) {
			info->artist = (char *)xmalloc (++tag_len);
			WavpackGetTagItem (wpc, "artist", info->artist, tag_len);
		}

		if ((tag_len = WavpackGetTagItem (wpc, "album", NULL, 0)) > 0) {
			info->album = (char *)xmalloc (++tag_len);
			WavpackGetTagItem (wpc, "album", info->album, tag_len);
		}

		if ((tag_len = WavpackGetTagItem (wpc, "track", NULL, 0)) > 0) {
			tag = (char *)xmalloc (++tag_len);
			WavpackGetTagItem (wpc, "track", tag, tag_len);
			info->track = atoi (tag);
			free (tag);
		}

		info->filled |= TAGS_COMMENTS;
	}

	WavpackCloseFile (wpc);
}
예제 #6
0
EmErrorCode WvDecoder::Open(const std::string& url)
{
    m_Ctx = WavpackOpenFileInput(url.c_str(), nullptr, 0, 0);
    if (m_Ctx == nullptr)
        return ErrorCode::DecoderFailedToOpen;

    if (WavpackGetNumSamples(m_Ctx) == (uint32_t) -1)
        return ErrorCode::DecoderFailedToInit;

    m_Duration = (double)WavpackGetNumSamples(m_Ctx) / WavpackGetSampleRate(m_Ctx) * 1000;
    m_Channels = WavpackGetNumChannels(m_Ctx);
    m_SampleRate = WavpackGetSampleRate(m_Ctx);
    m_BitsPerSample = WavpackGetBitsPerSample(m_Ctx);
    m_BytesPerSample = WavpackGetBytesPerSample(m_Ctx);

    // one sample may not be enough to build a full channel
    m_UnitCount = WavpackGetNumSamples(m_Ctx)/m_Channels;
    m_UnitIndex = 0;

    m_Buf.resize(10 * m_Channels * sizeof(int32_t)); // 1~10 full-samples

    return ErrorCode::Ok;
}
예제 #7
0
파일: wavpack_plugin.c 프로젝트: azuwis/mpd
/*
 * Reads metainfo from the specified file.
 */
static struct tag *
wavpack_tagdup(const char *fname)
{
	WavpackContext *wpc;
	struct tag *tag;
	char error[ERRORLEN];
	char *s;
	int size, allocated_size;

	wpc = WavpackOpenFileInput(fname, error, OPEN_TAGS, 0);
	if (wpc == NULL) {
		g_warning(
			"failed to open WavPack file \"%s\": %s\n",
			fname, error
		);
		return NULL;
	}

	tag = tag_new();
	tag->time = WavpackGetNumSamples(wpc);
	tag->time /= WavpackGetSampleRate(wpc);

	allocated_size = 0;
	s = NULL;

	for (unsigned i = 0; i < G_N_ELEMENTS(tagtypes); ++i) {
		size = WavpackGetTagItem(wpc, tagtypes[i].name, NULL, 0);
		if (size > 0) {
			++size; /* EOS */

			if (s == NULL) {
				s = g_malloc(size);
				allocated_size = size;
			} else if (size > allocated_size) {
				char *t = (char *)g_realloc(s, size);
				allocated_size = size;
				s = t;
			}

			WavpackGetTagItem(wpc, tagtypes[i].name, s, size);
			tag_add_item(tag, tagtypes[i].type, s);
		}
	}

	g_free(s);

	WavpackCloseFile(wpc);

	return tag;
}
예제 #8
0
static void wav_data_init (struct wavpack_data *data)
{
	data->sample_num = WavpackGetNumSamples (data->wpc);
	data->sample_rate = WavpackGetSampleRate (data->wpc);
	data->channels = WavpackGetReducedChannels (data->wpc);
	data->duration = data->sample_num / data->sample_rate;
	data->mode = WavpackGetMode (data->wpc);
	data->avg_bitrate = WavpackGetAverageBitrate (data->wpc, 1) / 1000;

	data->ok = 1;
	debug ("File opened. S_n %d. S_r %d. Time %d. Avg_Bitrate %d.",
		data->sample_num, data->sample_rate,
		data->duration, data->avg_bitrate
		);
}
예제 #9
0
static int wavpack_assign_values(struct mpxplay_filehand_buffered_func_s *fbfs,void *fbds,struct wavpack_decoder_data *wpdi,struct mpxplay_infile_info_s *miis)
{
 struct mpxplay_audio_decoder_info_s *adi=miis->audio_decoder_infos;
 unsigned int encmode;
 unsigned long pcmdatalen;

 adi->filechannels = adi->outchannels = WavpackGetReducedChannels(wpdi->wpc);//WavpackGetNumChannels(wpdi->wpc);
 if((adi->outchannels<PCM_MIN_CHANNELS) || (adi->outchannels>PCM_MAX_CHANNELS))
  return 0;

 adi->bits = WavpackGetBitsPerSample(wpdi->wpc);
 if((adi->bits<PCM_MIN_BITS) || (adi->bits>PCM_MAX_BITS))
  return 0;

 adi->freq = WavpackGetSampleRate(wpdi->wpc);
 wpdi->bytes_per_sample = WavpackGetBytesPerSample(wpdi->wpc);
 if(!adi->freq || !wpdi->bytes_per_sample)
  return 0;

 pcmdatalen=WavpackGetNumSamples(wpdi->wpc);

 miis->timemsec=(float)pcmdatalen*1000.0/adi->freq;

 encmode=WavpackGetMode(wpdi->wpc);
 if(encmode&MODE_FLOAT){
  adi->infobits|=ADI_FLAG_FLOATOUT;
  adi->bits=1;
  wpdi->bytes_per_sample=sizeof(MPXPLAY_PCMOUT_FLOAT_T);
 }

 miis->longname="WavPack ";

 if(encmode&MODE_HYBRID){
  adi->bitrate=(long)((float)miis->filesize*8.0/1000.0*(float)adi->freq/(float)pcmdatalen);
 }else{
  long compr_ratio;
  adi->bitratetext=malloc(MPXPLAY_ADITEXTSIZE_BITRATE+8);
  if(!adi->bitratetext)
   return 0;
  compr_ratio=(long)(1000.0*(float)miis->filesize/(float)pcmdatalen/(float)wpdi->bytes_per_sample/(float)adi->filechannels);
  sprintf(adi->bitratetext,"%2d/%2.2d.%1.1d%%",adi->bits,compr_ratio/10,compr_ratio%10);
 }

 return 1;
}
예제 #10
0
static void
wv_get_song_info(char *filename, char **title, int *length)
{
    assert(filename != NULL);
    char error_buff[80];
    WavpackContext *ctx = WavpackOpenFileInput(filename, error_buff, OPEN_TAGS | OPEN_WVC, 0);
    if (ctx == NULL) {
        printf("wavpack: Error opening file: \"%s: %s\"\n", filename, error_buff);
        return;
    }
    int sample_rate = WavpackGetSampleRate(ctx);
    int num_channels = WavpackGetNumChannels(ctx);
    DBG("reading %s at %d rate with %d channels\n", filename, sample_rate, num_channels);

