//
// Read up samples from madSynth_
// If needed, read some more MP3 data, decode them and synth them
// Place in audioBuffer_.
// Return number of samples read.
//
size_t MadDecoder::decode(size_t numPendingTotal, AudioBuffer* buffer)
{
size_t numProcessedTotal = 0;
int numChannels = getNumChannels();
mad_fixed_t sample;
do {
size_t numPendingFrame = (madSynth_.pcm.length - currentFrame_) * numChannels;
numPendingFrame = std::min(numPendingTotal, numPendingFrame);
size_t numProcessedFrame = 0;
while (numProcessedFrame < numPendingFrame)
{
for (int channel = 0; channel < numChannels; channel++)
{
sample = madSynth_.pcm.samples[channel][currentFrame_];
if (sample < -MAD_F_ONE)
sample = -MAD_F_ONE;
else if (sample >= MAD_F_ONE)
sample = MAD_F_ONE - 1;
float fSample = (float)(sample / (float)(1L << MAD_F_FRACBITS));
float* pos = buffer->getHead() + numProcessedTotal;
*pos = fSample;
numProcessedFrame++;
numProcessedTotal++;
}
currentFrame_++;
}
numPendingTotal -= numProcessedFrame;
if (numPendingTotal == 0)
break;
if (madStream_.error == MAD_ERROR_BUFLEN) { // check whether input buffer needs a refill
if (readMpgFile() == false) // eof
break;
}
if (mad_frame_decode(&madFrame_, &madStream_))
{
if (MAD_RECOVERABLE(madStream_.error)) {
consumeId3Tag();
continue;
}
else {
if (madStream_.error == MAD_ERROR_BUFLEN)
continue;
else
THROW(std::exception, "unrecoverable frame level error (%s).", mad_stream_errorstr(&madStream_));
}
}
mad_timer_add(&madTimer_, madFrame_.header.duration);
mad_synth_frame(&madSynth_, &madFrame_);
currentFrame_ = 0;
numMpegFrames_++;
} while (true);
return numProcessedTotal;
}
static int mp3_decode_frame (mp3_s *mp3)
{
if (mp3->stream.buffer == NULL || mp3->stream.error == MAD_ERROR_BUFLEN) {
if (mp3_fill_input (mp3) < 0) {
return DECODE_BREAK;
}
}
if (mad_frame_decode (&mp3->frame, &mp3->stream)) {
if (MAD_RECOVERABLE (mp3->stream.error)) {
return DECODE_CONT;
} else {
if (mp3->stream.error == MAD_ERROR_BUFLEN) {
return DECODE_CONT;
} else {
printf ("Unrecoverable frame error: \'%s\'\n", mad_stream_errorstr (&mp3->stream));
mp3->status = 1;
return DECODE_BREAK;
}
}
}
mp3->frame_count++;
return DECODE_OK;
}
void decode() {
// While we need to fill the buffer...
while (
(mad_frame_decode(&frame, &stream) == -1) &&
((stream.error == MAD_ERROR_BUFLEN) || (stream.error == MAD_ERROR_BUFPTR))
)
{
// Fill up the remainder of the file buffer.
int tmp;
tmp = fillFileBuffer();
if (tmp==2) {
endofstream = 1;
//MusicPlayerNextSong();
/*!
*
* Put here the function that does something when the file has ended.
*
*/
}
// Give new buffer to the stream.
mad_stream_buffer(&stream, fileBuffer, sizeof(fileBuffer));
}
// Add to the timer the stream duration
mad_timer_add(&timer, frame.header.duration);
// Synth the frame.
mad_synth_frame(&synth, &frame);
}
static int read_frame(sh_audio_t *sh){
mad_decoder_t *this = sh->context;
int len;
while((len=demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
sh->a_in_buffer_size-sh->a_in_buffer_len))>0){
sh->a_in_buffer_len+=len;
while(1){
int ret;
mad_stream_buffer (&this->stream, sh->a_in_buffer, sh->a_in_buffer_len);
ret=mad_frame_decode (&this->frame, &this->stream);
if (this->stream.next_frame) {
int num_bytes =
(char*)sh->a_in_buffer+sh->a_in_buffer_len - (char*)this->stream.next_frame;
memmove(sh->a_in_buffer, this->stream.next_frame, num_bytes);
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"libmad: %d bytes processed\n",sh->a_in_buffer_len-num_bytes);
sh->a_in_buffer_len = num_bytes;
}
if (ret == 0) return 1; // OK!!!
// error! try to resync!
if(this->stream.error==MAD_ERROR_BUFLEN) break;
}
}
mp_msg(MSGT_DECAUDIO,MSGL_INFO,"Cannot sync MAD frame\n");
return 0;
}
size_t MadDecoder::start(FILE* handle)
{
handle_ = handle;
decodeBuffer_.resize(bufferSize_, false);
ASSERT(decodeBuffer_.size() == bufferSize_);
unsigned char* buffer = decodeBuffer_.getHead();
int64 durationMsec = getDurationMs(buffer, bufferSize_);
mad_stream_init(&madStream_);
mad_frame_init(&madFrame_);
mad_synth_init(&madSynth_);
mad_timer_reset(&madTimer_);
// Decode at least one valid frame to find out the input format.
// The decoded frame will be saved off so that it can be processed later.
//
size_t bytesRead = fread(buffer, (size_t)1, bufferSize_, handle_);
if (bytesRead != bufferSize_ && ferror(handle_))
THROW(std::exception, "%s", strerror(errno));
mad_stream_buffer(&madStream_, buffer, bytesRead);
// Find a valid frame before starting up.
// This make sure that we have a valid MP3
// and also skips past ID3v2 tags at the beginning of the audio file.
//
madStream_.error = MAD_ERROR_NONE;
while (mad_frame_decode(&madFrame_, &madStream_))
{
// check whether input buffer needs a refill
if (madStream_.error == MAD_ERROR_BUFLEN) {
if (readMpgFile() == false) break; // eof
else continue;
}
consumeId3Tag(); // consume any ID3 tags
// FIXME: We should probably detect when we've read
// a bunch of non-ID3 data and still haven't found a
// frame. In that case we can abort early without
// scanning the whole file.
//
madStream_.error = MAD_ERROR_NONE;
}
if (madStream_.error) {
THROW(std::exception, "No valid MP3 frame found");
}
mad_timer_add(&madTimer_, madFrame_.header.duration);
mad_synth_frame(&madSynth_, &madFrame_);
//unsigned int precision_ = 16;
currentFrame_ = 0;
numMpegFrames_ = 0;
initialized_ = true;
return (size_t)(durationMsec * .001 * getSampleRate() + .5); // number of sample frames
}
/*****************************************************************************
* DoWork: decode an MPEG audio frame.
*****************************************************************************/
static void DoWork( filter_t * p_filter,
block_t * p_in_buf, block_t * p_out_buf )
{
filter_sys_t *p_sys = p_filter->p_sys;
p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
p_out_buf->i_buffer = p_in_buf->i_nb_samples * sizeof(vlc_fixed_t) *
aout_FormatNbChannels( &p_filter->fmt_out.audio );
/* Do the actual decoding now. */
mad_stream_buffer( &p_sys->mad_stream, p_in_buf->p_buffer,
p_in_buf->i_buffer );
if ( mad_frame_decode( &p_sys->mad_frame, &p_sys->mad_stream ) == -1 )
{
msg_Dbg( p_filter, "libmad error: %s",
mad_stream_errorstr( &p_sys->mad_stream ) );
p_sys->i_reject_count = 3;
}
else if( p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY )
{
p_sys->i_reject_count = 3;
}
if( p_sys->i_reject_count > 0 )
{
memset( p_out_buf->p_buffer, 0, p_out_buf->i_buffer );
p_sys->i_reject_count--;
return;
}
mad_synth_frame( &p_sys->mad_synth, &p_sys->mad_frame );
struct mad_pcm * p_pcm = &p_sys->mad_synth.pcm;
unsigned int i_samples = p_pcm->length;
mad_fixed_t const * p_left = p_pcm->samples[0];
mad_fixed_t const * p_right = p_pcm->samples[1];
mad_fixed_t * p_samples = (mad_fixed_t *)p_out_buf->p_buffer;
assert( i_samples == p_out_buf->i_nb_samples );
/* Interleave and keep buffers in mad_fixed_t format */
if ( p_pcm->channels == 2 )
{
while ( i_samples-- )
{
*p_samples++ = *p_left++;
*p_samples++ = *p_right++;
}
}
else
{
assert( p_pcm->channels == 1 );
memcpy( p_samples, p_left, i_samples * sizeof(mad_fixed_t) );
}
}
/* Returns 1 if a recoverable error occured, 0 else. */
int mad_js_decode_frame(madfile_t *mf)
{
int dec;
dec = mad_frame_decode(&mf->frame, &mf->stream);
if (dec) {
if (MAD_RECOVERABLE(mf->stream.error)
|| mf->stream.error == MAD_ERROR_BUFLEN) {
return 1;
} else {
_mad_js_raise(mad_stream_errorstr(&mf->stream));
}
}
mad_synth_frame(&mf->synth, &mf->frame);
return 0;
}
static enum mp3_action
decodeNextFrame(struct mp3_data *data)
{
if ((data->stream).buffer == NULL
|| (data->stream).error == MAD_ERROR_BUFLEN) {
if (!mp3_fill_buffer(data))
return DECODE_BREAK;
}
if (mad_frame_decode(&data->frame, &data->stream)) {
#ifdef HAVE_ID3TAG
if ((data->stream).error == MAD_ERROR_LOSTSYNC) {
signed long tagsize = id3_tag_query((data->stream).
this_frame,
(data->stream).
bufend -
(data->stream).
this_frame);
if (tagsize > 0) {
mad_stream_skip(&(data->stream), tagsize);
return DECODE_CONT;
}
}
#endif
if (MAD_RECOVERABLE((data->stream).error)) {
return DECODE_SKIP;
} else {
if ((data->stream).error == MAD_ERROR_BUFLEN)
return DECODE_CONT;
else {
g_warning("unrecoverable frame level error "
"(%s).\n",
mad_stream_errorstr(&data->stream));
return DECODE_BREAK;
}
}
}
return DECODE_OK;
}
bool madaudiofile::getNextAudioFrame() {
#ifdef HAVE_MAD_H
#ifdef MADAUDIO_PRINT_DEBUG
printf("Now in getNextAudioFrame()\n");
#endif
u_int8_t *OutputPtr = OutputBuffer;
u_int8_t *GuardPtr = NULL;
while(OutputPtr < OutputBufferEnd) {
/* The input bucket must be filled if it becomes empty or if
* it's the first execution of the loop.
*/
if(Stream->buffer==NULL || Stream->error==MAD_ERROR_BUFLEN ||
frame_count == 0) {
size_t ReadSize,Remaining;
unsigned char *ReadStart;
/* {2} libmad may not consume all bytes of the input
* buffer. If the last frame in the buffer is not wholly
* contained by it, then that frame's start is pointed by
* the next_frame member of the Stream structure. This
* common situation occurs when mad_frame_decode() fails,
* sets the stream error code to MAD_ERROR_BUFLEN, and
* sets the next_frame pointer to a non NULL value. (See
* also the comment marked {4} bellow.)
