SINT SoundSourceWV::readSampleFrames(
SINT numberOfFrames, CSAMPLE* sampleBuffer) {
if (sampleBuffer == nullptr) {
// NOTE(uklotzde): The WavPack API does not provide any
// functions for skipping samples in the audio stream. Calling
// API functions with a nullptr buffer does not return. Since
// we don't want to read samples into a temporary buffer that
// has to be allocated we are seeking to the position after
// the skipped samples.
SINT curFrameIndexBefore = m_curFrameIndex;
SINT curFrameIndexAfter = seekSampleFrame(m_curFrameIndex + numberOfFrames);
DEBUG_ASSERT(curFrameIndexBefore <= curFrameIndexAfter);
DEBUG_ASSERT(m_curFrameIndex == curFrameIndexAfter);
return curFrameIndexAfter - curFrameIndexBefore;
}
// static assert: sizeof(CSAMPLE) == sizeof(int32_t)
SINT unpackCount = WavpackUnpackSamples(m_wpc,
reinterpret_cast<int32_t*>(sampleBuffer), numberOfFrames);
DEBUG_ASSERT(unpackCount >= 0);
if (!(WavpackGetMode(m_wpc) & MODE_FLOAT)) {
// signed integer -> float
const SINT sampleCount = frames2samples(unpackCount);
for (SINT i = 0; i < sampleCount; ++i) {
const int32_t sampleValue =
reinterpret_cast<int32_t*>(sampleBuffer)[i];
sampleBuffer[i] = CSAMPLE(sampleValue) * m_sampleScaleFactor;
}
}
m_curFrameIndex += unpackCount;
return unpackCount;
}
index_t
soundfile_read_double(soundfile_t *sf, double *buffer, index_t n_frames) {
if (sf->t == sft_libsndfile) {
return sf_read_double(sf->p, buffer, n_frames);
} else {
int buffer_size = 65536;
int32_t b[buffer_size*sf->channels];
index_t frames_read = 0;
double divisor = 1L << (sf->bits_per_sample - 1);
while (1) {
int to_read = MIN(buffer_size, n_frames - frames_read);
if (to_read <= 0)
return frames_read;
int n = WavpackUnpackSamples(sf->p, b, to_read);
if (n < 0 || (n == 0 && frames_read == 0))
die("WavpackUnpackSamples returned %d", n);
if (n == 0)
return frames_read;
for (index_t f = 0; f < n; f++) {
for (int c = 0; c < sf->channels; c++)
buffer[frames_read*sf->channels+c] = b[f*sf->channels+c]/divisor;
frames_read++;
}
}
}
}
static int wav_decode (void *prv_data, char *buf, int buf_len,
struct sound_params *sound_params)
{
struct wavpack_data *data = (struct wavpack_data *)prv_data;
int ret, i, s_num, iBps, oBps;
int8_t * buf8 = (int8_t *)buf;
int16_t * buf16 = (int16_t *)buf;
int32_t * buf32 = (int32_t *)buf;
iBps = data->channels * WavpackGetBytesPerSample (data->wpc);
oBps = (iBps == 6) ? 8 : iBps;
s_num = buf_len / oBps;
decoder_error_clear (&data->error);
int32_t *dbuf = (int32_t *)xcalloc (
s_num, data->channels * 4);
ret = WavpackUnpackSamples (data->wpc, dbuf, s_num );
if (ret == 0) {
free (dbuf);
return 0;
}
if (data->mode & MODE_FLOAT) {
sound_params->fmt = SFMT_FLOAT;
memcpy (buf, dbuf, ret * oBps);
} else {
debug ("iBps %d", iBps);
switch (iBps / data->channels){
case 4: for (i = 0; i < ret * data->channels; i++)
buf32[i] = dbuf[i];
sound_params->fmt = SFMT_S32 | SFMT_NE;
break;
case 3: for (i = 0; i < ret * data->channels; i++)
buf32[i] = dbuf[i] * 256;
sound_params->fmt = SFMT_S32 | SFMT_NE;
break;
case 2: for (i = 0; i < ret * data->channels; i++)
buf16[i] = dbuf[i];
sound_params->fmt = SFMT_S16 | SFMT_NE;
break;
case 1: for (i = 0; i < ret * data->channels; i++)
buf8[i] = dbuf[i];
sound_params->fmt = SFMT_S8 | SFMT_NE;
}
}
sound_params->channels = data->channels;
sound_params->rate = data->sample_rate;
free (dbuf);
return ret * oBps ;
}
decoded_samples WavPackDecoder::decode(const char* data, std::size_t bytes)
{
_data = data;
_dataSize = bytes;