    *length = (int)(WavpackGetNumSamples(ctx) / sample_rate) * 1000,
     *title = generate_title(filename, ctx);
    DBG("title for %s = %s\n", filename, *title);
    WavpackCloseFile(ctx);
}
예제 #11
0
SoundSource::OpenResult SoundSourceWV::tryOpen(const AudioSourceConfig& audioSrcCfg) {
    DEBUG_ASSERT(!m_wpc);
    char msg[80]; // hold possible error message
    int openFlags = OPEN_WVC | OPEN_NORMALIZE;
    if ((kChannelCountMono == audioSrcCfg.getChannelCount()) ||
            (kChannelCountStereo == audioSrcCfg.getChannelCount())) {
        openFlags |= OPEN_2CH_MAX;
    }

    // We use WavpackOpenFileInputEx to support Unicode paths on windows
    // http://www.wavpack.com/lib_use.txt
    QString wavPackFileName = getLocalFileName();
    m_pWVFile = new QFile(wavPackFileName);
    m_pWVFile->open(QFile::ReadOnly);
    QString correctionFileName(wavPackFileName + "c");
    if (QFile::exists(correctionFileName)) {
        // If there is a correction file, open it as well
        m_pWVCFile = new QFile(correctionFileName);
        m_pWVCFile->open(QFile::ReadOnly);
    }
    m_wpc = WavpackOpenFileInputEx(&s_streamReader, m_pWVFile, m_pWVCFile,
            msg, openFlags, 0);
    if (!m_wpc) {
        qDebug() << "SSWV::open: failed to open file : " << msg;
        return OpenResult::FAILED;
    }

    setChannelCount(WavpackGetReducedChannels(m_wpc));
    setSamplingRate(WavpackGetSampleRate(m_wpc));
    setFrameCount(WavpackGetNumSamples(m_wpc));

    if (WavpackGetMode(m_wpc) & MODE_FLOAT) {
        m_sampleScaleFactor = CSAMPLE_PEAK;
    } else {
        const int bitsPerSample = WavpackGetBitsPerSample(m_wpc);
        const uint32_t wavpackPeakSampleValue = 1u
                << (bitsPerSample - 1);
        m_sampleScaleFactor = CSAMPLE_PEAK / wavpackPeakSampleValue;
    }

    return OpenResult::SUCCEEDED;
}
예제 #12
0
bool AKSampler_Plugin::loadCompressedSampleFile(AKSampleFileDescriptor& sfd, float volBoostDb)
{
    char errMsg[100];
    WavpackContext* wpc = WavpackOpenFileInput(sfd.path, errMsg, OPEN_2CH_MAX, 0);
    if (wpc == 0)
    {
        printf("Wavpack error loading %s: %s\n", sfd.path, errMsg);
        return false;
    }
    
    AKSampleDataDescriptor sdd;
    sdd.sampleDescriptor = sfd.sampleDescriptor;
    sdd.sampleRate = (float)WavpackGetSampleRate(wpc);
    sdd.channelCount = WavpackGetReducedChannels(wpc);
    sdd.sampleCount = WavpackGetNumSamples(wpc);
    sdd.isInterleaved = sdd.channelCount > 1;
    sdd.data = new float[sdd.channelCount * sdd.sampleCount];
    
    int mode = WavpackGetMode(wpc);
    WavpackUnpackSamples(wpc, (int32_t*)sdd.data, sdd.sampleCount);
    if ((mode & MODE_FLOAT) == 0)
    {
        // convert samples to floating-point
        int bps = WavpackGetBitsPerSample(wpc);
        float scale = 1.0f / (1 << (bps - 1));
        float* pf = sdd.data;
        int32_t* pi = (int32_t*)pf;
        for (int i = 0; i < (sdd.sampleCount * sdd.channelCount); i++)
            *pf++ = scale * *pi++;
    }
    if (volBoostDb != 0.0f)
    {
        float scale = exp(volBoostDb / 20.0f);
        float* pf = sdd.data;
        for (int i = 0; i < (sdd.sampleCount * sdd.channelCount); i++)
            *pf++ *= scale;
    }
    
    loadSampleData(sdd);
    delete[] sdd.data;
    return true;
}
예제 #13
0
gboolean Wavpack_Header_Read_File_Info(gchar *filename, ET_File_Info *ETFileInfo)
{
    WavpackContext *wpc;

    wpc = WavpackOpenFileInput(filename, NULL, 0, 0);

    if ( wpc == NULL ) {
        return FALSE;
    }

    ETFileInfo->version     = WavpackGetVersion(wpc);
    /* .wvc correction file not counted */
    ETFileInfo->bitrate     = WavpackGetAverageBitrate(wpc, 0)/1000;
    ETFileInfo->samplerate  = WavpackGetSampleRate(wpc);
    ETFileInfo->mode        = WavpackGetNumChannels(wpc);
    ETFileInfo->size        = WavpackGetFileSize(wpc);
    ETFileInfo->duration    = WavpackGetNumSamples(wpc)/ETFileInfo->samplerate;

    WavpackCloseFile(wpc);

    return TRUE;
}
예제 #14
0
    bool attach(const char *filename)
    {
        ctx = WavpackOpenFileInput(filename, error_buff, OPEN_TAGS | OPEN_WVC | OPEN_NORMALIZE, 0);

        if (ctx == NULL) {
            return false;
        }

        sample_rate = WavpackGetSampleRate(ctx);
        num_channels = WavpackGetNumChannels(ctx);
        bytes_per_sample = WavpackGetBytesPerSample(ctx);
        input = (int32_t *)calloc(BUFFER_SIZE, num_channels * sizeof(int32_t));
        output = (int16_t *)calloc(BUFFER_SIZE, num_channels * sizeof(int16_t));
        memset (shaping_error, 0, sizeof (shaping_error));
        mod->set_info(generate_title(filename, ctx),
                      (int) (WavpackGetNumSamples(ctx) / sample_rate) * 1000,
                      (int) WavpackGetAverageBitrate(ctx, true),
                      (int) sample_rate, num_channels);
        play_gain = calculate_gain (ctx);
        DBG("gain value = %g\n", play_gain);
        return true;
    }
예제 #15
0
static void *
DecodeThread(void *a)
{
    ape_tag tag;
    char *filename = (char *) a;
    int bps_updateCounter = 0;
    int bps;
    int i;
    WavpackDecoder d(&mod);

    if (!d.attach(filename)) {
        printf("wavpack: Error opening file: \"%s\"\n", filename);
        killDecodeThread = true;
        return end_thread();
    }
    bps = WavpackGetBytesPerSample(d.ctx) * d.num_channels;
    DBG("reading %s at %d rate with %d channels\n", filename, d.sample_rate, d.num_channels);

    if (!d.open_audio()) {
        DBG("error opening xmms audio channel\n");
        killDecodeThread = true;
        AudioError = true;
        openedAudio = false;
    }
    else {
        DBG("opened xmms audio channel\n");
        openedAudio = true;
    }
    unsigned status;
    char *display = generate_title(filename, d.ctx);
    int length = (int) (1000 * WavpackGetNumSamples(d.ctx));