*
* When this occurs, the remaining unused bytes must be
* put back at the beginning of the buffer and taken in
* account before refilling the buffer. This means that
* the input buffer must be large enough to hold a whole
* frame at the highest observable bit-rate (currently 448
* kb/s). XXX=XXX Is 2016 bytes the size of the largest
* frame? (448000*(1152/32000))/8
*/
if(Stream->next_frame!=NULL) {
Remaining=Stream->bufend-Stream->next_frame;
memmove(InputBuffer,Stream->next_frame,Remaining);
ReadStart=InputBuffer+Remaining;
ReadSize=INPUT_BUFFER_SIZE-Remaining;
}
else
ReadSize=INPUT_BUFFER_SIZE,
ReadStart=InputBuffer,
Remaining=0;
/* Fill-in the buffer. If an error occurs print a message
* and leave the decoding loop. If the end of stream is
* reached we also leave the loop but the return status is
* left untouched.
*/
ReadSize=readData(ReadStart,1,ReadSize,file);
if(ReadSize<=0) {
break; // probably end of file reached and nothing was read
}
/* {3} When decoding the last frame of a file, it must be
* followed by MAD_BUFFER_GUARD zero bytes if one wants to
* decode that last frame. When the end of file is
* detected we append that quantity of bytes at the end of
* the available data. Note that the buffer can't overflow
* as the guard size was allocated but not used the the
* buffer management code. (See also the comment marked
* {1}.)
*
* In a message to the mad-dev mailing list on May 29th,
* 2001, Rob Leslie explains the guard zone as follows:
*
* "The reason for MAD_BUFFER_GUARD has to do with the
* way decoding is performed. In Layer III, Huffman
* decoding may inadvertently read a few bytes beyond
* the end of the buffer in the case of certain invalid
* input. This is not detected until after the fact. To
* prevent this from causing problems, and also to
* ensure the next frame's main_data_begin pointer is
* always accessible, MAD requires MAD_BUFFER_GUARD
* (currently 8) bytes to be present in the buffer past
* the end of the current frame in order to decode the
* frame."
*/
if(reachedEOF) {
GuardPtr=ReadStart+ReadSize;
memset(GuardPtr,0,MAD_BUFFER_GUARD);
ReadSize+=MAD_BUFFER_GUARD;
}
/* Pipe the new buffer content to libmad's stream decoder
* facility.
*/
mad_stream_buffer(Stream,InputBuffer,ReadSize+Remaining);
Stream->error=(mad_error)0;
}
/* Decode the next MPEG frame. The streams is read from the
* buffer, its constituents are break down and stored the the
* Frame structure, ready for examination/alteration or PCM
* synthesis. Decoding options are carried in the Frame
* structure from the Stream structure.
*
* Error handling: mad_frame_decode() returns a non zero value
* when an error occurs. The error condition can be checked in
* the error member of the Stream structure. A mad error is
* recoverable or fatal, the error status is checked with the
* MAD_RECOVERABLE macro.
*
* {4} When a fatal error is encountered all decoding
* activities shall be stopped, except when a MAD_ERROR_BUFLEN
* is signaled. This condition means that the
* mad_frame_decode() function needs more input to complete
* its work. One shall refill the buffer and repeat the
* mad_frame_decode() call. Some bytes may be left unused at
* the end of the buffer if those bytes forms an incomplete
* frame. Before refilling, the remaining bytes must be moved
* to the beginning of the buffer and used for input for the
* next mad_frame_decode() invocation. (See the comments
* marked {2} earlier for more details.)
*
* Recoverable errors are caused by malformed bit-streams, in
* this case one can call again mad_frame_decode() in order to
* skip the faulty part and re-sync to the next frame.
*/
if(mad_frame_decode(Frame,Stream)) {
if(MAD_RECOVERABLE(Stream->error)) {
/* Do not print a message if the error is a loss of
* synchronization and this loss is due to the end of
* stream guard bytes. (See the comments marked {3}
* supra for more informations about guard bytes.)
*/
if(Stream->error!=MAD_ERROR_LOSTSYNC ||
Stream->this_frame!=GuardPtr) {
}
continue;
}
else
if(Stream->error==MAD_ERROR_BUFLEN)
continue; // end of buffer reached --> read more from the file
else {
fprintf(stderr,
"madaudiofile: unrecoverable frame level error (%i).\n",
Stream->error);
break;
}
}
if(frame_count == 0) {
samplerate = Frame->header.samplerate;
#ifdef MADAUDIO_PRINT_DEBUG
printf("Initially setting samplerate to %i\n",samplerate);
#endif
}
#ifdef MADAUDIO_PRINT_DEBUG
if(Frame->header.samplerate != samplerate)
printf("Obs: samplerate changed to %i!\n",Frame->header.samplerate);
#endif
/* Accounting. The computed frame duration is in the frame
* header structure. It is expressed as a fixed point number
* whole data type is mad_timer_t. It is different from the
* samples fixed point format and unlike it, it can't directly
* be added or subtracted. The timer module provides several
* functions to operate on such numbers. Be careful there, as
* some functions of libmad's timer module receive some of
* their mad_timer_t arguments by value!
*/
frame_count++;
mad_timer_add(Timer,Frame->header.duration);
/* Once decoded the frame is synthesized to PCM samples. No errors
* are reported by mad_synth_frame();
*/
mad_synth_frame(Synth,Frame);
/* Synthesized samples must be converted from libmad's fixed
* point number to the consumer format. Here we use unsigned
* 16 bit big endian integers on two channels. Integer samples
* are temporarily stored in a buffer that is flushed when
* full.
*/
int unsigned sr = 0;
for(int i=0;i<Synth->pcm.length;i++) {
signed short Sample;
u_int8_t left[2];
u_int8_t right[2];
/* Left channel */
Sample=MadFixedToSshort(Synth->pcm.samples[0][i]);
left[0]=Sample>>8;
left[1]=Sample&0xff;
/* Right channel. If the decoded stream is monophonic then
* the right output channel is the same as the left one.
*/
if(MAD_NCHANNELS(&Frame->header)==2)
Sample=MadFixedToSshort(Synth->pcm.samples[1][i]);
right[0]=Sample>>8;
right[1]=Sample&0xff;
while(sr < samplerate) {
/* Left channel */
*(OutputPtr++)=left[1];
*(OutputPtr++)=left[0];
/* Right channel. If the decoded stream is monophonic then
* the right output channel is the same as the left one.
*/
*(OutputPtr++)=right[1];
*(OutputPtr++)=right[0];
sr += Synth->pcm.samplerate;
}
sr -= samplerate;
}
if (OutputPtr>=OutputBufferEnd+OUTPUT_BUFFER_SAFETY) {
char cstr[200];
sprintf(cstr,"getNextAudioFrame() writing outside of OutputBuffer:"
" %p >= %p VERY BAD!!!\n", OutputPtr,
OutputBufferEnd+OUTPUT_BUFFER_SAFETY);
throw(string(cstr));
}
} // while(OutputPtr < OutputBufferEnd)
outputLength = OutputPtr-OutputBuffer;
outputPosition = 0;
if (outputLength == 0)
reachedEOF = true;
// If we have breaked out of the loop, but it wasn't because of EOF,
// return false.
if (OutputPtr < OutputBufferEnd && !reachedEOF)
return false;
return true;
#else
cerr << "libmad not available" << endl;
return false;
#endif // HAVE_MAD_H
}
static gint
xmms_mad_read (xmms_xform_t *xform, gpointer buf, gint len, xmms_error_t *err)
{
xmms_mad_data_t *data;
xmms_samples16_t *out = (xmms_samples16_t *)buf;
gint ret;
gint j;
gint read = 0;
data = xmms_xform_private_data_get (xform);
j = 0;
while (read < len) {
/* use already synthetized frame first */
if (data->synthpos < data->synth.pcm.length) {
out[j++] = scale_linear (data->synth.pcm.samples[0][data->synthpos]);
if (data->channels == 2) {
out[j++] = scale_linear (data->synth.pcm.samples[1][data->synthpos]);
read += 2 * xmms_sample_size_get (XMMS_SAMPLE_FORMAT_S16);
} else {
read += xmms_sample_size_get (XMMS_SAMPLE_FORMAT_S16);
}
data->synthpos++;
continue;
}
/* then try to decode another frame */
if (mad_frame_decode (&data->frame, &data->stream) != -1) {
/* mad_synthpop_frame - go Depeche! */
mad_synth_frame (&data->synth, &data->frame);
if (data->frames_to_skip) {
data->frames_to_skip--;
data->synthpos = 0x7fffffff;
} else if (data->samples_to_skip) {
if (data->samples_to_skip > data->synth.pcm.length) {
data->synthpos = 0x7fffffff;
data->samples_to_skip -= data->synth.pcm.length;
} else {
data->synthpos = data->samples_to_skip;
data->samples_to_skip = 0;
}
} else {
if (data->samples_to_play == 0) {
return read;
} else if (data->samples_to_play > 0) {
if (data->synth.pcm.length > data->samples_to_play) {
data->synth.pcm.length = data->samples_to_play;
}
data->samples_to_play -= data->synth.pcm.length;
}
data->synthpos = 0;
}
continue;
}
/* if there is no frame to decode stream more data */
if (data->stream.next_frame) {
guchar *buffer = data->buffer;
const guchar *nf = data->stream.next_frame;
memmove (data->buffer, data->stream.next_frame,
data->buffer_length = (&buffer[data->buffer_length] - nf));
}
ret = xmms_xform_read (xform,
(gchar *)data->buffer + data->buffer_length,
4096 - data->buffer_length,
err);
if (ret <= 0) {
return ret;
}
data->buffer_length += ret;
mad_stream_buffer (&data->stream, data->buffer, data->buffer_length);
}
return read;
}
static void mp_render_ex (void *dest, unsigned nsamp)
{
short *sout = (short *) dest;
int localerrors = 0;
if (!mp_playing || mp_paused)
{
memset (dest, 0, nsamp * 4);
return;
}
while (1)
{
// write any leftover data from last MP3 frame
while (mp_leftoversamps > 0 && nsamp > 0)
{
short s = mp_fixtoshort (Synth.pcm.samples[0][mp_leftoversamppos]);
*sout++ = s;
if (Synth.pcm.channels == 2)
s = mp_fixtoshort (Synth.pcm.samples[1][mp_leftoversamppos]);
// if mono, just duplicate the first channel again
*sout++ = s;
mp_leftoversamps -= 1;
mp_leftoversamppos += 1;
nsamp -= 1;
}
if (nsamp == 0)
return; // done
// decode next valid MP3 frame
while (mad_frame_decode (&Frame, &Stream) != 0)
{
if (MAD_RECOVERABLE (Stream.error))
{ // unspecified problem with one frame.