_pos = 0;
if (! _context) {
_context = WavpackOpenFileInputEx(&_streamReader, this, 0, _errMsgStorage.get(), OPEN_STREAMING, 0);
if (! _context) {
std::string errmsg("Could not set up input stream: ");
errmsg += std::string(_errMsgStorage.get());
throw std::runtime_error(errmsg);
}
}
uint32_t sampleCount = WavpackUnpackSamples(_context, _decodeBuffer.get(), DECODE_BUFFER_SZ);
uint32_t sampleCountFinal = sampleCount;
int32_t* p = _decodeBuffer.get();
char* outBytes = _decodedPcm.get();
//unpack the samples back to original bit depth
switch (BIT_DEPTH) {
case 16:
while (sampleCount > 0) {
*((int16_t *)outBytes) = (int16_t)(*p);
outBytes += sizeof (int16_t);
p++;
sampleCount--;
}
break;
case 8:
while (sampleCount > 0) {
*outBytes++ = (char)(*p);
p++;
sampleCount--;
}
break;
}
decoded_samples ret;
ret.NumSamples = sampleCountFinal;
ret.SampleData = _decodedPcm.get();
return ret;
}
static int WAVPACK_infile_decode(struct mpxplay_filehand_buffered_func_s *fbfs,void *fbds,struct mpxplay_infile_info_s *miis)
{
mpxplay_audio_decoder_info_s *adi=miis->audio_decoder_infos;
struct wavpack_decoder_data *wpdi=miis->private_data;
WavpackUpdateStreamhand(wpdi->wpc,fbds);
adi->pcm_samplenum=WavpackUnpackSamples(wpdi->wpc,(long *)adi->pcm_bufptr,adi->pcm_framelen)*adi->outchannels;
if(!adi->pcm_samplenum)
return MPXPLAY_ERROR_INFILE_NODATA;
if(wpdi->bytes_per_sample<4)
format_samples(wpdi->bytes_per_sample,(unsigned char *)adi->pcm_bufptr,(long *)adi->pcm_bufptr,adi->pcm_samplenum);
return MPXPLAY_ERROR_INFILE_OK;
}
SINT SoundSourceWV::readSampleFrames(
SINT numberOfFrames, CSAMPLE* sampleBuffer) {
// static assert: sizeof(CSAMPLE) == sizeof(int32_t)
SINT unpackCount = WavpackUnpackSamples(m_wpc,
reinterpret_cast<int32_t*>(sampleBuffer), numberOfFrames);
if (!(WavpackGetMode(m_wpc) & MODE_FLOAT)) {
// signed integer -> float
const SINT sampleCount = frames2samples(unpackCount);
for (SINT i = 0; i < sampleCount; ++i) {
const int32_t sampleValue =
reinterpret_cast<int32_t*>(sampleBuffer)[i];
sampleBuffer[i] = CSAMPLE(sampleValue) * m_sampleScaleFactor;
}
}
return unpackCount;
}
static int decode_data(char *target, int max_size)
{
int bps, channels;
uint32_t samples_unpacked = 0;
channels = get_channels();
bps = WavpackGetBytesPerSample(wpc);
if (max_size >= 1024) {
samples_unpacked = WavpackUnpackSamples(wpc, temp_buffer, 256 / channels);
total_unpacked_samples += samples_unpacked;
if (samples_unpacked)
format_samples(bps, (uchar *)target, temp_buffer, samples_unpacked * channels);
} else {
printf("vorbis: Target buffer too small: %d < 1024\n", max_size);
}
return samples_unpacked * (WavpackGetBitsPerSample(wpc) / 4);
}
ReadableSampleFrames SoundSourceWV::readSampleFramesClamped(
WritableSampleFrames writableSampleFrames) {
const SINT firstFrameIndex = writableSampleFrames.frameIndexRange().start();
if (m_curFrameIndex != firstFrameIndex) {
if (WavpackSeekSample(m_wpc, firstFrameIndex)) {
m_curFrameIndex = firstFrameIndex;
} else {
kLogger.warning()
<< "Could not seek to first frame index"
<< firstFrameIndex;
m_curFrameIndex = WavpackGetSampleIndex(m_wpc);
return ReadableSampleFrames(IndexRange::between(m_curFrameIndex, m_curFrameIndex));
}
}
DEBUG_ASSERT(m_curFrameIndex == firstFrameIndex);
const SINT numberOfFramesTotal = writableSampleFrames.frameLength();
static_assert(sizeof(CSAMPLE) == sizeof(int32_t),
"CSAMPLE and int32_t must have the same size");
CSAMPLE* pOutputBuffer = writableSampleFrames.writableData();
SINT unpackCount = WavpackUnpackSamples(m_wpc,