    while (!killDecodeThread) {
        if (isSeek != -1) {
            DBG("seeking to position %d\n", isSeek);
            WavpackSeekSample(d.ctx, isSeek * d.sample_rate);
            isSeek = -1;
        }
        if (paused == 0
                && (mod.output->buffer_free() >=
                    (1152 * 2 *
                     (16 / 8)) << (mod.output->buffer_playing()? 1 : 0))) {
            status =
                WavpackUnpackSamples(d.ctx, d.input, BUFFER_SIZE);
            if (status == (unsigned) (-1)) {
                printf("wavpack: Error decoding file.\n");
                break;
            }
            else if (status == 0) {
                killDecodeThread = true;
                break;
            }
            else {
                d.process_buffer(status);
            }
        }
        else {
            xmms_usleep(10000);
        }
    }
    return end_thread();
}
예제 #16
0
/*
 * This does the main decoding thing.
 * Requires an already opened WavpackContext.
 */
static void
wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek)
{
	GError *error = NULL;
	bool is_float;
	enum sample_format sample_format;
	struct audio_format audio_format;
	format_samples_t format_samples;
	float total_time;
	int bytes_per_sample, output_sample_size;

	is_float = (WavpackGetMode(wpc) & MODE_FLOAT) != 0;
	sample_format =
		wavpack_bits_to_sample_format(is_float,
					      WavpackGetBytesPerSample(wpc));

	if (!audio_format_init_checked(&audio_format,
				       WavpackGetSampleRate(wpc),
				       sample_format,
				       WavpackGetNumChannels(wpc), &error)) {
		g_warning("%s", error->message);
		g_error_free(error);
		return;
	}

	if (is_float) {
		format_samples = format_samples_float;
	} else {
		format_samples = format_samples_int;
	}

	total_time = WavpackGetNumSamples(wpc);
	total_time /= audio_format.sample_rate;
	bytes_per_sample = WavpackGetBytesPerSample(wpc);
	output_sample_size = audio_format_frame_size(&audio_format);

	/* wavpack gives us all kind of samples in a 32-bit space */
	int32_t chunk[1024];
	const uint32_t samples_requested = G_N_ELEMENTS(chunk) /
		audio_format.channels;

	decoder_initialized(decoder, &audio_format, can_seek, total_time);

	enum decoder_command cmd = decoder_get_command(decoder);
	while (cmd != DECODE_COMMAND_STOP) {
		if (cmd == DECODE_COMMAND_SEEK) {
			if (can_seek) {
				unsigned where = decoder_seek_where(decoder) *
					audio_format.sample_rate;

				if (WavpackSeekSample(wpc, where)) {
					decoder_command_finished(decoder);
				} else {
					decoder_seek_error(decoder);
				}
			} else {
				decoder_seek_error(decoder);
			}
		}

		uint32_t samples_got = WavpackUnpackSamples(wpc, chunk,
							    samples_requested);
		if (samples_got == 0)
			break;

		int bitrate = (int)(WavpackGetInstantBitrate(wpc) / 1000 +
				    0.5);
		format_samples(bytes_per_sample, chunk,
			       samples_got * audio_format.channels);

		cmd = decoder_data(decoder, NULL, chunk,
				   samples_got * output_sample_size,
				   bitrate);
	}
}
예제 #17
0
int snd_load_wv(const char *filename)
{
	SAMPLE *snd;
	int sid = -1;
	char error[100];
	WavpackContext *context;
	
	/* don't waste memory on sound when we are stress testing */
	if(config.dbg_stress)
		return -1;
		
	/* no need to load sound when we are running with no sound */
	if(!sound_enabled)
		return 1;

	file = engine_openfile(filename, IOFLAG_READ); /* TODO: use system.h stuff for this */
	if(!file)
	{
		dbg_msg("sound/wv", "failed to open %s", filename);
		return -1;
	}

	sid = snd_alloc_id();
	if(sid < 0)
		return -1;
	snd = &samples[sid];

	context = WavpackOpenFileInput(read_data, error);
	if (context)
	{
		int samples = WavpackGetNumSamples(context);
		int bitspersample = WavpackGetBitsPerSample(context);
		unsigned int samplerate = WavpackGetSampleRate(context);
		int channels = WavpackGetNumChannels(context);
		int *data;
		int *src;
		short *dst;
		int i;

		snd->channels = channels;
		snd->rate = samplerate;

		if(snd->channels > 2)
		{
			dbg_msg("sound/wv", "file is not mono or stereo. filename='%s'", filename);
			return -1;
		}

		/*
		if(snd->rate != 44100)
		{
			dbg_msg("sound/wv", "file is %d Hz, not 44100 Hz. filename='%s'", snd->rate, filename);
			return -1;
		}*/
		
		if(bitspersample != 16)
		{
			dbg_msg("sound/wv", "bps is %d, not 16, filname='%s'", bitspersample, filename);
			return -1;
		}

		data = (int *)mem_alloc(4*samples*channels, 1);
		WavpackUnpackSamples(context, data, samples); /* TODO: check return value */
		src = data;
		
		snd->data = (short *)mem_alloc(2*samples*channels, 1);
		dst = snd->data;

		for (i = 0; i < samples*channels; i++)
			*dst++ = (short)*src++;

		mem_free(data);

		snd->num_frames = samples;
		snd->loop_start = -1;
		snd->loop_end = -1;
	}
	else
	{
		dbg_msg("sound/wv", "failed to open %s: %s", filename, error);
	}

	io_close(file);
	file = NULL;

	if(config.debug)
		dbg_msg("sound/wv", "loaded %s", filename);

	rate_convert(sid);
	return sid;
}
예제 #18
0
static bool_t wv_play (InputPlayback * playback, const char * filename,
 VFSFile * file, int start_time, int stop_time, bool_t pause)
{
    if (file == NULL)
        return FALSE;

    int32_t *input = NULL;
    void *output = NULL;
    int sample_rate, num_channels, bits_per_sample;
    unsigned num_samples;
    WavpackContext *ctx = NULL;
    VFSFile *wvc_input = NULL;
    bool_t error = FALSE;

    if (! wv_attach (filename, file, & wvc_input, & ctx, NULL, OPEN_TAGS |
     OPEN_WVC))
    {
        fprintf (stderr, "Error opening Wavpack file '%s'.", filename);
        error = TRUE;
        goto error_exit;
    }

    sample_rate = WavpackGetSampleRate(ctx);
    num_channels = WavpackGetNumChannels(ctx);
    bits_per_sample = WavpackGetBitsPerSample(ctx);
    num_samples = WavpackGetNumSamples(ctx);

    if (!playback->output->open_audio(SAMPLE_FMT(bits_per_sample), sample_rate, num_channels))
    {
        fprintf (stderr, "Error opening audio output.");
        error = TRUE;
        goto error_exit;
    }

    if (pause)
        playback->output->pause(TRUE);

    input = malloc(BUFFER_SIZE * num_channels * sizeof(uint32_t));
    output = malloc(BUFFER_SIZE * num_channels * SAMPLE_SIZE(bits_per_sample));
    if (input == NULL || output == NULL)
        goto error_exit;

    playback->set_gain_from_playlist(playback);

    pthread_mutex_lock (& mutex);

    playback->set_params(playback, (int) WavpackGetAverageBitrate(ctx, num_channels),
        sample_rate, num_channels);

    seek_value = (start_time > 0) ? start_time : -1;
    stop_flag = FALSE;

    playback->set_pb_ready(playback);

    pthread_mutex_unlock (& mutex);

    while (!stop_flag && (stop_time < 0 ||
     playback->output->written_time () < stop_time))
    {
        int ret;
        unsigned samples_left;