// try the next frame, but bail if we get a bunch of crap in a row;
// likely indicates a larger problem (and if we don't bail, we could
// spend arbitrarily long amounts of time looking for the next good
// packet)
localerrors++;
if (localerrors == 10)
{
lprintf (LO_WARN, "mad_frame_decode: Lots of errors. Most recent %s\n", mad_stream_errorstr (&Stream));
mp_playing = 0;
memset (sout, 0, nsamp * 4);
return;
}
}
else if (Stream.error == MAD_ERROR_BUFLEN)
{ // EOF
// FIXME: in order to not drop the last frame, there must be at least MAD_BUFFER_GUARD
// of extra bytes (with value 0) at the end of the file. current implementation
// drops last frame
if (mp_looping)
{ // rewind, then go again
mad_stream_buffer (&Stream, mp_data, mp_len);
continue;
}
else
{ // stop
mp_playing = 0;
memset (sout, 0, nsamp * 4);
return;
}
}
else
{ // oh well.
lprintf (LO_WARN, "mad_frame_decode: Unrecoverable error %s\n", mad_stream_errorstr (&Stream));
mp_playing = 0;
memset (sout, 0, nsamp * 4);
return;
}
}
// got a good frame, so synth it and dispatch it.
mad_synth_frame (&Synth, &Frame);
mp_leftoversamps = Synth.pcm.length;
mp_leftoversamppos = 0;
}
// NOT REACHED
}
// MP3 decode player
void MP3Player(void)
{
FRESULT res;
uint8_t *ReadStart;
uint8_t *GuardPtr;
volatile uint8_t u8PCMBufferTargetIdx = 0;
volatile uint32_t pcmbuf_idx, i;
volatile unsigned int Mp3FileOffset=0;
uint16_t sampleL, sampleR;
pcmbuf_idx = 0;
memset((void *)&audioInfo, 0, sizeof(audioInfo));
/* Parse MP3 header */
MP3_ParseHeaderInfo(MP3_FILE);
/* First the structures used by libmad must be initialized. */
mad_stream_init(&Stream);
mad_frame_init(&Frame);
mad_synth_init(&Synth);
/* Open MP3 file */
res = f_open(&mp3FileObject, MP3_FILE, FA_OPEN_EXISTING | FA_READ);
if (res != FR_OK) {
printf("Open file error \r\n");
return;
}
/* Open I2S1 interface and set to slave mode, stereo channel, I2S format */
I2S_Open(I2S1, I2S_MODE_SLAVE, 16000, I2S_DATABIT_16, I2S_STEREO, I2S_FORMAT_I2S, I2S_I2S);
/* Initialize WAU8822 codec */
WAU8822_Setup();
/* Configure wau8822 for specific sampling rate */
WAU8822_ConfigSampleRate(audioInfo.mp3SampleRate);
/* Set MCLK and enable MCLK */
I2S_EnableMCLK(I2S1, 12000000);
while(1) {
if(Stream.buffer==NULL || Stream.error==MAD_ERROR_BUFLEN) {
if(Stream.next_frame != NULL) {
/* Get the remaining frame */
Remaining = Stream.bufend - Stream.next_frame;
memmove(MadInputBuffer, Stream.next_frame, Remaining);
ReadStart = MadInputBuffer + Remaining;
ReadSize = FILE_IO_BUFFER_SIZE - Remaining;
} else {
ReadSize = FILE_IO_BUFFER_SIZE,
ReadStart = MadInputBuffer,
Remaining = 0;
}
/* read the file from SDCard */
res = f_read(&mp3FileObject, ReadStart, ReadSize, &ReturnSize);
if((res != FR_OK) || f_eof(&mp3FileObject)) {
printf("Stop !(%x)\n\r", res);
goto stop;
}
/* if the file is over */
if (ReadSize > ReturnSize) {
GuardPtr=ReadStart+ReadSize;
memset(GuardPtr,0,MAD_BUFFER_GUARD);
ReadSize+=MAD_BUFFER_GUARD;
}
Mp3FileOffset = Mp3FileOffset + ReturnSize;
/* Pipe the new buffer content to libmad's stream decoder
* facility.
*/
mad_stream_buffer(&Stream,MadInputBuffer,ReadSize+Remaining);
Stream.error=(enum mad_error)0;
}
/* decode a frame from the mp3 stream data */
if(mad_frame_decode(&Frame,&Stream)) {
if(MAD_RECOVERABLE(Stream.error)) {
/*if(Stream.error!=MAD_ERROR_LOSTSYNC ||
Stream.this_frame!=GuardPtr)
{
}*/
continue;
} else {
/* the current frame is not full, need to read the remaining part */
if(Stream.error==MAD_ERROR_BUFLEN) {
continue;
} else {
printf("Something error!!\n");
/* play the next file */
audioInfo.mp3FileEndFlag = 1;
goto stop;
}
}
}
/* Once decoded the frame is synthesized to PCM samples. No errors
* are reported by mad_synth_frame();
*/
mad_synth_frame(&Synth,&Frame);
//
// decode finished, try to copy pcm data to audio buffer
//
if(audioInfo.mp3Playing) {
//if next buffer is still full (playing), wait until it's empty
if(aPCMBuffer_Full[u8PCMBufferTargetIdx] == 1)
while(aPCMBuffer_Full[u8PCMBufferTargetIdx]);
} else {
if((aPCMBuffer_Full[0] == 1) && (aPCMBuffer_Full[1] == 1 )) { //all buffers are full, wait
StartPlay();
}
}
for(i=0; i<(int)Synth.pcm.length; i++) {
/* Get the left/right samples */
sampleL = Synth.pcm.samples[0][i];
sampleR = Synth.pcm.samples[1][i];
/* Fill PCM data to I2S(PDMA) buffer */
aPCMBuffer[u8PCMBufferTargetIdx][pcmbuf_idx++] = sampleR | (sampleL << 16);
/* Need change buffer ? */
if(pcmbuf_idx == PCM_BUFFER_SIZE) {
aPCMBuffer_Full[u8PCMBufferTargetIdx] = 1; //set full flag
u8PCMBufferTargetIdx ^= 1;
pcmbuf_idx = 0;
// printf("change to ==>%d ..\n", u8PCMBufferTargetIdx);
/* if next buffer is still full (playing), wait until it's empty */
if((aPCMBuffer_Full[u8PCMBufferTargetIdx] == 1) && (audioInfo.mp3Playing))
while(aPCMBuffer_Full[u8PCMBufferTargetIdx]);
}
}
}
stop:
printf("Exit MP3\r\n");
mad_synth_finish(&Synth);
mad_frame_finish(&Frame);
mad_stream_finish(&Stream);
f_close(&mp3FileObject);
StopPlay();
}
/*
* NAME: scan_header()
* DESCRIPTION: read the initial frame(s) to get stream statistics
*/
static
int scan_header(CPs_InStream* pInStream, struct mad_header *header, struct xing *xing)
{
struct mad_stream stream;
struct mad_frame frame;
unsigned char buffer[8192];
unsigned int buflen = 0;
int count = 0, result = 0;
mad_stream_init(&stream);
mad_frame_init(&frame);
if (xing)
xing->flags = 0;
while (1)
{
if (buflen < sizeof(buffer))
{
// DWORD bytes;
unsigned int bytes;
if (pInStream->Read(pInStream, buffer + buflen, sizeof(buffer) - buflen, &bytes) == FALSE
|| bytes == 0)
{
result = -1;
break;
}
buflen += bytes;
}
mad_stream_buffer(&stream, buffer, buflen);
while (1)
{
if (mad_frame_decode(&frame, &stream) == -1)
{
if (!MAD_RECOVERABLE(stream.error))
break;
continue;
}
if (count++ ||
(xing && parse_xing(xing, stream.anc_ptr,
stream.anc_bitlen) == 0))
break;
}
if (count || stream.error != MAD_ERROR_BUFLEN)
break;
memmove(buffer, stream.next_frame,
buflen = &buffer[buflen] - stream.next_frame);
}
if (count)
{
if (header)
*header = frame.header;
}
else
result = -1;
mad_frame_finish(&frame);
mad_stream_finish(&stream);
return result;
}
static OMX_ERRORTYPE
decode_buffer (const void * ap_obj)
{
mp3d_prc_t * p_obj = (mp3d_prc_t *) ap_obj;
unsigned char * p_guardzone = NULL;
int status = 0;
assert (p_obj->p_outhdr_);
/* Check if there is any remaining PCM data from a previous run of the
* decoding loop that needs to be synthesised */
if ((0 != p_obj->next_synth_sample_
&& p_obj->next_synth_sample_ < p_obj->synth_.pcm.length)
&& (p_obj->p_outhdr_->nFilledLen < p_obj->p_outhdr_->nAllocLen))
{
p_obj->next_synth_sample_
= synthesize_samples (p_obj, p_obj->next_synth_sample_);
}
while (p_obj->p_outhdr_)
{
/* The input bucket must be filled if it becomes empty or if
* it's the first execution of the loop.
*/
if ((NULL == p_obj->stream_.buffer
|| MAD_ERROR_BUFLEN == p_obj->stream_.error)
&& p_obj->p_inhdr_ != NULL)
{
size_t read_size = 0;
unsigned char * p_read_start = NULL;
p_obj->remaining_ = 0;
if (p_obj->stream_.next_frame != NULL)
{
p_obj->remaining_
= p_obj->stream_.bufend - p_obj->stream_.next_frame;
memmove (p_obj->in_buff_, p_obj->stream_.next_frame,
p_obj->remaining_);
p_read_start = p_obj->in_buff_ + p_obj->remaining_;
read_size = INPUT_BUFFER_SIZE - p_obj->remaining_;
}
else
{
read_size = INPUT_BUFFER_SIZE;
p_read_start = p_obj->in_buff_;
p_obj->remaining_ = 0;
}
/* Fill-in the buffer. If an error occurs print a message
* and leave the decoding loop. If the end of stream is
* reached we also leave the loop but the return status is
* left untouched.
*/
read_size = read_from_omx_buffer (p_obj, p_read_start, read_size,
p_obj->p_inhdr_);
if (read_size == 0)
{
if ((p_obj->p_inhdr_->nFlags & OMX_BUFFERFLAG_EOS) != 0)
{
TIZ_TRACE (handleOf (p_obj), "end of input stream");
status = 2;
}
else
{
TIZ_TRACE (handleOf (p_obj), "read_size <= 0");
status = 1;
}
break;
}
/* TODO */
/* if(BstdFileEofP (BstdFile)) */
/* { */
/* p_guardzone = p_read_start + read_size; */
/* memset (p_guardzone, 0, MAD_BUFFER_GUARD); */
/* read_size += MAD_BUFFER_GUARD; */
/* } */
/* Pipe the new buffer content to libmad's stream decoder
* facility.