reinterpret_cast<int32_t*>(pOutputBuffer), numberOfFramesTotal);
DEBUG_ASSERT(unpackCount >= 0);
DEBUG_ASSERT(unpackCount <= numberOfFramesTotal);
if (!(WavpackGetMode(m_wpc) & MODE_FLOAT)) {
// signed integer -> float
const SINT sampleCount = frames2samples(unpackCount);
for (SINT i = 0; i < sampleCount; ++i) {
const int32_t sampleValue =
*reinterpret_cast<int32_t*>(pOutputBuffer);
*pOutputBuffer++ = CSAMPLE(sampleValue) * m_sampleScaleFactor;
}
}
const auto resultRange = IndexRange::forward(m_curFrameIndex, unpackCount);
m_curFrameIndex += unpackCount;
return ReadableSampleFrames(
resultRange,
SampleBuffer::ReadableSlice(
writableSampleFrames.writableData(),
frames2samples(unpackCount)));
}
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;
}
EmErrorCode WvDecoder::DecodeUnit(char* data, uint32_t& used, uint32_t& unitCount)
{
if (m_UnitIndex < m_UnitCount) {
unitCount = 1;
// according to wavpack's source code cli/wvunpack.c
uint32_t nsamples = WavpackUnpackSamples(m_Ctx, (int32_t*)&m_Buf[0], unitCount*m_Channels);
format_samples(m_BytesPerSample, (unsigned char*)data, &m_Buf[0], nsamples*m_Channels);
used = nsamples * m_BytesPerSample * m_Channels;
m_UnitIndex += unitCount;
m_BitRate = WavpackGetInstantBitrate(m_Ctx)/1000;
return ErrorCode::Ok;
}
used = 0;
unitCount = m_UnitCount;
m_BitRate = 0;
return ErrorCode::DecoderOutOfRange;
}
/* this is called for each file to process */
enum codec_status codec_run(void)
{
WavpackContext *wpc;
char error [80];
/* rockbox: comment 'set but unused' variables
int bps;
*/
int nchans, sr_100;
intptr_t param;
if (codec_init())
return CODEC_ERROR;
ci->seek_buffer (ci->id3->offset);
/* Create a decoder instance */
wpc = WavpackOpenFileInput (read_callback, error);
if (!wpc)
return CODEC_ERROR;
ci->configure(DSP_SWITCH_FREQUENCY, WavpackGetSampleRate (wpc));
codec_set_replaygain(ci->id3);
/* bps = WavpackGetBytesPerSample (wpc); */
nchans = WavpackGetReducedChannels (wpc);
ci->configure(DSP_SET_STEREO_MODE, nchans == 2 ? STEREO_INTERLEAVED : STEREO_MONO);
sr_100 = ci->id3->frequency / 100;
ci->set_elapsed (WavpackGetSampleIndex (wpc) / sr_100 * 10);
/* The main decoder loop */
while (1) {
int32_t nsamples;
enum codec_command_action action = ci->get_command(¶m);
if (action == CODEC_ACTION_HALT)
break;
if (action == CODEC_ACTION_SEEK_TIME) {
int curpos_ms = WavpackGetSampleIndex (wpc) / sr_100 * 10;
int n, d, skip;
if (param > curpos_ms) {
n = param - curpos_ms;
d = ci->id3->length - curpos_ms;
skip = (int)((int64_t)(ci->filesize - ci->curpos) * n / d);
ci->seek_buffer (ci->curpos + skip);
}
else if (curpos_ms != 0) {
n = curpos_ms - param;
d = curpos_ms;
skip = (int)((int64_t) ci->curpos * n / d);
ci->seek_buffer (ci->curpos - skip);
}
wpc = WavpackOpenFileInput (read_callback, error);
if (!wpc)
{
ci->seek_complete();
break;
}
ci->set_elapsed (WavpackGetSampleIndex (wpc) / sr_100 * 10);
ci->seek_complete();
}
nsamples = WavpackUnpackSamples (wpc, temp_buffer, BUFFER_SIZE / nchans);
if (!nsamples)
break;
ci->pcmbuf_insert (temp_buffer, NULL, nsamples);
ci->set_elapsed (WavpackGetSampleIndex (wpc) / sr_100 * 10);
}
return CODEC_OK;
}
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;
}
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;
}
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();
}
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;
}
/*
* 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);
}
}
static int
wv_read (DB_fileinfo_t *_info, char *bytes, int size) {