        /* Handle seek and pause requests */
        pthread_mutex_lock (& mutex);

        if (seek_value >= 0)
        {
            playback->output->flush (seek_value);
            WavpackSeekSample (ctx, (int64_t) seek_value * sample_rate / 1000);
            seek_value = -1;
        }

        pthread_mutex_unlock (& mutex);

        /* Decode audio data */
        samples_left = num_samples - WavpackGetSampleIndex(ctx);

        ret = WavpackUnpackSamples(ctx, input, BUFFER_SIZE);
        if (samples_left == 0)
            stop_flag = TRUE;
        else if (ret < 0)
        {
            fprintf (stderr, "Error decoding file.\n");
            break;
        }
        else
        {
            /* Perform audio data conversion and output */
            unsigned i;
            int32_t *rp = input;
            int8_t *wp = output;
            int16_t *wp2 = output;
            int32_t *wp4 = output;

            if (bits_per_sample == 8)
            {
                for (i = 0; i < ret * num_channels; i++, wp++, rp++)
                    *wp = *rp & 0xff;
            }
            else if (bits_per_sample == 16)
            {
                for (i = 0; i < ret * num_channels; i++, wp2++, rp++)
                    *wp2 = *rp & 0xffff;
            }
            else if (bits_per_sample == 24 || bits_per_sample == 32)
            {
                for (i = 0; i < ret * num_channels; i++, wp4++, rp++)
                    *wp4 = *rp;
            }

            playback->output->write_audio(output, ret * num_channels * SAMPLE_SIZE(bits_per_sample));
        }
    }

error_exit:

    free(input);
    free(output);
    wv_deattach (wvc_input, ctx);

    stop_flag = TRUE;
    return ! error;
}
예제 #19
0
int CSound::DecodeWV(int SampleID, const void *pData, unsigned DataSize)
{
	if(SampleID == -1 || SampleID >= NUM_SAMPLES)
		return -1;

	CSample *pSample = &m_aSamples[SampleID];
	char aError[100];
	WavpackContext *pContext;

	ms_pWVBuffer = pData;
	ms_WVBufferSize = DataSize;
	ms_WVBufferPosition = 0;

	pContext = WavpackOpenFileInput(ReadData, aError);
	if (pContext)
	{
		int NumSamples = WavpackGetNumSamples(pContext);
		int BitsPerSample = WavpackGetBitsPerSample(pContext);
		unsigned int SampleRate = WavpackGetSampleRate(pContext);
		int NumChannels = WavpackGetNumChannels(pContext);
		int *pSrc;
		short *pDst;
		int i;

		pSample->m_Channels = NumChannels;
		pSample->m_Rate = SampleRate;

		if(pSample->m_Channels > 2)
		{
			dbg_msg("sound/wv", "file is not mono or stereo.");
			return -1;
		}

		if(BitsPerSample != 16)
		{
			dbg_msg("sound/wv", "bps is %d, not 16", BitsPerSample);
			return -1;
		}

		int *pBuffer = (int *)mem_alloc(4*NumSamples*NumChannels, 1);
		WavpackUnpackSamples(pContext, pBuffer, NumSamples); // TODO: check return value
		pSrc = pBuffer;

		pSample->m_pData = (short *)mem_alloc(2*NumSamples*NumChannels, 1);
		pDst = pSample->m_pData;

		for (i = 0; i < NumSamples*NumChannels; i++)
			*pDst++ = (short)*pSrc++;

		mem_free(pBuffer);

		pSample->m_NumFrames = NumSamples;
		pSample->m_LoopStart = -1;
		pSample->m_LoopEnd = -1;
		pSample->m_PausedAt = 0;
	}
	else
	{
		dbg_msg("sound/wv", "failed to decode sample (%s)", aError);
		return -1;
	}

	return SampleID;
}
예제 #20
0
static int get_length(void)
{
	return WavpackGetNumSamples(wpc) / WavpackGetSampleRate(wpc);
}
예제 #21
0
bool WavPackDecoder::Open(CFErrorRef *error)
{
	if(IsOpen()) {
		LOGGER_WARNING("org.sbooth.AudioEngine.AudioDecoder.WavPack", "Open() called on an AudioDecoder that is already open");		
		return true;
	}

	// Ensure the input source is open
	if(!mInputSource->IsOpen() && !mInputSource->Open(error))
		return false;

	mStreamReader.read_bytes = read_bytes_callback;
	mStreamReader.get_pos = get_pos_callback;
	mStreamReader.set_pos_abs = set_pos_abs_callback;
	mStreamReader.set_pos_rel = set_pos_rel_callback;
	mStreamReader.push_back_byte = push_back_byte_callback;
	mStreamReader.get_length = get_length_callback;
	mStreamReader.can_seek = can_seek_callback;
	
	char errorBuf [80];
	
	// Setup converter
	mWPC = WavpackOpenFileInputEx(&mStreamReader, this, nullptr, errorBuf, OPEN_WVC | OPEN_NORMALIZE, 0);
	if(nullptr == mWPC) {
		if(error) {
			CFStringRef description = CFCopyLocalizedString(CFSTR("The file “%@” is not a valid WavPack file."), "");
			CFStringRef failureReason = CFCopyLocalizedString(CFSTR("Not a WavPack file"), "");
			CFStringRef recoverySuggestion = CFCopyLocalizedString(CFSTR("The file's extension may not match the file's type."), "");
			
			*error = CreateErrorForURL(AudioDecoderErrorDomain, AudioDecoderInputOutputError, description, mInputSource->GetURL(), failureReason, recoverySuggestion);
			
			CFRelease(description), description = nullptr;
			CFRelease(failureReason), failureReason = nullptr;
			CFRelease(recoverySuggestion), recoverySuggestion = nullptr;
		}
		
		return false;
	}
	
	// Floating-point and lossy files will be handed off in the canonical Core Audio format
	int mode = WavpackGetMode(mWPC);
	if(MODE_FLOAT & mode || !(MODE_LOSSLESS & mode)) {
		// Canonical Core Audio format
		mFormat.mFormatID			= kAudioFormatLinearPCM;
		mFormat.mFormatFlags		= kAudioFormatFlagsNativeFloatPacked | kAudioFormatFlagIsNonInterleaved;
		
		mFormat.mSampleRate			= WavpackGetSampleRate(mWPC);
		mFormat.mChannelsPerFrame	= WavpackGetNumChannels(mWPC);		
		mFormat.mBitsPerChannel		= 8 * sizeof(float);
		
		mFormat.mBytesPerPacket		= (mFormat.mBitsPerChannel / 8);
		mFormat.mFramesPerPacket	= 1;
		mFormat.mBytesPerFrame		= mFormat.mBytesPerPacket * mFormat.mFramesPerPacket;
		
		mFormat.mReserved			= 0;
	}
	else {
		mFormat.mFormatID			= kAudioFormatLinearPCM;
		mFormat.mFormatFlags		= kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsNonInterleaved;