*/
mad_stream_buffer (&p_obj->stream_, p_obj->in_buff_,
read_size + p_obj->remaining_);
p_obj->stream_.error = 0;
}
if (mad_frame_decode (&p_obj->frame_, &p_obj->stream_) == -1)
{
if (MAD_RECOVERABLE (p_obj->stream_.error))
{
if (p_obj->stream_.error != MAD_ERROR_LOSTSYNC
|| p_obj->stream_.this_frame != p_guardzone)
{
TIZ_TRACE (handleOf (p_obj),
"recoverable frame level error (%s)",
mad_stream_errorstr (&p_obj->stream_));
}
continue;
}
else
{
if (p_obj->stream_.error == MAD_ERROR_BUFLEN)
{
if (!p_obj->p_inhdr_)
{
TIZ_TRACE (handleOf (p_obj),
"p_obj->stream_.error==MAD_ERROR_BUFLEN "
"p_obj->p_inhdr_=[NULL]");
break;
}
else
{
TIZ_TRACE (handleOf (p_obj),
"p_obj->stream_.error==MAD_ERROR_BUFLEN "
"p_obj->p_inhdr_=[%p] nFilledLen [%d]",
p_obj->p_inhdr_, p_obj->p_inhdr_->nFilledLen);
continue;
}
}
else
{
TIZ_TRACE (handleOf (p_obj),
"unrecoverable frame level error (%s).",
mad_stream_errorstr (&p_obj->stream_));
status = 2;
break;
}
}
}
/* The characteristics of the stream's first frame is printed The first
* frame is representative of the entire stream.
*/
if (0 == p_obj->frame_count_)
{
store_stream_metadata (p_obj, &(p_obj->frame_.header));
}
p_obj->frame_count_++;
mad_timer_add (&p_obj->timer_, p_obj->frame_.header.duration);
/* Once decoded the frame is synthesized to PCM samples. No errors
* are reported by mad_synth_frame();
*/
mad_synth_frame (&p_obj->synth_, &p_obj->frame_);
p_obj->next_synth_sample_
= synthesize_samples (p_obj, p_obj->next_synth_sample_);
}
(void) status;
/* TODO */
/* if (p_obj->p_outhdr_ != NULL */
/* && p_obj->p_outhdr_->nFilledLen != 0 */
/* && status == 2) */
/* { */
/* const tiz_srv_t *p_parent = ap_obj; */
/* void *p_krn = tiz_get_krn (p_parent->p_hdl_); */
/* p_obj->eos_ = true; */
/* p_obj->p_outhdr_->nFlags |= OMX_BUFFERFLAG_EOS; */
/* TIZ_LOG (TIZ_PRIORITY_TRACE, handleOf (p_obj), */
/* "Releasing output buffer [%p] ..." */
/* "nFilledLen = [%d] OMX_BUFFERFLAG_EOS", */
/* p_obj->p_outhdr_, p_obj->p_outhdr_->nFilledLen); */
/* tiz_krn_release_buffer (tiz_get_krn (handleOf (ap_obj)), 1,
* p_obj->p_outhdr_); */
/* /\* p_obj->p_outhdr_ = NULL; *\/ */
/* } */
return OMX_ErrorNone;
}
void CodecMpeg1::process(const QByteArray &data,bool is_last)
{
#ifdef HAVE_LIBMAD
float pcm[32768*4];
int frame_offset=0;
int err_count=0;
mpeg1_mpeg.append(data);
mad_stream_buffer(&mpeg1_mad_stream,(const unsigned char *)mpeg1_mpeg.data(),
mpeg1_mpeg.length());
do {
while(mad_frame_decode(&mpeg1_mad_frame,&mpeg1_mad_stream)==0) {
mad_synth_frame(&mpeg1_mad_synth,&mpeg1_mad_frame);
for(int i=0;i<mpeg1_mad_synth.pcm.length;i++) {
for(int j=0;j<mpeg1_mad_synth.pcm.channels;j++) {
pcm[frame_offset+i*mpeg1_mad_synth.pcm.channels+j]=
(float)mad_f_todouble(mpeg1_mad_synth.pcm.samples[j][i]);
}
}
frame_offset+=(mpeg1_mad_synth.pcm.length*mpeg1_mad_synth.pcm.channels);
}
if(MAD_RECOVERABLE(mpeg1_mad_stream.error)!=0) {
if(err_count++>10) {
Reset();
break;
}
}
} while(mpeg1_mad_stream.error!=MAD_ERROR_BUFLEN);
if(mpeg1_mad_stream.error==MAD_ERROR_BUFLEN) {
if(frame_offset>0) {
mpeg1_mad_header=mpeg1_mad_frame.header;
if(!isFramed()) {
int channels=2;
if(mpeg1_mad_frame.header.mode==MAD_MODE_SINGLE_CHANNEL) {
channels=1;
}
setFramed(channels,mpeg1_mad_frame.header.samplerate,
mpeg1_mad_frame.header.bitrate/1000);
}
else {
writePcm(pcm,frame_offset/channels(),is_last);
}
}
mpeg1_mpeg=
mpeg1_mpeg.right(mpeg1_mad_stream.bufend-mpeg1_mad_stream.next_frame);
}
if(is_last) {
frame_offset=0;
mpeg1_mpeg.append(0,MAD_BUFFER_GUARD);
mad_stream_buffer(&mpeg1_mad_stream,(const unsigned char *)mpeg1_mpeg.data(),
mpeg1_mpeg.length());
do {
while(mad_frame_decode(&mpeg1_mad_frame,&mpeg1_mad_stream)==0) {
mad_synth_frame(&mpeg1_mad_synth,&mpeg1_mad_frame);
for(int i=0;i<mpeg1_mad_synth.pcm.length;i++) {
for(int j=0;j<mpeg1_mad_synth.pcm.channels;j++) {
pcm[frame_offset+i*mpeg1_mad_synth.pcm.channels+j]=
(float)mad_f_todouble(mpeg1_mad_synth.pcm.samples[j][i]);
}
}
frame_offset+=(mpeg1_mad_synth.pcm.length*mpeg1_mad_synth.pcm.channels);
}
if(MAD_RECOVERABLE(mpeg1_mad_stream.error)!=0) {
if(err_count++>10) {
Reset();
break;
}
}
} while(mpeg1_mad_stream.error!=MAD_ERROR_BUFLEN);
writePcm(pcm,frame_offset/channels(),is_last);
}
#endif // HAVE_LIBMAD
}
/* Read up to len samples from p->Synth
* If needed, read some more MP3 data, decode them and synth them
* Place in buf[].
* Return number of samples read. */
static int mp3_decode(snd_stream_t *stream, byte *buf, int len)
{
mp3_priv_t *p = (mp3_priv_t *) stream->priv;
int donow, i, done = 0;
mad_fixed_t sample;
int chan, x;
do
{
x = (p->Synth.pcm.length - p->cursamp) * stream->info.channels;
donow = min(len, x);
i = 0;
while (i < donow)
{
for (chan = 0; chan < stream->info.channels; chan++)
{
sample = p->Synth.pcm.samples[chan][p->cursamp];
/* convert from fixed to short,
* write in host-endian format. */
if (sample <= -MAD_F_ONE)
sample = -0x7FFF;
else if (sample >= MAD_F_ONE)
sample = 0x7FFF;
else
sample >>= (MAD_F_FRACBITS + 1 - 16);
if (host_bigendian)
{
*buf++ = (sample >> 8) & 0xFF;
*buf++ = sample & 0xFF;
}
else /* assumed LITTLE_ENDIAN. */
{
*buf++ = sample & 0xFF;
*buf++ = (sample >> 8) & 0xFF;
}
i++;
}
p->cursamp++;
}
len -= donow;
done += donow;
if (len == 0)
break;
/* check whether input buffer needs a refill */
if (p->Stream.error == MAD_ERROR_BUFLEN)
{
if (mp3_inputdata(stream) == -1)
{
/* check feof() ?? */
Con_DPrintf("mp3 EOF\n");
break;
}
}
if (mad_frame_decode(&p->Frame, &p->Stream))
{
if (MAD_RECOVERABLE(p->Stream.error))
{
mp3_inputtag(stream);
continue;
}
else
{
if (p->Stream.error == MAD_ERROR_BUFLEN)
continue;
else
{
Con_Printf("MP3: unrecoverable frame level error (%s)\n",
mad_stream_errorstr(&p->Stream));
break;
}
}
}
p->FrameCount++;
mad_timer_add(&p->Timer, p->Frame.header.duration);
mad_synth_frame(&p->Synth, &p->Frame);
p->cursamp = 0;
} while (1);
int Mp3Decoder::Read(u8 * buffer, int buffer_size, int pos UNUSED)
{
if(!file_fd)
return -1;
if(Format == VOICE_STEREO_16BIT)
buffer_size &= ~0x0003;
else
buffer_size &= ~0x0001;
u8 * write_pos = buffer;
u8 * write_end = buffer+buffer_size;
while(1)
{
while(SynthPos < Synth.pcm.length)
{
if(write_pos >= write_end)
return write_pos-buffer;
*((s16 *) write_pos) = FixedToShort(Synth.pcm.samples[0][SynthPos]);
write_pos += 2;
if(MAD_NCHANNELS(&Frame.header) == 2)
{
*((s16 *) write_pos) = FixedToShort(Synth.pcm.samples[1][SynthPos]);
write_pos += 2;
}
SynthPos++;
}
if(Stream.buffer == NULL || Stream.error == MAD_ERROR_BUFLEN)
{
u8 * ReadStart = ReadBuffer;
int ReadSize = SoundBlockSize*SoundBlocks;
int Remaining = 0;
if(Stream.next_frame != NULL)
{
Remaining = Stream.bufend - Stream.next_frame;
memmove(ReadBuffer, Stream.next_frame, Remaining);
ReadStart += Remaining;
ReadSize -= Remaining;
}
ReadSize = file_fd->read(ReadStart, ReadSize);
if(ReadSize <= 0)
{
GuardPtr = ReadStart;
memset(GuardPtr, 0, MAD_BUFFER_GUARD);
ReadSize = MAD_BUFFER_GUARD;
}
CurPos += ReadSize;
mad_stream_buffer(&Stream, ReadBuffer, Remaining+ReadSize);
}
if(mad_frame_decode(&Frame,&Stream))
{
if(MAD_RECOVERABLE(Stream.error))
{
if(Stream.error != MAD_ERROR_LOSTSYNC || !GuardPtr)
continue;
}
else
{
if(Stream.error != MAD_ERROR_BUFLEN)
return -1;
else if(Stream.error == MAD_ERROR_BUFLEN && GuardPtr)
return -1;
}
}
mad_timer_add(&Timer,Frame.header.duration);
mad_synth_frame(&Synth,&Frame);
SynthPos = 0;
}
return 0;
}
int
mp3_mad_stream_frame (mp3_info_t *info) {
int eof = 0;
while (!eof && (info->mad_stream.buffer == NULL || info->buffer.decode_remaining <= 0)) {
// read more MPEG data if needed
if(info->mad_stream.buffer==NULL || info->mad_stream.error==MAD_ERROR_BUFLEN) {
// copy part of last frame to beginning
if (info->mad_stream.next_frame && info->mad_stream.bufend <= info->mad_stream.next_frame) {
eof = 1;
break;
}
if (info->mad_stream.next_frame != NULL) {
info->buffer.remaining = info->mad_stream.bufend - info->mad_stream.next_frame;
memmove (info->buffer.input, info->mad_stream.next_frame, info->buffer.remaining);
}
int size = READBUFFER - info->buffer.remaining;
int bytesread = 0;
uint8_t *bytes = info->buffer.input + info->buffer.remaining;
bytesread = deadbeef->fread (bytes, 1, size, info->buffer.file);
if (!bytesread) {
// add guard
eof = 1;
memset (bytes, 0, 8);
bytesread = 8;
}
if (bytesread < size) {
// end of file
size -= bytesread;
bytes += bytesread;
}
bytesread += info->buffer.remaining;
mad_stream_buffer(&info->mad_stream,info->buffer.input,bytesread);
if (info->mad_stream.buffer==NULL) {
// check sync bits
if (bytes[0] != 0xff || (bytes[1]&(3<<5)) != (3<<5)) {
trace ("mp3: read didn't start at frame boundary!\ncmp3_scan_stream is broken\n");
}
else {
trace ("mp3: streambuffer=NULL\n");
}
}
}
info->mad_stream.error=0;
// decode next frame
if(mad_frame_decode(&info->mad_frame, &info->mad_stream))
{
if(MAD_RECOVERABLE(info->mad_stream.error))
{
if(info->mad_stream.error!=MAD_ERROR_LOSTSYNC) {
// printf ("mp3: recoverable frame level error (%s)\n", MadErrorString(&info->stream));
}
if (info->buffer.lead_in_frames > 0) {
info->buffer.lead_in_frames--;
}
continue;
}
else {
if(info->mad_stream.error==MAD_ERROR_BUFLEN) {
// printf ("mp3: recoverable frame level error (%s)\n", MadErrorString(&info->stream));
continue;
}
else
{
// printf ("mp3: unrecoverable frame level error (%s).\n", MadErrorString(&info->stream));
return -1; // fatal error
}
}
}
mad_synth_frame(&info->mad_synth,&info->mad_frame);
if (info->buffer.lead_in_frames > 0) {
info->buffer.lead_in_frames--;
info->buffer.decode_remaining = 0;
continue;
}
info->info.fmt.samplerate = info->mad_frame.header.samplerate;
// synthesize single frame
info->buffer.decode_remaining = info->mad_synth.pcm.length;
deadbeef->streamer_set_bitrate (info->mad_frame.header.bitrate/1000);
break;
}
return eof;
}
SINT SoundSourceMp3::readSampleFrames(
SINT numberOfFrames, CSAMPLE* sampleBuffer,
SINT sampleBufferSize, bool readStereoSamples) {
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(getSampleBufferSize(numberOfFrames, readStereoSamples) <= sampleBufferSize);
const SINT numberOfFramesTotal = math_min(
numberOfFrames, getMaxFrameIndex() - m_curFrameIndex);
CSAMPLE* pSampleBuffer = sampleBuffer;
SINT numberOfFramesRemaining = numberOfFramesTotal;
while (0 < numberOfFramesRemaining) {
if (0 >= m_madSynthCount) {
// When all decoded output data has been consumed...