wvctx_t *info = (wvctx_t *)_info;
int currentsample = WavpackGetSampleIndex (info->ctx);
int samplesize = _info->fmt.channels * _info->fmt.bps / 8;
if (size / samplesize + currentsample > info->endsample) {
size = (info->endsample - currentsample + 1) * samplesize;
trace ("wv: size truncated to %d bytes (%d samples), cursample=%d, endsample=%d\n", size, info->endsample - currentsample + 1, currentsample, info->endsample);
if (size <= 0) {
return 0;
}
}
int initsize = size;
int n;
if (WavpackGetMode (info->ctx) & MODE_FLOAT) {
_info->fmt.is_float = 1;
}
if (_info->fmt.is_float || _info->fmt.bps == 32) {
n = WavpackUnpackSamples (info->ctx, (int32_t *)bytes, size / samplesize);
size -= n * samplesize;
}
else {
int32_t buffer[size/(_info->fmt.bps / 8)];
n = WavpackUnpackSamples (info->ctx, (int32_t *)buffer, size / samplesize);
size -= n * samplesize;
n *= _info->fmt.channels;
// convert from int32 to input (what???)
int32_t *p = buffer;
if (_info->fmt.bps == 16) {
while (n > 0) {
*((int16_t *)bytes) = (int16_t)(*p);
bytes += sizeof (int16_t);
p++;
n--;
}
}
else if (_info->fmt.bps == 8) {
while (n > 0) {
*bytes++ = (char)(*p);
p++;
n--;
}
}
else if (_info->fmt.bps == 24) {
while (n > 0) {
*bytes++ = (*p)&0xff;
*bytes++ = ((*p)&0xff00)>>8;
*bytes++ = ((*p)&0xff0000)>>16;
p++;
n--;
}
}
}
_info->readpos = (float)(WavpackGetSampleIndex (info->ctx)-info->startsample)/WavpackGetSampleRate (info->ctx);
#ifndef TINYWV
deadbeef->streamer_set_bitrate (WavpackGetInstantBitrate (info->ctx) / 1000);
#endif
return initsize-size;
}
static gint
xmms_wavpack_read (xmms_xform_t *xform, xmms_sample_t *buf, gint len,
xmms_error_t *error)
{
xmms_wavpack_data_t *data;
gint mono_samples, samples;
xmms_samples32_t *buf32;
gint i;
g_return_val_if_fail (xform, -1);
data = xmms_xform_private_data_get (xform);
g_return_val_if_fail (data, -1);
switch (data->bits_per_sample) {
case 8:
mono_samples = len;
xmms_wavpack_ensure_buf (data, mono_samples);
buf32 = data->buf;
break;
case 12:
case 16:
mono_samples = len / 2;
xmms_wavpack_ensure_buf (data, mono_samples);
buf32 = data->buf;
break;
case 24:
mono_samples = len / 4;
buf32 = buf;
break;
case 32:
mono_samples = len / 4;
buf32 = buf;
break;
}
samples = mono_samples / data->channels;
samples = WavpackUnpackSamples (data->ctx, buf32, samples);
mono_samples = samples * data->channels;
switch (data->bits_per_sample) {
case 8:
len = mono_samples;
for (i=0; i<mono_samples; ++i) {
((xmms_samples8_t *) buf)[i] = data->buf[i];
}
break;
case 12:
len = mono_samples * 2;
for (i=0; i<mono_samples; ++i) {
((xmms_samples16_t *) buf)[i] = data->buf[i] << 4;
}
break;
case 16:
len = mono_samples * 2;
for (i=0; i<mono_samples; ++i) {
((xmms_samples16_t *) buf)[i] = data->buf[i];
}
break;
case 24:
len = mono_samples * 4;
for (i=0; i<mono_samples; ++i) {
((xmms_samples32_t *) buf)[i] <<= 8;
}
break;
case 32:
len = mono_samples * 4;
break;
}
return len;
}
UInt32 WavPackDecoder::ReadAudio(AudioBufferList *bufferList, UInt32 frameCount)
{
if(!IsOpen() || nullptr == bufferList || bufferList->mNumberBuffers != mFormat.mChannelsPerFrame || 0 == frameCount)
return 0;
// Reset output buffer data size
for(UInt32 i = 0; i < bufferList->mNumberBuffers; ++i)
bufferList->mBuffers[i].mDataByteSize = 0;
UInt32 framesRemaining = frameCount;
UInt32 totalFramesRead = 0;
while(0 < framesRemaining) {
UInt32 framesToRead = std::min(framesRemaining, static_cast<UInt32>(BUFFER_SIZE_FRAMES));