		// Don't set kAudioFormatFlagIsAlignedHigh for 32-bit integer files
		mFormat.mFormatFlags		|= (32 == WavpackGetBitsPerSample(mWPC) ? kAudioFormatFlagIsPacked : kAudioFormatFlagIsAlignedHigh);

		mFormat.mSampleRate			= WavpackGetSampleRate(mWPC);
		mFormat.mChannelsPerFrame	= WavpackGetNumChannels(mWPC);
		mFormat.mBitsPerChannel		= WavpackGetBitsPerSample(mWPC);
		
		mFormat.mBytesPerPacket		= sizeof(int32_t);
		mFormat.mFramesPerPacket	= 1;
		mFormat.mBytesPerFrame		= mFormat.mBytesPerPacket * mFormat.mFramesPerPacket;
		
		mFormat.mReserved			= 0;
	}
	
	mTotalFrames						= WavpackGetNumSamples(mWPC);
	
	// Set up the source format
	mSourceFormat.mFormatID				= 'WVPK';
	
	mSourceFormat.mSampleRate			= WavpackGetSampleRate(mWPC);
	mSourceFormat.mChannelsPerFrame		= WavpackGetNumChannels(mWPC);
	mSourceFormat.mBitsPerChannel		= WavpackGetBitsPerSample(mWPC);
	
	// Setup the channel layout
	switch(mFormat.mChannelsPerFrame) {
		case 1:		mChannelLayout = CreateChannelLayoutWithTag(kAudioChannelLayoutTag_Mono);			break;
		case 2:		mChannelLayout = CreateChannelLayoutWithTag(kAudioChannelLayoutTag_Stereo);			break;
		case 4:		mChannelLayout = CreateChannelLayoutWithTag(kAudioChannelLayoutTag_Quadraphonic);	break;
	}
	
	mBuffer = static_cast<int32_t *>(calloc(BUFFER_SIZE_FRAMES * mFormat.mChannelsPerFrame, sizeof(int32_t)));

	if(nullptr == mBuffer) {
		if(error)
			*error = CFErrorCreate(kCFAllocatorDefault, kCFErrorDomainPOSIX, ENOMEM, nullptr);
		
		return false;		
	}

	mIsOpen = true;
	return true;
}
bool WavPackDecoder::Open(CFErrorRef *error)
{
	if(IsOpen()) {
		log4cxx::LoggerPtr logger = log4cxx::Logger::getLogger("org.sbooth.AudioEngine.AudioDecoder.WavPack");
		LOG4CXX_WARN(logger, "Open() called on an AudioDecoder that is already open");		
		return true;
	}

	// Ensure the input source is open
	if(!mInputSource->IsOpen() && !mInputSource->Open(error))
		return false;

	mStreamReader.read_bytes = read_bytes_callback;
	mStreamReader.get_pos = get_pos_callback;
	mStreamReader.set_pos_abs = set_pos_abs_callback;
	mStreamReader.set_pos_rel = set_pos_rel_callback;
	mStreamReader.push_back_byte = push_back_byte_callback;
	mStreamReader.get_length = get_length_callback;
	mStreamReader.can_seek = can_seek_callback;
	
	char errorBuf [80];
	
	// Setup converter
	mWPC = WavpackOpenFileInputEx(&mStreamReader, this, NULL, errorBuf, OPEN_WVC | OPEN_NORMALIZE, 0);
	if(NULL == mWPC) {
		if(error) {
			CFMutableDictionaryRef errorDictionary = CFDictionaryCreateMutable(kCFAllocatorDefault, 
																			   32,
																			   &kCFTypeDictionaryKeyCallBacks,
																			   &kCFTypeDictionaryValueCallBacks);
			
			CFStringRef displayName = CreateDisplayNameForURL(mInputSource->GetURL());
			CFStringRef errorString = CFStringCreateWithFormat(kCFAllocatorDefault, 
															   NULL, 
															   CFCopyLocalizedString(CFSTR("The file “%@” is not a valid WavPack file."), ""), 
															   displayName);
			
			CFDictionarySetValue(errorDictionary, 
								 kCFErrorLocalizedDescriptionKey, 
								 errorString);
			
			CFDictionarySetValue(errorDictionary, 
								 kCFErrorLocalizedFailureReasonKey, 
								 CFCopyLocalizedString(CFSTR("Not a WavPack file"), ""));
			
			CFDictionarySetValue(errorDictionary, 
								 kCFErrorLocalizedRecoverySuggestionKey, 
								 CFCopyLocalizedString(CFSTR("The file's extension may not match the file's type."), ""));
			
			CFRelease(errorString), errorString = NULL;
			CFRelease(displayName), displayName = NULL;
			
			*error = CFErrorCreate(kCFAllocatorDefault, 
								   AudioDecoderErrorDomain, 
								   AudioDecoderInputOutputError, 
								   errorDictionary);
			
			CFRelease(errorDictionary), errorDictionary = NULL;				
		}
		
		return false;
	}
	
	// Floating-point and lossy files will be handed off in the canonical Core Audio format
	int mode = WavpackGetMode(mWPC);
	if(MODE_FLOAT & mode || !(MODE_LOSSLESS & mode)) {
		// Canonical Core Audio format
		mFormat.mFormatID			= kAudioFormatLinearPCM;
		mFormat.mFormatFlags		= kAudioFormatFlagsNativeFloatPacked | kAudioFormatFlagIsNonInterleaved;
		
		mFormat.mSampleRate			= WavpackGetSampleRate(mWPC);
		mFormat.mChannelsPerFrame	= WavpackGetNumChannels(mWPC);		
		mFormat.mBitsPerChannel		= 8 * sizeof(float);
		
		mFormat.mBytesPerPacket		= (mFormat.mBitsPerChannel / 8);
		mFormat.mFramesPerPacket	= 1;
		mFormat.mBytesPerFrame		= mFormat.mBytesPerPacket * mFormat.mFramesPerPacket;
		
		mFormat.mReserved			= 0;
	}
	else {
		mFormat.mFormatID			= kAudioFormatLinearPCM;
		mFormat.mFormatFlags		= kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsAlignedHigh | kAudioFormatFlagIsNonInterleaved;
		
		mFormat.mSampleRate			= WavpackGetSampleRate(mWPC);
		mFormat.mChannelsPerFrame	= WavpackGetNumChannels(mWPC);
		mFormat.mBitsPerChannel		= WavpackGetBitsPerSample(mWPC);
		
		mFormat.mBytesPerPacket		= sizeof(int32_t);
		mFormat.mFramesPerPacket	= 1;
		mFormat.mBytesPerFrame		= mFormat.mBytesPerPacket * mFormat.mFramesPerPacket;
		
		mFormat.mReserved			= 0;
	}
	
	mTotalFrames						= WavpackGetNumSamples(mWPC);
	