DEBUG_ASSERT(0 == m_madSynthCount);
// ...decode the next MP3 frame
DEBUG_ASSERT(m_madStream.buffer);
DEBUG_ASSERT(m_madStream.this_frame);
// WARNING: Correctly evaluating and handling the result
// of mad_frame_decode() has proven to be extremely tricky.
// Don't change anything at the following lines of code
// unless you know what you are doing!!!
unsigned char const* pMadThisFrame = m_madStream.this_frame;
if (mad_frame_decode(&m_madFrame, &m_madStream)) {
// Something went wrong when decoding the frame...
if (MAD_ERROR_BUFLEN == m_madStream.error) {
// Abort
break;
}
if (isUnrecoverableError(m_madStream)) {
qWarning() << "Unrecoverable MP3 frame decoding error:"
<< mad_stream_errorstr(&m_madStream);
// Abort decoding
break;
}
if (isRecoverableError(m_madStream)) {
if (pMadThisFrame != m_madStream.this_frame) {
// Ignore all recoverable errors (and especially
// "lost synchronization" warnings) while skipping
// over prefetched frames after seeking.
if (pSampleBuffer) {
qWarning() << "Recoverable MP3 frame decoding error:"
<< mad_stream_errorstr(&m_madStream);
} else {
// Decoded samples will simply be discarded
qDebug() << "Recoverable MP3 frame decoding error while skipping:"
<< mad_stream_errorstr(&m_madStream);
}
}
// Continue decoding
}
}
if (pMadThisFrame == m_madStream.this_frame) {
qDebug() << "Retry decoding MP3 frame @" << m_curFrameIndex;
// Retry decoding
continue;
}
DEBUG_ASSERT(isStreamValid(m_madStream));
#ifndef QT_NO_DEBUG_OUTPUT
const SINT madFrameChannelCount = MAD_NCHANNELS(&m_madFrame.header);
if (madFrameChannelCount != getChannelCount()) {
qDebug() << "MP3 frame header with mismatching number of channels"
<< madFrameChannelCount << "<>" << getChannelCount();
}
#endif
// Once decoded the frame is synthesized to PCM samples
mad_synth_frame(&m_madSynth, &m_madFrame);
#ifndef QT_NO_DEBUG_OUTPUT
const SINT madSynthSampleRate = m_madSynth.pcm.samplerate;
if (madSynthSampleRate != getSamplingRate()) {
qDebug() << "Reading MP3 data with different sampling rate"
<< madSynthSampleRate << "<>" << getSamplingRate();
}
#endif
m_madSynthCount = m_madSynth.pcm.length;
DEBUG_ASSERT(0 < m_madSynthCount);
}
const SINT synthReadCount = math_min(
m_madSynthCount, numberOfFramesRemaining);
if (pSampleBuffer) {
DEBUG_ASSERT(m_madSynthCount <= m_madSynth.pcm.length);
const SINT madSynthOffset =
m_madSynth.pcm.length - m_madSynthCount;
DEBUG_ASSERT(madSynthOffset < m_madSynth.pcm.length);
const SINT madSynthChannelCount = m_madSynth.pcm.channels;
DEBUG_ASSERT(0 < madSynthChannelCount);
DEBUG_ASSERT(madSynthChannelCount <= getChannelCount());
#ifndef QT_NO_DEBUG_OUTPUT
if (madSynthChannelCount != getChannelCount()) {
qDebug() << "Reading MP3 data with different number of channels"
<< madSynthChannelCount << "<>" << getChannelCount();
}
#endif
if (kChannelCountMono == madSynthChannelCount) {
// MP3 frame contains a mono signal
if (readStereoSamples || (kChannelCountStereo == getChannelCount())) {
// The reader explicitly requested a stereo signal
// or the AudioSource itself provides a stereo signal.
// Mono -> Stereo: Copy 1st channel twice
for (SINT i = 0; i < synthReadCount; ++i) {
const CSAMPLE sampleValue = madScaleSampleValue(
m_madSynth.pcm.samples[0][madSynthOffset + i]);
*pSampleBuffer++ = sampleValue;
*pSampleBuffer++ = sampleValue;
}
} else {
// Mono -> Mono: Copy 1st channel
for (SINT i = 0; i < synthReadCount; ++i) {
const CSAMPLE sampleValue = madScaleSampleValue(
m_madSynth.pcm.samples[0][madSynthOffset + i]);
*pSampleBuffer++ = sampleValue;
}
}
} else {
// MP3 frame contains a stereo signal
DEBUG_ASSERT(kChannelCountStereo == madSynthChannelCount);
// If the MP3 frame contains a stereo signal then the whole
// AudioSource must also provide 2 channels, because the
// maximum channel count of all MP3 frames is used.
DEBUG_ASSERT(kChannelCountStereo == getChannelCount());
// Stereo -> Stereo: Copy 1st + 2nd channel
for (SINT i = 0; i < synthReadCount; ++i) {
*pSampleBuffer++ = madScaleSampleValue(
m_madSynth.pcm.samples[0][madSynthOffset + i]);
*pSampleBuffer++ = madScaleSampleValue(
m_madSynth.pcm.samples[1][madSynthOffset + i]);
}
}
}
// consume decoded output data
m_madSynthCount -= synthReadCount;
m_curFrameIndex += synthReadCount;
numberOfFramesRemaining -= synthReadCount;
}
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(numberOfFramesTotal >= numberOfFramesRemaining);
return numberOfFramesTotal - numberOfFramesRemaining;
}
static void mad_decode_data (audio_decoder_t *this_gen, buf_element_t *buf) {
mad_decoder_t *this = (mad_decoder_t *) this_gen;
int bytes_in_buffer_at_pts;
lprintf ("decode data, size: %d, decoder_flags: %d\n", buf->size, buf->decoder_flags);
if (buf->size>(INPUT_BUF_SIZE-this->bytes_in_buffer)) {
xprintf (this->xstream->xine, XINE_VERBOSITY_DEBUG,
"libmad: ALERT input buffer too small (%d bytes, %d avail)!\n",
buf->size, INPUT_BUF_SIZE-this->bytes_in_buffer);
buf->size = INPUT_BUF_SIZE-this->bytes_in_buffer;
}
if ((buf->decoder_flags & BUF_FLAG_HEADER) == 0) {
/* reset decoder on leaving preview mode */
if ((buf->decoder_flags & BUF_FLAG_PREVIEW) == 0) {
if (this->preview_mode) {
mad_reset (this_gen);
}
} else {
this->preview_mode = 1;
}
bytes_in_buffer_at_pts = this->bytes_in_buffer;
xine_fast_memcpy (&this->buffer[this->bytes_in_buffer],
buf->content, buf->size);
this->bytes_in_buffer += buf->size;
/*
printf ("libmad: decode data - doing it\n");
*/
mad_stream_buffer (&this->stream, this->buffer,
this->bytes_in_buffer);
if (this->bytes_in_buffer < MAD_MIN_SIZE && buf->pts == 0)
return;
if (!this->needs_more_data) {
this->pts = buf->pts;
if (buf->decoder_flags & BUF_FLAG_AUDIO_PADDING) {
this->start_padding = buf->decoder_info[1];
this->end_padding = buf->decoder_info[2];
} else {
this->start_padding = 0;
this->end_padding = 0;
}
}
while (1) {
if (mad_frame_decode (&this->frame, &this->stream) != 0) {
if (this->stream.next_frame) {
int num_bytes =
this->buffer + this->bytes_in_buffer - this->stream.next_frame;
/* printf("libmad: MAD_ERROR_BUFLEN\n"); */
memmove(this->buffer, this->stream.next_frame, num_bytes);
this->bytes_in_buffer = num_bytes;
}
switch (this->stream.error) {
case MAD_ERROR_BUFLEN:
/* libmad wants more data */
this->needs_more_data = 1;
return;
default:
lprintf ("error 0x%04X, mad_stream_buffer %d bytes\n", this->stream.error, this->bytes_in_buffer);
mad_stream_buffer (&this->stream, this->buffer,
this->bytes_in_buffer);
}
} else {
int mode = (this->frame.header.mode == MAD_MODE_SINGLE_CHANNEL) ? AO_CAP_MODE_MONO : AO_CAP_MODE_STEREO;
if (!this->output_open
|| (this->output_sampling_rate != this->frame.header.samplerate)
|| (this->output_mode != mode)) {
lprintf ("audio sample rate %d mode %08x\n", this->frame.header.samplerate, mode);
/* the mpeg audio demuxer can set audio bitrate */
if (! _x_stream_info_get(this->xstream, XINE_STREAM_INFO_AUDIO_BITRATE)) {
_x_stream_info_set(this->xstream, XINE_STREAM_INFO_AUDIO_BITRATE,
this->frame.header.bitrate);
}
/* the mpeg audio demuxer can set this meta info */
if (! _x_meta_info_get(this->xstream, XINE_META_INFO_AUDIOCODEC)) {
switch (this->frame.header.layer) {
case MAD_LAYER_I:
_x_meta_info_set_utf8(this->xstream, XINE_META_INFO_AUDIOCODEC,
"MPEG audio layer 1 (lib: MAD)");
break;
case MAD_LAYER_II:
_x_meta_info_set_utf8(this->xstream, XINE_META_INFO_AUDIOCODEC,
"MPEG audio layer 2 (lib: MAD)");
break;
case MAD_LAYER_III:
_x_meta_info_set_utf8(this->xstream, XINE_META_INFO_AUDIOCODEC,
"MPEG audio layer 3 (lib: MAD)");
break;
default:
_x_meta_info_set_utf8(this->xstream, XINE_META_INFO_AUDIOCODEC,
"MPEG audio (lib: MAD)");
}
}
if (this->output_open) {