// Wavpack uses "complete" samples (one sample across all channels), i.e. a Core Audio frame
uint32_t samplesRead = WavpackUnpackSamples(mWPC, mBuffer, framesToRead);
if(0 == samplesRead)
break;
// The samples returned are handled differently based on the file's mode
int mode = WavpackGetMode(mWPC);
// Floating point files require no special handling other than deinterleaving
if(MODE_FLOAT & mode) {
float *inputBuffer = reinterpret_cast<float *>(mBuffer);
// Deinterleave the samples
for(UInt32 channel = 0; channel < mFormat.mChannelsPerFrame; ++channel) {
float *floatBuffer = static_cast<float *>(bufferList->mBuffers[channel].mData);
for(UInt32 sample = channel; sample < samplesRead * mFormat.mChannelsPerFrame; sample += mFormat.mChannelsPerFrame)
*floatBuffer++ = inputBuffer[sample];
bufferList->mBuffers[channel].mNumberChannels = 1;
bufferList->mBuffers[channel].mDataByteSize = static_cast<UInt32>(samplesRead * sizeof(float));
}
}
// Lossless files will be handed off as integers
else if(MODE_LOSSLESS & mode) {
// WavPack hands us 32-bit signed ints with the samples low-aligned; shift them to high alignment
UInt32 shift = static_cast<UInt32>(8 * (sizeof(int32_t) - WavpackGetBytesPerSample(mWPC)));
// Deinterleave the 32-bit samples and shift to high-alignment
for(UInt32 channel = 0; channel < mFormat.mChannelsPerFrame; ++channel) {
int32_t *shiftedBuffer = static_cast<int32_t *>(bufferList->mBuffers[channel].mData);
for(UInt32 sample = channel; sample < samplesRead * mFormat.mChannelsPerFrame; sample += mFormat.mChannelsPerFrame)
*shiftedBuffer++ = mBuffer[sample] << shift;
bufferList->mBuffers[channel].mNumberChannels = 1;
bufferList->mBuffers[channel].mDataByteSize = static_cast<UInt32>(samplesRead * sizeof(int32_t));
}
}
// Convert lossy files to float
else {
float scaleFactor = (1 << ((WavpackGetBytesPerSample(mWPC) * 8) - 1));
// Deinterleave the 32-bit samples and convert to float
for(UInt32 channel = 0; channel < mFormat.mChannelsPerFrame; ++channel) {
float *floatBuffer = static_cast<float *>(bufferList->mBuffers[channel].mData);
for(UInt32 sample = channel; sample < samplesRead * mFormat.mChannelsPerFrame; sample += mFormat.mChannelsPerFrame)
*floatBuffer++ = mBuffer[sample] / scaleFactor;
bufferList->mBuffers[channel].mNumberChannels = 1;
bufferList->mBuffers[channel].mDataByteSize = static_cast<UInt32>(samplesRead * sizeof(float));
}
}
totalFramesRead += samplesRead;
framesRemaining -= samplesRead;
}
mCurrentFrame += totalFramesRead;
return totalFramesRead;
}
uint32_t wavpack_buffer_decoder_unpack(wavpack_buffer_decoder* wbd,
int32_t* buffer, uint32_t samples)
{
return WavpackUnpackSamples (wbd->wpc, buffer, samples);
}
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;
}
/* this is the codec entry point */
enum codec_status codec_main(void)
{
WavpackContext *wpc;
char error [80];
int bps, nchans, sr_100;
int retval;
/* Generic codec initialisation */
ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
next_track:
if (codec_init()) {
retval = CODEC_ERROR;
goto exit;
}
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
/* Create a decoder instance */
wpc = WavpackOpenFileInput (read_callback, error);
if (!wpc) {
retval = CODEC_ERROR;
goto done;
}
ci->configure(DSP_SWITCH_FREQUENCY, WavpackGetSampleRate (wpc));
codec_set_replaygain(ci->id3);
bps = WavpackGetBytesPerSample (wpc);
nchans = WavpackGetReducedChannels (wpc);
ci->configure(DSP_SET_STEREO_MODE, nchans == 2 ? STEREO_INTERLEAVED : STEREO_MONO);
sr_100 = ci->id3->frequency / 100;
ci->set_elapsed (0);