	// Set up the source format
	mSourceFormat.mFormatID				= 'WVPK';
	
	mSourceFormat.mSampleRate			= WavpackGetSampleRate(mWPC);
	mSourceFormat.mChannelsPerFrame		= WavpackGetNumChannels(mWPC);
	mSourceFormat.mBitsPerChannel		= WavpackGetBitsPerSample(mWPC);
	
	// Setup the channel layout
	switch(mFormat.mChannelsPerFrame) {
		case 1:		mChannelLayout = CreateChannelLayoutWithTag(kAudioChannelLayoutTag_Mono);			break;
		case 2:		mChannelLayout = CreateChannelLayoutWithTag(kAudioChannelLayoutTag_Stereo);			break;
		case 4:		mChannelLayout = CreateChannelLayoutWithTag(kAudioChannelLayoutTag_Quadraphonic);	break;
	}
	
	mBuffer = static_cast<int32_t *>(calloc(BUFFER_SIZE_FRAMES * mFormat.mChannelsPerFrame, sizeof(int32_t)));

	if(NULL == mBuffer) {
		if(error)
			*error = CFErrorCreate(kCFAllocatorDefault, kCFErrorDomainPOSIX, ENOMEM, NULL);
		
		return false;		
	}

	mIsOpen = true;
	return true;
}
예제 #23
0
파일: wavpack.c 프로젝트: dsheeler/xmms2
static gboolean
xmms_wavpack_init (xmms_xform_t *xform)
{
	xmms_wavpack_data_t *data;
	xmms_sample_format_t sample_format;
	gint samplerate;
	/* the maximum length of error really isn't defined... stupid */
	gchar error[1024];

	g_return_val_if_fail (xform, FALSE);

	if (!xmms_apetag_read (xform)) {
		XMMS_DBG ("Failed to read APEv2 tag");
	}

	data = g_new0 (xmms_wavpack_data_t, 1);
	g_return_val_if_fail (data, FALSE);

	xmms_xform_private_data_set (xform, data);

	data->reader.read_bytes = wavpack_read_bytes;
	data->reader.get_pos = wavpack_get_pos;
	data->reader.set_pos_abs = wavpack_set_pos_abs;
	data->reader.set_pos_rel = wavpack_set_pos_rel;
	data->reader.push_back_byte = wavpack_push_back_byte;
	data->reader.get_length = wavpack_get_length;
	data->reader.can_seek = wavpack_can_seek;

	data->ctx = WavpackOpenFileInputEx (&data->reader,
	                                    xform, xform,
	                                    error, OPEN_TAGS, 0);

	if (!data->ctx) {
		xmms_log_error ("Unable to open wavpack file: %s", error);
		xmms_xform_private_data_set (xform, NULL);
		xmms_wavpack_free_data (data);
		return FALSE;
	}

	data->channels = WavpackGetNumChannels (data->ctx);
	data->bits_per_sample = WavpackGetBitsPerSample (data->ctx);
	data->num_samples = WavpackGetNumSamples (data->ctx);
	samplerate = WavpackGetSampleRate (data->ctx);

	xmms_xform_metadata_set_int (xform,
	                             XMMS_MEDIALIB_ENTRY_PROPERTY_DURATION,
	                             (int) (1000.0 * data->num_samples / samplerate));
	xmms_xform_metadata_set_int (xform,
	                             XMMS_MEDIALIB_ENTRY_PROPERTY_SAMPLERATE,
	                             samplerate);
	xmms_xform_metadata_set_int (xform,
	                             XMMS_MEDIALIB_ENTRY_PROPERTY_BITRATE,
	                             (int) WavpackGetAverageBitrate (data->ctx, FALSE));

	switch (data->bits_per_sample) {
	case 8:
		sample_format = XMMS_SAMPLE_FORMAT_S8;
		break;
	case 12:
	case 16:
		sample_format = XMMS_SAMPLE_FORMAT_S16;
		break;
	case 24:
	case 32:
		sample_format = XMMS_SAMPLE_FORMAT_S32;
		break;
	default:
		xmms_log_error ("Unsupported bits-per-sample in wavpack file: %d",
		                data->bits_per_sample);
		xmms_xform_private_data_set (xform, NULL);
		xmms_wavpack_free_data (data);
		return FALSE;
	}

	xmms_xform_outdata_type_add (xform,
	                             XMMS_STREAM_TYPE_MIMETYPE,
	                             "audio/pcm",
	                             XMMS_STREAM_TYPE_FMT_FORMAT,
	                             sample_format,
	                             XMMS_STREAM_TYPE_FMT_CHANNELS,
	                             data->channels,
	                             XMMS_STREAM_TYPE_FMT_SAMPLERATE,
	                             samplerate,
	                             XMMS_STREAM_TYPE_END);

	return TRUE;
}
예제 #24
0
파일: wavpack_plugin.c 프로젝트: azuwis/mpd
/*
 * This does the main decoding thing.
 * Requires an already opened WavpackContext.
 */
static void
wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
	       struct replay_gain_info *replay_gain_info)
{
	struct audio_format audio_format;
	format_samples_t format_samples;
	char chunk[CHUNK_SIZE];
	int samples_requested, samples_got;
	float total_time, current_time;
	int bytes_per_sample, output_sample_size;
	int position;

	audio_format.sample_rate = WavpackGetSampleRate(wpc);
	audio_format.channels = WavpackGetReducedChannels(wpc);
	audio_format.bits = WavpackGetBitsPerSample(wpc);

	/* round bitwidth to 8-bit units */
	audio_format.bits = (audio_format.bits + 7) & (~7);
	/* mpd handles max 24-bit samples */
	if (audio_format.bits > 24) {
		audio_format.bits = 24;
	}

	if (!audio_format_valid(&audio_format)) {
		g_warning("Invalid audio format: %u:%u:%u\n",
			  audio_format.sample_rate,
			  audio_format.bits,
			  audio_format.channels);
		return;
	}

	if ((WavpackGetMode(wpc) & MODE_FLOAT) == MODE_FLOAT) {
		format_samples = format_samples_float;
	} else {
		format_samples = format_samples_int;
	}

	total_time = WavpackGetNumSamples(wpc);
	total_time /= audio_format.sample_rate;
	bytes_per_sample = WavpackGetBytesPerSample(wpc);
	output_sample_size = audio_format_frame_size(&audio_format);