this->xstream->audio_out->close (this->xstream->audio_out, this->xstream);
this->output_open = 0;
}
if (!this->output_open) {
this->output_open = (this->xstream->audio_out->open) (this->xstream->audio_out,
this->xstream, 16,
this->frame.header.samplerate,
mode) ;
}
if (!this->output_open) {
return;
}
this->output_sampling_rate = this->frame.header.samplerate;
this->output_mode = mode;
}
mad_synth_frame (&this->synth, &this->frame);
if ( (buf->decoder_flags & BUF_FLAG_PREVIEW) == 0 ) {
unsigned int nchannels, nsamples;
mad_fixed_t const *left_ch, *right_ch;
struct mad_pcm *pcm = &this->synth.pcm;
audio_buffer_t *audio_buffer;
uint16_t *output;
int bitrate;
int pts_offset;
audio_buffer = this->xstream->audio_out->get_buffer (this->xstream->audio_out);
output = audio_buffer->mem;
nchannels = pcm->channels;
nsamples = pcm->length;
left_ch = pcm->samples[0];
right_ch = pcm->samples[1];
/* padding */
if (this->start_padding || this->end_padding) {
/* check padding validity */
if (nsamples < (this->start_padding + this->end_padding)) {
lprintf("invalid padding data");
this->start_padding = 0;
this->end_padding = 0;
}
lprintf("nsamples=%d, start_padding=%d, end_padding=%d\n",
nsamples, this->start_padding, this->end_padding);
nsamples -= this->start_padding + this->end_padding;
left_ch += this->start_padding;
right_ch += this->start_padding;
}
audio_buffer->num_frames = nsamples;
audio_buffer->vpts = this->pts;
while (nsamples--) {
/* output sample(s) in 16-bit signed little-endian PCM */
*output++ = scale(*left_ch++);
if (nchannels == 2)
*output++ = scale(*right_ch++);
}
audio_buffer->num_frames = pcm->length;
/* pts computing */
if (this->frame.header.bitrate > 0) {
bitrate = this->frame.header.bitrate;
} else {
bitrate = _x_stream_info_get(this->xstream, XINE_STREAM_INFO_AUDIO_BITRATE);
lprintf("offset %d bps\n", bitrate);
}
audio_buffer->vpts = buf->pts;
if (audio_buffer->vpts && (bitrate > 0)) {
pts_offset = (bytes_in_buffer_at_pts * 8 * 90) / (bitrate / 1000);
lprintf("pts: %"PRId64", offset: %d pts, %d bytes\n", buf->pts, pts_offset, bytes_in_buffer_at_pts);
if (audio_buffer->vpts < pts_offset)
pts_offset = audio_buffer->vpts;
audio_buffer->vpts -= pts_offset;
}
this->xstream->audio_out->put_buffer (this->xstream->audio_out, audio_buffer, this->xstream);
this->pts = buf->pts;
buf->pts = 0;
if (buf->decoder_flags & BUF_FLAG_AUDIO_PADDING) {
this->start_padding = buf->decoder_info[1];
this->end_padding = buf->decoder_info[2];
buf->decoder_info[1] = 0;
buf->decoder_info[2] = 0;
} else {
this->start_padding = 0;
this->end_padding = 0;
}
}
lprintf ("decode worked\n");
}
}
}
}
/* Reads the next frame from the file. Returns true on success or
false on failure. */
static int
read_next_frame(mad_data *mp3_mad) {
if (mp3_mad->stream.buffer == NULL ||
mp3_mad->stream.error == MAD_ERROR_BUFLEN) {
size_t read_size;
size_t remaining;
unsigned char *read_start;
/* There might be some bytes in the buffer left over from last
time. If so, move them down and read more bytes following
them. */
if (mp3_mad->stream.next_frame != NULL) {
remaining = mp3_mad->stream.bufend - mp3_mad->stream.next_frame;
memmove(mp3_mad->input_buffer, mp3_mad->stream.next_frame, remaining);
read_start = mp3_mad->input_buffer + remaining;
read_size = MAD_INPUT_BUFFER_SIZE - remaining;
} else {
read_size = MAD_INPUT_BUFFER_SIZE;
read_start = mp3_mad->input_buffer;
remaining = 0;
}
/* Now read additional bytes from the input file. */
read_size = SDL_RWread(mp3_mad->src, read_start, 1, read_size);
if (read_size <= 0) {
if ((mp3_mad->status & (MS_input_eof | MS_input_error)) == 0) {
if (read_size == 0) {
mp3_mad->status |= MS_input_eof;
} else {
mp3_mad->status |= MS_input_error;
}
/* At the end of the file, we must stuff MAD_BUFFER_GUARD
number of 0 bytes. */
SDL_memset(read_start + read_size, 0, MAD_BUFFER_GUARD);
read_size += MAD_BUFFER_GUARD;
}
}
/* Now feed those bytes into the libmad stream. */
mad_stream_buffer(&mp3_mad->stream, mp3_mad->input_buffer,
read_size + remaining);
mp3_mad->stream.error = MAD_ERROR_NONE;
}
/* Now ask libmad to extract a frame from the data we just put in
its buffer. */
if (mad_frame_decode(&mp3_mad->frame, &mp3_mad->stream)) {
if (MAD_RECOVERABLE(mp3_mad->stream.error)) {
return 0;
} else if (mp3_mad->stream.error == MAD_ERROR_BUFLEN) {
return 0;
} else {
mp3_mad->status |= MS_decode_error;
return 0;
}
}
mp3_mad->frames_read++;
mad_timer_add(&mp3_mad->next_frame_start, mp3_mad->frame.header.duration);
return 1;
}
/*****************************************************************************
* DoWork: decode an MPEG audio frame.
*****************************************************************************/
static void DoWork( filter_t * p_filter,
aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
{
filter_sys_t *p_sys = p_filter->p_sys;
p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
p_out_buf->i_buffer = p_in_buf->i_nb_samples * sizeof(vlc_fixed_t) *
aout_FormatNbChannels( &p_filter->fmt_out.audio );
/* Do the actual decoding now. */
mad_stream_buffer( &p_sys->mad_stream, p_in_buf->p_buffer,
p_in_buf->i_buffer );
if ( mad_frame_decode( &p_sys->mad_frame, &p_sys->mad_stream ) == -1 )
{
msg_Dbg( p_filter, "libmad error: %s",
mad_stream_errorstr( &p_sys->mad_stream ) );
p_sys->i_reject_count = 3;
}
else if( p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY )
{
p_sys->i_reject_count = 3;
}
if( p_sys->i_reject_count > 0 )
{
if( p_filter->fmt_out.audio.i_format == VLC_CODEC_FL32 )
{
int i;
int i_size = p_out_buf->i_buffer / sizeof(float);
float * a = (float *)p_out_buf->p_buffer;
for ( i = 0 ; i < i_size ; i++ )
*a++ = 0.0;
}
else
{
memset( p_out_buf->p_buffer, 0, p_out_buf->i_buffer );
}
p_sys->i_reject_count--;
return;
}
mad_synth_frame( &p_sys->mad_synth, &p_sys->mad_frame );
struct mad_pcm * p_pcm = &p_sys->mad_synth.pcm;
unsigned int i_samples = p_pcm->length;
mad_fixed_t const * p_left = p_pcm->samples[0];
mad_fixed_t const * p_right = p_pcm->samples[1];
assert( i_samples == p_out_buf->i_nb_samples );
if ( p_filter->fmt_out.audio.i_format == VLC_CODEC_FI32 )
{
/* Interleave and keep buffers in mad_fixed_t format */
mad_fixed_t * p_samples = (mad_fixed_t *)p_out_buf->p_buffer;
if ( p_pcm->channels == 2 )
{
while ( i_samples-- )
{
*p_samples++ = *p_left++;
*p_samples++ = *p_right++;
}
}
else
{
assert( p_pcm->channels == 1 );
vlc_memcpy( p_samples, p_left, i_samples * sizeof(mad_fixed_t) );
}
}
else
{
/* float32 */
float * p_samples = (float *)p_out_buf->p_buffer;
const float f_temp = (float)FIXED32_ONE;
if ( p_pcm->channels == 2 )
{
while ( i_samples-- )
{
*p_samples++ = (float)*p_left++ / f_temp;
*p_samples++ = (float)*p_right++ / f_temp;
}
}
else
{
assert( p_pcm->channels == 1 );
while ( i_samples-- )
{
*p_samples++ = (float)*p_left++ / f_temp;
}
}
}
}
void XMp3Decomp::Decompress(XS16* dstWord, XS32 nSamples)
{
long bytesCleared;
XS32 nBytes = nSamples * snd->BytesPerBlock();
XS8 *dstByte = (XS8 *) dstWord;
// XBOOL seeking = (!dstWord) ? (true) : false;
bytesCleared = GetBufferedData(dstByte, nBytes);
if (!isInitMad())
{
if (dstByte)
{
memset(dstByte, 0x00, nBytes);
}
goto exit_gracefully;
}
else
{
nBytes -= bytesCleared;
// dst byte can be zero during seeking
if (dstByte)
dstByte += bytesCleared;
}
if (m_firstBuf) // first enter
{
// read head. and give data
bool b = fillBufferMad();
assert(b);
m_firstBuf = false;
}
// check first if there is any buffered data
if (bytesCleared == nBytes)
goto exit_gracefully;
mad_error madError;
while(nBytes > 0)
{
int success = mad_frame_decode(&Frame, &Stream);
madError = Stream.error;
if (success == 0)
{
// do success things.