/* The main decoder loop */
while (1) {
int32_t nsamples;
if (ci->seek_time && ci->taginfo_ready && ci->id3->length) {
ci->seek_time--;
int curpos_ms = WavpackGetSampleIndex (wpc) / sr_100 * 10;
int n, d, skip;
if (ci->seek_time > curpos_ms) {
n = ci->seek_time - curpos_ms;
d = ci->id3->length - curpos_ms;
skip = (int)((int64_t)(ci->filesize - ci->curpos) * n / d);
ci->seek_buffer (ci->curpos + skip);
}
else {
n = curpos_ms - ci->seek_time;
d = curpos_ms;
skip = (int)((int64_t) ci->curpos * n / d);
ci->seek_buffer (ci->curpos - skip);
}
wpc = WavpackOpenFileInput (read_callback, error);
ci->seek_complete();
if (!wpc)
break;
ci->set_elapsed (WavpackGetSampleIndex (wpc) / sr_100 * 10);
ci->yield ();
}
nsamples = WavpackUnpackSamples (wpc, temp_buffer, BUFFER_SIZE / nchans);
if (!nsamples || ci->stop_codec || ci->new_track)
break;
ci->yield ();
if (ci->stop_codec || ci->new_track)
break;
ci->pcmbuf_insert (temp_buffer, NULL, nsamples);
ci->set_elapsed (WavpackGetSampleIndex (wpc) / sr_100 * 10);
ci->yield ();
}
retval = CODEC_OK;
done:
if (ci->request_next_track())
goto next_track;
exit:
return retval;
}
static GstFlowReturn
gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
{
GstWavpackDec *dec;
GstBuffer *outbuf = NULL;
GstFlowReturn ret = GST_FLOW_OK;
WavpackHeader wph;
int32_t decoded, unpacked_size;
gboolean format_changed;
gint width, depth, i, j, max;
gint32 *dec_data = NULL;
guint8 *out_data;
GstMapInfo map, omap;
dec = GST_WAVPACK_DEC (bdec);
g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
gst_buffer_map (buf, &map, GST_MAP_READ);
/* check input, we only accept framed input with complete chunks */
if (map.size < sizeof (WavpackHeader))
goto input_not_framed;
if (!gst_wavpack_read_header (&wph, map.data))
goto invalid_header;
if (map.size < wph.ckSize + 4 * 1 + 4)
goto input_not_framed;
if (!(wph.flags & INITIAL_BLOCK))
goto input_not_framed;
dec->wv_id.buffer = map.data;
dec->wv_id.length = map.size;
dec->wv_id.position = 0;
/* create a new wavpack context if there is none yet but if there
* was already one (i.e. caps were set on the srcpad) check whether
* the new one has the same caps */
if (!dec->context) {
gchar error_msg[80];
dec->context = WavpackOpenFileInputEx (dec->stream_reader,
&dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
/* expect this to work */
if (!dec->context) {
GST_WARNING_OBJECT (dec, "Couldn't decode buffer: %s", error_msg);
goto context_failed;
}
}
g_assert (dec->context != NULL);
format_changed =
(dec->sample_rate != WavpackGetSampleRate (dec->context)) ||
(dec->channels != WavpackGetNumChannels (dec->context)) ||
(dec->depth != WavpackGetBytesPerSample (dec->context) * 8) ||
(dec->channel_mask != WavpackGetChannelMask (dec->context));
if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (dec)) ||
format_changed) {
gint channel_mask;
dec->sample_rate = WavpackGetSampleRate (dec->context);
dec->channels = WavpackGetNumChannels (dec->context);
dec->depth = WavpackGetBytesPerSample (dec->context) * 8;
channel_mask = WavpackGetChannelMask (dec->context);
if (channel_mask == 0)
channel_mask = gst_wavpack_get_default_channel_mask (dec->channels);
dec->channel_mask = channel_mask;
gst_wavpack_dec_negotiate (dec);
/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
* is decoded or after the format has changed */
gst_wavpack_dec_post_tags (dec);
}
/* alloc output buffer */
dec_data = g_malloc (4 * wph.block_samples * dec->channels);
/* decode */
decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples);
if (decoded != wph.block_samples)
goto decode_error;
unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels;
outbuf = gst_buffer_new_and_alloc (unpacked_size);
/* legacy; pass along offset, whatever that might entail */
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET (buf);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = omap.data;
width = dec->width;
depth = dec->depth;
max = dec->channels * wph.block_samples;
if (width == 8) {
gint8 *outbuffer = (gint8 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint8) (dec_data[i + reorder_map[j]]);
}
} else if (width == 16) {
gint16 *outbuffer = (gint16 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint16) (dec_data[i + reorder_map[j]]);
}
} else if (dec->width == 32) {
gint32 *outbuffer = (gint32 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
if (width != depth) {
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ =
(gint32) (dec_data[i + reorder_map[j]] << (width - depth));
}
} else {
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint32) (dec_data[i + reorder_map[j]]);
}
}
} else {
g_assert_not_reached ();
}
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unmap (buf, &map);
buf = NULL;
g_free (dec_data);
ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
out:
if (buf)
gst_buffer_unmap (buf, &map);
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
}
return ret;
/* ERRORS */
input_not_framed:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
ret = GST_FLOW_ERROR;
goto out;
}
invalid_header:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
ret = GST_FLOW_ERROR;
goto out;
}
context_failed:
{
GST_AUDIO_DECODER_ERROR (bdec, 1, LIBRARY, INIT, (NULL),
("error creating Wavpack context"), ret);
goto out;
}
decode_error:
{
const gchar *reason = "unknown";
if (dec->context) {
reason = WavpackGetErrorMessage (dec->context);
} else {
reason = "couldn't create decoder context";
}
GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
("decoding error: %s", reason), ret);
g_free (dec_data);
if (ret == GST_FLOW_OK)
gst_audio_decoder_finish_frame (bdec, NULL, 1);
goto out;
}
}
/*
* 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);
}
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;
}
static GstFlowReturn
gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
{
GstWavpackDec *dec;
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
WavpackHeader wph;
int32_t decoded, unpacked_size;
gboolean format_changed;
dec = GST_WAVPACK_DEC (GST_PAD_PARENT (pad));
/* check input, we only accept framed input with complete chunks */
if (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader))
goto input_not_framed;
if (!gst_wavpack_read_header (&wph, GST_BUFFER_DATA (buf)))
goto invalid_header;
if (GST_BUFFER_SIZE (buf) < wph.ckSize + 4 * 1 + 4)
goto input_not_framed;
if (!(wph.flags & INITIAL_BLOCK))
goto input_not_framed;
dec->wv_id.buffer = GST_BUFFER_DATA (buf);
dec->wv_id.length = GST_BUFFER_SIZE (buf);
dec->wv_id.position = 0;
/* create a new wavpack context if there is none yet but if there
* was already one (i.e. caps were set on the srcpad) check whether
* the new one has the same caps */
if (!dec->context) {
gchar error_msg[80];
dec->context = WavpackOpenFileInputEx (dec->stream_reader,
&dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
if (!dec->context) {
GST_WARNING ("Couldn't decode buffer: %s", error_msg);
dec->error_count++;
if (dec->error_count <= WAVPACK_DEC_MAX_ERRORS) {