	/* wavpack gives us all kind of samples in a 32-bit space */
	samples_requested = sizeof(chunk) / (4 * audio_format.channels);

	decoder_initialized(decoder, &audio_format, can_seek, total_time);

	position = 0;

	do {
		if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
			if (can_seek) {
				int where;

				where = decoder_seek_where(decoder);
				where *= audio_format.sample_rate;
				if (WavpackSeekSample(wpc, where)) {
					position = where;
					decoder_command_finished(decoder);
				} else {
					decoder_seek_error(decoder);
				}
			} else {
				decoder_seek_error(decoder);
			}
		}

		if (decoder_get_command(decoder) == DECODE_COMMAND_STOP) {
			break;
		}

		samples_got = WavpackUnpackSamples(
			wpc, (int32_t *)chunk, samples_requested
		);
		if (samples_got > 0) {
			int bitrate = (int)(WavpackGetInstantBitrate(wpc) /
			              1000 + 0.5);
			position += samples_got;
			current_time = position;
			current_time /= audio_format.sample_rate;

			format_samples(
				bytes_per_sample, chunk,
				samples_got * audio_format.channels
			);

			decoder_data(
				decoder, NULL, chunk,
				samples_got * output_sample_size,
				current_time, bitrate,
				replay_gain_info
			);
		}
	} while (samples_got > 0);
}
예제 #25
0
파일: wavpack.c 프로젝트: jubalh/deadbeef
static DB_playItem_t *
wv_insert (ddb_playlist_t *plt, DB_playItem_t *after, const char *fname) {
    DB_FILE *fp = deadbeef->fopen (fname);
    if (!fp) {
        return NULL;
    }
    char error[80];
#ifdef TINYWV
    WavpackContext *ctx = WavpackOpenFileInput (wv_read_stream, fp, error);
#else
    WavpackContext *ctx = WavpackOpenFileInputEx (&wsr, fp, NULL, error, 0, 0);
#endif
    if (!ctx) {
        fprintf (stderr, "wavpack error: %s\n", error);
        deadbeef->fclose (fp);
        return NULL;
    }
    int totalsamples = WavpackGetNumSamples (ctx);
    int samplerate = WavpackGetSampleRate (ctx);
    float duration = (float)totalsamples / samplerate;

    DB_playItem_t *it = deadbeef->pl_item_alloc_init (fname, plugin.plugin.id);
    deadbeef->pl_add_meta (it, ":FILETYPE", "wv");
    deadbeef->plt_set_item_duration (plt, it, duration);
    trace ("wv: totalsamples=%d, samplerate=%d, duration=%f\n", totalsamples, samplerate, duration);

#if 0
    int num = WavpackGetNumTagItems (ctx);
    trace ("num tag items: %d\n", num);

    for (int i = 0; i < num; i++) {
        char str[1024];
        WavpackGetTagItemIndexed (ctx, i, str, sizeof (str));
        trace ("tag item: %s\n", str);
    }

#endif
    int apeerr = deadbeef->junk_apev2_read (it, fp);
    if (!apeerr) {
        trace ("wv: ape tag found\n");
    }
    int v1err = deadbeef->junk_id3v1_read (it, fp);
    if (!v1err) {
        trace ("wv: id3v1 tag found\n");
    }
    deadbeef->pl_add_meta (it, "title", NULL);

    char s[100];
    snprintf (s, sizeof (s), "%lld", deadbeef->fgetlength (fp));
    deadbeef->pl_add_meta (it, ":FILE_SIZE", s);
    snprintf (s, sizeof (s), "%d", WavpackGetBytesPerSample (ctx) * 8);
    deadbeef->pl_add_meta (it, ":BPS", s);
    snprintf (s, sizeof (s), "%d", WavpackGetNumChannels (ctx));
    deadbeef->pl_add_meta (it, ":CHANNELS", s);
    snprintf (s, sizeof (s), "%d", WavpackGetSampleRate (ctx));
    deadbeef->pl_add_meta (it, ":SAMPLERATE", s);
    snprintf (s, sizeof (s), "%d", (int)(WavpackGetAverageBitrate (ctx, 1) / 1000));
    deadbeef->pl_add_meta (it, ":BITRATE", s);
    snprintf (s, sizeof (s), "%s", (WavpackGetMode (ctx) & MODE_FLOAT) ? "FLOAT" : "INTEGER");
    deadbeef->pl_add_meta (it, ":WAVPACK_MODE", s);

    // embedded cue
    deadbeef->pl_lock ();
    const char *cuesheet = deadbeef->pl_find_meta (it, "cuesheet");
    if (cuesheet) {
        trace ("found cuesheet: %s\n", cuesheet);
        DB_playItem_t *last = deadbeef->plt_insert_cue_from_buffer (plt, after, it, cuesheet, strlen (cuesheet), totalsamples, samplerate);
        if (last) {
            deadbeef->pl_unlock ();
            deadbeef->fclose (fp);
            WavpackCloseFile (ctx);
            deadbeef->pl_item_unref (it);
            deadbeef->pl_item_unref (last);
            return last;
        }
    }
    deadbeef->pl_unlock ();
    // cue file on disc
    DB_playItem_t *cue_after = deadbeef->plt_insert_cue (plt, after, it, totalsamples, samplerate);
    if (cue_after) {
        deadbeef->fclose (fp);
        WavpackCloseFile (ctx);
        deadbeef->pl_item_unref (it);
        deadbeef->pl_item_unref (cue_after);
        return cue_after;
    }

    after = deadbeef->plt_insert_item (plt, after, it);
    deadbeef->pl_item_unref (it);

    deadbeef->fclose (fp);
    WavpackCloseFile (ctx);
    return after;
}
예제 #26
0
파일: wavpack.c 프로젝트: jubalh/deadbeef
static int
wv_init (DB_fileinfo_t *_info, DB_playItem_t *it) {
    wvctx_t *info = (wvctx_t *)_info;
    deadbeef->pl_lock ();
    info->file = deadbeef->fopen (deadbeef->pl_find_meta (it, ":URI"));
    deadbeef->pl_unlock ();
    if (!info->file) {
        return -1;
    }

#ifndef TINYWV
    deadbeef->pl_lock ();
    const char *uri = deadbeef->pl_find_meta (it, ":URI");
    char *c_fname = alloca (strlen (uri) + 2);
    if (c_fname) {
        strcpy (c_fname, uri);
        strcat (c_fname, "c");
        info->c_file = deadbeef->fopen (c_fname);
    }
    else {
        fprintf (stderr, "wavpack warning: failed to alloc memory for correction file name\n");
    }
    deadbeef->pl_unlock ();
#endif

    char error[80];
#ifdef TINYWV
    info->ctx = WavpackOpenFileInput (wv_read_stream, info->file, error);
#else
    info->ctx = WavpackOpenFileInputEx (&wsr, info->file, info->c_file, error, OPEN_NORMALIZE, 0);
#endif
    if (!info->ctx) {
        fprintf (stderr, "wavpack error: %s\n", error);
        return -1;
    }
    _info->plugin = &plugin;
    _info->fmt.bps = WavpackGetBytesPerSample (info->ctx) * 8;
    _info->fmt.channels = WavpackGetNumChannels (info->ctx);
    _info->fmt.samplerate = WavpackGetSampleRate (info->ctx);
    _info->fmt.is_float = (WavpackGetMode (info->ctx) & MODE_FLOAT) ? 1 : 0;

    // FIXME: streamer and maybe output plugins need to be fixed to support
    // arbitrary channelmask