mad_synth_frame(&Synth,&Frame);
struct mad_pcm *pcm = &(Synth.pcm);
int nchannels = pcm->channels;
// int nsamples = pcm->length;
assert(Synth.pcm.length * nchannels * 2 <= kBufSize);
assert(Synth.pcm.length > 0);
XU8* p = pcmBuf;
for (int i = 0; i < Synth.pcm.length; i++)
{
signed int Sample = scale(Synth.pcm.samples[0][i]);
*p++ = Sample & 0xff;
*p++ = (Sample >> 8) & 0xff;
if (nchannels == 2)
{
Sample = scale(Synth.pcm.samples[0][i]);
*p++ = Sample & 0xff;
*p++ = (Sample >> 8) & 0xff;
}
}
bufLength = Synth.pcm.length * nchannels * 2;
bytesCleared = GetBufferedData(dstByte, nBytes);
nBytes -= bytesCleared;
if (dstByte)
{
dstByte += bytesCleared;
}
}
static
int run_sync(struct mad_decoder *decoder)
{
enum mad_flow (*error_func)(void *, struct mad_stream *, struct mad_frame *);
void *error_data;
int bad_last_frame = 0;
struct mad_stream *stream;
struct mad_frame *frame;
struct mad_synth *synth;
int result = 0;
if (decoder->input_func == 0)
return 0;
if (decoder->error_func) {
error_func = decoder->error_func;
error_data = decoder->cb_data;
}
else {
error_func = error_default;
error_data = &bad_last_frame;
}
stream = &decoder->sync->stream;
frame = &decoder->sync->frame;
synth = &decoder->sync->synth;
mad_stream_init(stream);
mad_frame_init(frame);
mad_synth_init(synth);
mad_stream_options(stream, decoder->options);
do {
switch (decoder->input_func(decoder->cb_data, stream)) {
case MAD_FLOW_STOP:
goto done;
case MAD_FLOW_BREAK:
goto fail;
case MAD_FLOW_IGNORE:
continue;
case MAD_FLOW_CONTINUE:
break;
}
while (1) {
if (decoder->header_func) {
if (mad_header_decode(&frame->header, stream) == -1) {
if (!MAD_RECOVERABLE(stream->error))
break;
switch (error_func(error_data, stream, frame)) {
case MAD_FLOW_STOP:
goto done;
case MAD_FLOW_BREAK:
goto fail;
case MAD_FLOW_IGNORE:
case MAD_FLOW_CONTINUE:
default:
continue;
}
}
switch (decoder->header_func(decoder->cb_data, &frame->header)) {
case MAD_FLOW_STOP:
goto done;
case MAD_FLOW_BREAK:
goto fail;
case MAD_FLOW_IGNORE:
continue;
case MAD_FLOW_CONTINUE:
break;
}
}
if (mad_frame_decode(frame, stream) == -1) {
if (!MAD_RECOVERABLE(stream->error))
break;
switch (error_func(error_data, stream, frame)) {
case MAD_FLOW_STOP:
goto done;
case MAD_FLOW_BREAK:
goto fail;
case MAD_FLOW_IGNORE:
break;
case MAD_FLOW_CONTINUE:
default:
continue;
}
}
else
bad_last_frame = 0;
if (decoder->filter_func) {
switch (decoder->filter_func(decoder->cb_data, stream, frame)) {
case MAD_FLOW_STOP:
goto done;
case MAD_FLOW_BREAK:
goto fail;
case MAD_FLOW_IGNORE:
continue;
case MAD_FLOW_CONTINUE:
break;
}
}
mad_synth_frame(synth, frame);
if (decoder->output_func) {
switch (decoder->output_func(decoder->cb_data,
&frame->header, &synth->pcm)) {
case MAD_FLOW_STOP:
goto done;
case MAD_FLOW_BREAK:
goto fail;
case MAD_FLOW_IGNORE:
case MAD_FLOW_CONTINUE:
break;
}
}
}
}
while (stream->error == MAD_ERROR_BUFLEN);
fail:
result = -1;
done:
mad_synth_finish(synth);
mad_frame_finish(frame);
mad_stream_finish(stream);
return result;
}
static gboolean
xmms_mad_init (xmms_xform_t *xform)
{
struct mad_frame frame;
struct mad_stream stream;
xmms_error_t err;
guchar buf[40960];
xmms_mad_data_t *data;
int len;
const gchar *metakey;
g_return_val_if_fail (xform, FALSE);
data = g_new0 (xmms_mad_data_t, 1);
mad_stream_init (&data->stream);
mad_frame_init (&data->frame);
mad_synth_init (&data->synth);
xmms_xform_private_data_set (xform, data);
data->buffer_length = 0;
data->synthpos = 0x7fffffff;
mad_stream_init (&stream);
mad_frame_init (&frame);
len = xmms_xform_peek (xform, buf, 40960, &err);
mad_stream_buffer (&stream, buf, len);
while (mad_frame_decode (&frame, &stream) == -1) {
if (!MAD_RECOVERABLE (stream.error)) {
XMMS_DBG ("couldn't decode %02x %02x %02x %02x",buf[0],buf[1],buf[2],buf[3]);
mad_frame_finish (&frame);
mad_stream_finish (&stream);
return FALSE;
}
}
data->channels = frame.header.mode == MAD_MODE_SINGLE_CHANNEL ? 1 : 2;
data->samplerate = frame.header.samplerate;
if (frame.header.flags & MAD_FLAG_PROTECTION) {
XMMS_DBG ("Frame has protection enabled");
if (stream.anc_ptr.byte > stream.buffer + 2) {
stream.anc_ptr.byte = stream.anc_ptr.byte - 2;
}
}
data->samples_to_play = -1;
data->xing = xmms_xing_parse (stream.anc_ptr);
if (data->xing) {
xmms_xing_lame_t *lame;
XMMS_DBG ("File with Xing header!");
metakey = XMMS_MEDIALIB_ENTRY_PROPERTY_IS_VBR;
xmms_xform_metadata_set_int (xform, metakey, 1);
if (xmms_xing_has_flag (data->xing, XMMS_XING_FRAMES)) {
guint duration;
mad_timer_t timer;
timer = frame.header.duration;
mad_timer_multiply (&timer, xmms_xing_get_frames (data->xing));
duration = mad_timer_count (timer, MAD_UNITS_MILLISECONDS);
XMMS_DBG ("XING duration %d", duration);
metakey = XMMS_MEDIALIB_ENTRY_PROPERTY_DURATION;
xmms_xform_metadata_set_int (xform, metakey, duration);
if (xmms_xing_has_flag (data->xing, XMMS_XING_BYTES) && duration) {
guint tmp;
tmp = xmms_xing_get_bytes (data->xing) * ((guint64)8000) / duration;
XMMS_DBG ("XING bitrate %d", tmp);
metakey = XMMS_MEDIALIB_ENTRY_PROPERTY_BITRATE;
xmms_xform_metadata_set_int (xform, metakey, tmp);
}
}
lame = xmms_xing_get_lame (data->xing);
if (lame) {
/* FIXME: add a check for ignore_lame_headers from the medialib */
data->frames_to_skip = 1;
data->samples_to_skip = lame->start_delay;
data->samples_to_play = ((guint64) xmms_xing_get_frames (data->xing) * 1152ULL) -
lame->start_delay - lame->end_padding;
XMMS_DBG ("Samples to skip in the beginning: %d, total: %" G_GINT64_FORMAT,
data->samples_to_skip, data->samples_to_play);
/*
metakey = XMMS_MEDIALIB_ENTRY_PROPERTY_GAIN_ALBUM;
xmms_xform_metadata_set_int (xform, metakey, lame->audiophile_gain);
metakey = XMMS_MEDIALIB_ENTRY_PROPERTY_PEAK_TRACK;
xmms_xform_metadata_set_int (xform, metakey, lame->peak_amplitude);
metakey = XMMS_MEDIALIB_ENTRY_PROPERTY_GAIN_TRACK;
xmms_xform_metadata_set_int (xform, metakey, lame->radio_gain);
*/
}
} else {
gint filesize;
metakey = XMMS_MEDIALIB_ENTRY_PROPERTY_BITRATE;
xmms_xform_metadata_set_int (xform, metakey, frame.header.bitrate);
metakey = XMMS_MEDIALIB_ENTRY_PROPERTY_DURATION;
if (!xmms_xform_metadata_get_int (xform, metakey, &filesize)) {
metakey = XMMS_MEDIALIB_ENTRY_PROPERTY_SIZE;
if (xmms_xform_metadata_get_int (xform, metakey, &filesize)) {
gint32 val;
val = (gint32) (filesize * (gdouble) 8000.0 / frame.header.bitrate);
metakey = XMMS_MEDIALIB_ENTRY_PROPERTY_DURATION;
xmms_xform_metadata_set_int (xform, metakey, val);
}
}
}
/* seeking needs bitrate */
data->bitrate = frame.header.bitrate;
if (xmms_id3v1_get_tags (xform) < 0) {
mad_stream_finish (&data->stream);
mad_frame_finish (&data->frame);
mad_synth_finish (&data->synth);
if (data->xing) {
xmms_xing_free (data->xing);
}
return FALSE;
}
xmms_xform_outdata_type_add (xform,
XMMS_STREAM_TYPE_MIMETYPE,
"audio/pcm",
XMMS_STREAM_TYPE_FMT_FORMAT,
XMMS_SAMPLE_FORMAT_S16,
XMMS_STREAM_TYPE_FMT_CHANNELS,
data->channels,
XMMS_STREAM_TYPE_FMT_SAMPLERATE,
data->samplerate,
XMMS_STREAM_TYPE_END);
mad_frame_finish (&frame);
mad_stream_finish (&stream);
return TRUE;
}
static int
MP3Callback_22k(void *_buf2, unsigned int numSamples, void *pdata)
{
int i;
short *_buf = (short *)_buf2;
unsigned long samplesOut = 0;
numSamples /= 2;
// Playing , so mix up a buffer
if (samplesInOutput > 0)
{
if (samplesInOutput > numSamples) {
write22k_buf((short *) _buf, (short *) OutputBuffer, numSamples * 2);
samplesOut = numSamples;
samplesInOutput -= numSamples;
} else {
write22k_buf((short *) _buf, (short *) OutputBuffer, samplesInOutput * 2);
samplesOut = samplesInOutput;
samplesInOutput = 0;
}
}
while (samplesOut < numSamples)
{
if (Stream.buffer == NULL) {
Stream.error = 0;
mad_stream_buffer(&Stream, mp3_data, size);
}
if (Stream.error == MAD_ERROR_BUFLEN) {
Stream.error = 0;
if (isRepeatMode) {
MP3_Restart();
mad_stream_buffer(&Stream, mp3_data, size);
samplesInOutput = 0;
} else {
MP3_Stop();
return 0;
}
}
if (mad_frame_decode(&Frame, &Stream)) {
if (MAD_RECOVERABLE(Stream.error)) {
return 0;
} else
if (Stream.error == MAD_ERROR_BUFLEN) {
if (! isRepeatMode) {
MP3_Stop();
return 0;
}
} else {
MP3_Stop();
}
}
FrameCount++;
mad_timer_add(&Timer, Frame.header.duration);
mad_synth_frame(&Synth, &Frame);
for (i = 0; i < Synth.pcm.length; i++) {
signed short Sample;
if (samplesOut < numSamples) {
/* Left channel */
Sample = MadFixedToSshort(Synth.pcm.samples[0][i]);
_buf[((samplesOut * 2) * 2) ] = Sample;
_buf[((samplesOut * 2) * 2) + 1] = Sample;
/* Right channel. If the decoded stream is monophonic then
* the right output channel is the same as the left one.