goto out; /* just return OK for now */
} else {
goto decode_error;
}
}
}
g_assert (dec->context != NULL);
dec->error_count = 0;
format_changed =
(dec->sample_rate != WavpackGetSampleRate (dec->context)) ||
(dec->channels != WavpackGetNumChannels (dec->context)) ||
(dec->depth != WavpackGetBitsPerSample (dec->context)) ||
#ifdef WAVPACK_OLD_API
(dec->channel_mask != dec->context->config.channel_mask);
#else
(dec->channel_mask != WavpackGetChannelMask (dec->context));
#endif
if (!GST_PAD_CAPS (dec->srcpad) || format_changed) {
GstCaps *caps;
gint channel_mask;
dec->sample_rate = WavpackGetSampleRate (dec->context);
dec->channels = WavpackGetNumChannels (dec->context);
dec->depth = WavpackGetBitsPerSample (dec->context);
caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, dec->sample_rate,
"channels", G_TYPE_INT, dec->channels,
"depth", G_TYPE_INT, dec->depth,
"width", G_TYPE_INT, 32,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE, NULL);
#ifdef WAVPACK_OLD_API
channel_mask = dec->context->config.channel_mask;
#else
channel_mask = WavpackGetChannelMask (dec->context);
#endif
if (channel_mask == 0)
channel_mask = gst_wavpack_get_default_channel_mask (dec->channels);
dec->channel_mask = channel_mask;
/* Only set the channel layout for more than two channels
* otherwise things break unfortunately */
if (channel_mask != 0 && dec->channels > 2)
if (!gst_wavpack_set_channel_layout (caps, channel_mask))
GST_WARNING_OBJECT (dec, "Failed to set channel layout");
GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
/* should always succeed */
gst_pad_set_caps (dec->srcpad, caps);
gst_caps_unref (caps);
/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
* is decoded or after the format has changed */
gst_wavpack_dec_post_tags (dec);
}
/* alloc output buffer */
unpacked_size = 4 * wph.block_samples * dec->channels;
ret = gst_pad_alloc_buffer (dec->srcpad, GST_BUFFER_OFFSET (buf),
unpacked_size, GST_PAD_CAPS (dec->srcpad), &outbuf);
if (ret != GST_FLOW_OK)
goto out;
gst_buffer_copy_metadata (outbuf, buf, GST_BUFFER_COPY_TIMESTAMPS);
/* If we got a DISCONT buffer forward the flag. Nothing else
* has to be done as libwavpack doesn't store state between
* Wavpack blocks */
if (GST_BUFFER_IS_DISCONT (buf) || dec->next_block_index != wph.block_index)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
dec->next_block_index = wph.block_index + wph.block_samples;
/* decode */
decoded = WavpackUnpackSamples (dec->context,
(int32_t *) GST_BUFFER_DATA (outbuf), wph.block_samples);
if (decoded != wph.block_samples)
goto decode_error;
if ((outbuf = gst_audio_buffer_clip (outbuf, &dec->segment,
dec->sample_rate, 4 * dec->channels))) {
GST_LOG_OBJECT (dec, "pushing buffer with time %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
ret = gst_pad_push (dec->srcpad, outbuf);
}
out:
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
}
gst_buffer_unref (buf);
return ret;
/* ERRORS */
input_not_framed:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
invalid_header:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
decode_error:
{
const gchar *reason = "unknown";
if (dec->context) {
#ifdef WAVPACK_OLD_API
reason = dec->context->error_message;
#else
reason = WavpackGetErrorMessage (dec->context);
#endif
} else {
reason = "couldn't create decoder context";
}
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("Failed to decode wavpack stream: %s", reason));
gst_buffer_unref (outbuf);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