    // _info->fmt.channelmask = WavpackGetChannelMask (info->ctx);

    for (int i = 0; i < _info->fmt.channels; i++) {
        _info->fmt.channelmask |= 1 << i;
    }
    _info->readpos = 0;
    if (it->endsample > 0) {
        info->startsample = it->startsample;
        info->endsample = it->endsample;
        if (plugin.seek_sample (_info, 0) < 0) {
            return -1;
        }
    }
    else {
        info->startsample = 0;
        info->endsample = WavpackGetNumSamples (info->ctx)-1;
    }
    return 0;
}
예제 #27
0
static gboolean wv_play (InputPlayback * playback, const gchar * filename,
 VFSFile * file, gint start_time, gint stop_time, gboolean pause)
{
    if (file == NULL)
        return FALSE;

    gint32 *input = NULL;
    void *output = NULL;
    gint sample_rate, num_channels, bits_per_sample;
    guint num_samples;
    WavpackContext *ctx = NULL;
    VFSFile *wvc_input = NULL;
    gboolean error = FALSE;

    if (! wv_attach (filename, file, & wvc_input, & ctx, NULL, OPEN_TAGS |
     OPEN_WVC))
    {
        g_warning("Error opening Wavpack file '%s'.", filename);
        error = TRUE;
        goto error_exit;
    }

    sample_rate = WavpackGetSampleRate(ctx);
    num_channels = WavpackGetNumChannels(ctx);
    bits_per_sample = WavpackGetBitsPerSample(ctx);
    num_samples = WavpackGetNumSamples(ctx);

    if (!playback->output->open_audio(SAMPLE_FMT(bits_per_sample), sample_rate, num_channels))
    {
        g_warning("Error opening audio output.");
        error = TRUE;
        goto error_exit;
    }

    if (pause)
        playback->output->pause(TRUE);

    input = g_malloc(BUFFER_SIZE * num_channels * sizeof(guint32));
    output = g_malloc(BUFFER_SIZE * num_channels * SAMPLE_SIZE(bits_per_sample));
    if (input == NULL || output == NULL)
        goto error_exit;

    playback->set_gain_from_playlist(playback);

    g_mutex_lock(ctrl_mutex);

    playback->set_params(playback, (gint) WavpackGetAverageBitrate(ctx, num_channels),
        sample_rate, num_channels);

    seek_value = (start_time > 0) ? start_time : -1;
    stop_flag = FALSE;

    playback->set_pb_ready(playback);

    g_mutex_unlock(ctrl_mutex);

    while (!stop_flag && (stop_time < 0 ||
     playback->output->written_time () < stop_time))
    {
        gint ret;
        guint samples_left;

        /* Handle seek and pause requests */
        g_mutex_lock(ctrl_mutex);

        if (seek_value >= 0)
        {
            playback->output->flush (seek_value);
            WavpackSeekSample (ctx, (gint64) seek_value * sample_rate / 1000);
            seek_value = -1;
            g_cond_signal(ctrl_cond);
        }

        g_mutex_unlock(ctrl_mutex);

        /* Decode audio data */
        samples_left = num_samples - WavpackGetSampleIndex(ctx);

        ret = WavpackUnpackSamples(ctx, input, BUFFER_SIZE);
        if (samples_left == 0)
            stop_flag = TRUE;
        else if (ret < 0)
        {
            g_warning("Error decoding file.\n");
            break;
        }
        else
        {
            /* Perform audio data conversion and output */
            guint i;
            gint32 *rp = input;
            gint8 *wp = output;
            gint16 *wp2 = output;
            gint32 *wp4 = output;

            if (bits_per_sample == 8)
            {
                for (i = 0; i < ret * num_channels; i++, wp++, rp++)
                    *wp = *rp & 0xff;
            }
            else if (bits_per_sample == 16)
            {
                for (i = 0; i < ret * num_channels; i++, wp2++, rp++)
                    *wp2 = *rp & 0xffff;
            }
            else if (bits_per_sample == 24 || bits_per_sample == 32)
            {
                for (i = 0; i < ret * num_channels; i++, wp4++, rp++)
                    *wp4 = *rp;
            }

            playback->output->write_audio(output, ret * num_channels * SAMPLE_SIZE(bits_per_sample));
        }
    }

    /* Flush buffer */
    g_mutex_lock(ctrl_mutex);

    while (!stop_flag && playback->output->buffer_playing())
        g_usleep(20000);

    g_cond_signal(ctrl_cond);
    g_mutex_unlock(ctrl_mutex);

error_exit:

    g_free(input);
    g_free(output);
    wv_deattach (wvc_input, ctx);

    stop_flag = TRUE;
    playback->output->close_audio();
    return ! error;
}
예제 #28
0
파일: sound.cpp 프로젝트: CytraL/MineTee
int CSound::LoadWV(const char *pFilename)
{
	CSample *pSample;
	int SampleID = -1;
	char aError[100];
	WavpackContext *pContext;

	// don't waste memory on sound when we are stress testing
	if(g_Config.m_DbgStress)
		return -1;

	// no need to load sound when we are running with no sound
	if(!m_SoundEnabled)
		return 1;

	if(!m_pStorage)
		return -1;

	ms_File = m_pStorage->OpenFile(pFilename, IOFLAG_READ, IStorage::TYPE_ALL);
	if(!ms_File)
	{
		dbg_msg("sound/wv", "failed to open file. filename='%s'", pFilename);
		return -1;
	}

	SampleID = AllocID();
	if(SampleID < 0)
		return -1;
	pSample = &m_aSamples[SampleID];

	pContext = WavpackOpenFileInput(ReadData, aError);
	if (pContext)
	{
		int m_aSamples = WavpackGetNumSamples(pContext);
		int BitsPerSample = WavpackGetBitsPerSample(pContext);
		unsigned int SampleRate = WavpackGetSampleRate(pContext);
		int m_aChannels = WavpackGetNumChannels(pContext);
		int *pData;
		int *pSrc;
		short *pDst;
		int i;

		pSample->m_Channels = m_aChannels;
		pSample->m_Rate = SampleRate;

		if(pSample->m_Channels > 2)
		{
			dbg_msg("sound/wv", "file is not mono or stereo. filename='%s'", pFilename);
			return -1;
		}

		/*
		if(snd->rate != 44100)
		{
			dbg_msg("sound/wv", "file is %d Hz, not 44100 Hz. filename='%s'", snd->rate, filename);
			return -1;
		}*/

		if(BitsPerSample != 16)
		{
			dbg_msg("sound/wv", "bps is %d, not 16, filname='%s'", BitsPerSample, pFilename);
			return -1;
		}

		pData = (int *)mem_alloc(4*m_aSamples*m_aChannels, 1);
		WavpackUnpackSamples(pContext, pData, m_aSamples); // TODO: check return value
		pSrc = pData;

		pSample->m_pData = (short *)mem_alloc(2*m_aSamples*m_aChannels, 1);
		pDst = pSample->m_pData;

		for (i = 0; i < m_aSamples*m_aChannels; i++)
			*pDst++ = (short)*pSrc++;

		mem_free(pData);

		pSample->m_NumFrames = m_aSamples;
		pSample->m_LoopStart = -1;
		pSample->m_LoopEnd = -1;
		pSample->m_PausedAt = 0;
	}
	else
	{
		dbg_msg("sound/wv", "failed to open %s: %s", pFilename, aError);
	}

	io_close(ms_File);
	ms_File = NULL;

	if(g_Config.m_Debug)
		dbg_msg("sound/wv", "loaded %s", pFilename);

	RateConvert(SampleID);
	return SampleID;
}