*/
if (MAD_NCHANNELS(&Frame.header) == 2) {
Sample = MadFixedToSshort(Synth.pcm.samples[1][i]);
}
_buf[(((samplesOut * 2) + 1) * 2) ] = Sample;
_buf[(((samplesOut * 2) + 1) * 2) + 1] = Sample;
samplesOut++;
} else {
Sample = MadFixedToSshort(Synth.pcm.samples[0][i]);
OutputBuffer[samplesInOutput * 2] = Sample;
if (MAD_NCHANNELS(&Frame.header) == 2) {
Sample = MadFixedToSshort(Synth.pcm.samples[1][i]);
}
OutputBuffer[samplesInOutput * 2 + 1] = Sample;
samplesInOutput++;
}
}
}
return isPlaying;
}
void LibMadWrapper::render(pcm_t *const bufferToFill, const uint32_t Channels, frame_t framesToRender)
{
framesToRender = min(framesToRender, this->getFrames() - this->framesAlreadyRendered);
int32_t *pcm = static_cast<int32_t *>(bufferToFill);
// the outer loop, used for decoding and synthesizing MPEG frames
while (framesToRender > 0 && !this->stopFillBuffer)
{
// write back tempbuffer, i.e. frames weve buffered from previous calls to libmad (necessary due to inelegant API of libmad, i.e. cant tell how many frames to render during one call)
{
const size_t itemsToCpy = min<size_t>(this->tempBuf.size(), framesToRender * Channels);
memcpy(pcm, this->tempBuf.data(), itemsToCpy * sizeof(int32_t));
this->tempBuf.erase(this->tempBuf.begin(), this->tempBuf.begin() + itemsToCpy);
const size_t framesCpyd = itemsToCpy / Channels;
framesToRender -= framesCpyd;
this->framesAlreadyRendered += framesCpyd;
// again: adjust position
pcm += itemsToCpy;
}
int framesToDoNow = (framesToRender / gConfig.FramesToRender) > 0 ? gConfig.FramesToRender : framesToRender % gConfig.FramesToRender;
if (framesToDoNow == 0)
{
continue;
}
int ret = mad_frame_decode(&this->frame.Value, this->stream);
if (ret != 0)
{
if (this->stream->error == MAD_ERROR_LOSTSYNC)
{
long tagsize = id3_tag_query(this->stream->this_frame, this->stream->bufend - this->stream->this_frame);
if (tagsize > 0)
{
mad_stream_skip(this->stream, tagsize);
continue;
}
}
string errstr = mad_stream_errorstr(this->stream);
if (MAD_RECOVERABLE(this->stream->error))
{
errstr += " (recoverable)";
CLOG(LogLevel_t::Info, errstr);
continue;
}
else
{
errstr += " (not recoverable)";
CLOG(LogLevel_t::Warning, errstr);
break;
}
}
mad_synth_frame(&this->synth.Value, &this->frame.Value);
/* save PCM samples from synth.pcm */
/* &synth.pcm->samplerate contains the sampling frequency */
unsigned short nsamples = this->synth->pcm.length;
mad_fixed_t const *left_ch = this->synth->pcm.samples[0];
mad_fixed_t const *right_ch = this->synth->pcm.samples[1];
unsigned int item = 0;
/* audio normalization */
const float absoluteGain = (numeric_limits<int32_t>::max()) / (numeric_limits<int32_t>::max() * this->gainCorrection);
for (;
!this->stopFillBuffer &&
framesToDoNow > 0 && // frames left during this loop
framesToRender > 0 && // frames left during this call
nsamples > 0; // frames left from libmad
framesToRender--, nsamples--, framesToDoNow--)
{
int32_t sample;
/* output sample(s) in 24-bit signed little-endian PCM */
sample = LibMadWrapper::toInt24Sample(*left_ch++);
sample = gConfig.useAudioNormalization ? floor(sample * absoluteGain) : sample;
pcm[item++] = sample;
if (Channels == 2) // our buffer is for 2 channels
{
if (this->synth.Value.pcm.channels == 2) // ...but did mad also decoded for 2 channels?
{
sample = LibMadWrapper::toInt24Sample(*right_ch++);
sample = gConfig.useAudioNormalization ? floor(sample * absoluteGain) : sample;
pcm[item++] = sample;
}
else
{
// what? only one channel in a stereo file? well then: pseudo stereo
pcm[item++] = sample;
CLOG(LogLevel_t::Warning, "decoded only one channel, though this is a stereo file!");
}
}
this->framesAlreadyRendered++;
}
pcm += item /* % this->count*/;
// "bufferToFill" (i.e. "pcm") seems to be full, drain the rest pcm samples from libmad and temporarily save them
while (!this->stopFillBuffer && nsamples > 0)
{
int32_t sample;
/* output sample(s) in 24-bit signed little-endian PCM */
sample = LibMadWrapper::toInt24Sample(*left_ch++);
this->tempBuf.push_back(gConfig.useAudioNormalization ? floor(sample * absoluteGain) : sample);
if (Channels == 2)
{
sample = LibMadWrapper::toInt24Sample(*right_ch++);
this->tempBuf.push_back(gConfig.useAudioNormalization ? floor(sample * absoluteGain) : sample);
}
/* DONT do this: this->framesAlreadyRendered++; since we use framesAlreadyRendered as offset for "bufferToFill"*/
nsamples--;
}
if (item > this->count)
{
CLOG(LogLevel_t::Error, "THIS SHOULD NEVER HAPPEN: read " << item << " items but only expected " << this->count << "\n");
break;
}
}
}
/* mp3 playback callback
*/
static WaitressCbReturn_t BarPlayerMp3Cb (void *ptr, size_t size,
void *stream) {
const char *data = ptr;
struct audioPlayer *player = stream;
size_t i;
QUIT_PAUSE_CHECK;
if (!BarPlayerBufferFill (player, data, size)) {
return WAITRESS_CB_RET_ERR;
}
/* some "prebuffering" */
if (player->mode < PLAYER_RECV_DATA &&
player->bufferFilled < BAR_PLAYER_BUFSIZE / 2) {
return WAITRESS_CB_RET_OK;
}
mad_stream_buffer (&player->mp3Stream, player->buffer,
player->bufferFilled);
player->mp3Stream.error = 0;
do {
/* channels * max samples, found in mad.h */
signed short int madDecoded[2*1152], *madPtr = madDecoded;
if (mad_frame_decode (&player->mp3Frame, &player->mp3Stream) != 0) {
if (player->mp3Stream.error != MAD_ERROR_BUFLEN) {
BarUiMsg (player->settings, MSG_ERR,
"mp3 decoding error: %s\n",
mad_stream_errorstr (&player->mp3Stream));
return WAITRESS_CB_RET_ERR;
} else {
/* rebuffering required => exit loop */
break;
}
}
mad_synth_frame (&player->mp3Synth, &player->mp3Frame);
for (i = 0; i < player->mp3Synth.pcm.length; i++) {
/* left channel */
*(madPtr++) = applyReplayGain (BarPlayerMadToShort (
player->mp3Synth.pcm.samples[0][i]), player->scale);
/* right channel */
*(madPtr++) = applyReplayGain (BarPlayerMadToShort (
player->mp3Synth.pcm.samples[1][i]), player->scale);
}
if (player->mode < PLAYER_AUDIO_INITIALIZED) {
ao_sample_format format;
int audioOutDriver;
player->channels = player->mp3Synth.pcm.channels;
player->samplerate = player->mp3Synth.pcm.samplerate;
audioOutDriver = ao_default_driver_id();
memset (&format, 0, sizeof (format));
format.bits = 16;
format.channels = player->channels;
format.rate = player->samplerate;
format.byte_format = AO_FMT_NATIVE;
if ((player->audioOutDevice = ao_open_live (audioOutDriver,
&format, NULL)) == NULL) {
player->aoError = 1;
BarUiMsg (player->settings, MSG_ERR,
"Cannot open audio device\n");
return WAITRESS_CB_RET_ERR;
}
/* calc song length using the framerate of the first decoded frame */
player->songDuration = (unsigned long long int) player->waith.request.contentLength /
((unsigned long long int) player->mp3Frame.header.bitrate /
(unsigned long long int) BAR_PLAYER_MS_TO_S_FACTOR / 8LL);
/* must be > PLAYER_SAMPLESIZE_INITIALIZED, otherwise time won't
* be visible to user (ugly, but mp3 decoding != aac decoding) */
player->mode = PLAYER_RECV_DATA;
}
/* samples * length * channels */
ao_play (player->audioOutDevice, (char *) madDecoded,
player->mp3Synth.pcm.length * 2 * 2);
/* avoid division by 0 */
if (player->mode == PLAYER_RECV_DATA) {
/* same calculation as in aac player; don't need to divide by
* channels, length is number of samples for _one_ channel */
player->songPlayed += (unsigned long long int) player->mp3Synth.pcm.length *
(unsigned long long int) BAR_PLAYER_MS_TO_S_FACTOR /
(unsigned long long int) player->samplerate;
}
QUIT_PAUSE_CHECK;
} while (player->mp3Stream.error != MAD_ERROR_BUFLEN);
player->bufferRead += player->mp3Stream.next_frame - player->buffer;
BarPlayerBufferMove (player);
return WAITRESS_CB_RET_OK;
}
/**
* MP3音乐播放回调函数，
* 负责将解码数据填充声音缓存区
*
* @note 声音缓存区的格式为双声道，16位低字序
*
* @param buf 声音缓冲区指针
* @param reqn 缓冲区帧大小
* @param pdata 用户数据，无用
*/
static int mp3_audiocallback(void *buf, unsigned int reqn, void *pdata)
{
int avail_frame;
int snd_buf_frame_size = (int) reqn;
int ret;
double incr;
signed short *audio_buf = buf;
unsigned i;
uint16_t *output;
UNUSED(pdata);
if (g_status != ST_PLAYING) {
if (handle_seek() == -1) {
__end();
return -1;
}
xAudioClearSndBuf(buf, snd_buf_frame_size);
xrKernelDelayThread(100000);
return 0;
}
while (snd_buf_frame_size > 0) {
avail_frame = g_buff_frame_size - g_buff_frame_start;
if (avail_frame >= snd_buf_frame_size) {
send_to_sndbuf(audio_buf,
&g_buff[g_buff_frame_start * 2],
snd_buf_frame_size, 2);
g_buff_frame_start += snd_buf_frame_size;
audio_buf += snd_buf_frame_size * 2;
snd_buf_frame_size = 0;
} else {
send_to_sndbuf(audio_buf,
&g_buff[g_buff_frame_start * 2], avail_frame, 2);
snd_buf_frame_size -= avail_frame;
audio_buf += avail_frame * 2;
if (stream.buffer == NULL || stream.error == MAD_ERROR_BUFLEN) {
size_t read_size, remaining = 0;
uint8_t *read_start;
int bufsize;
if (stream.next_frame != NULL) {
remaining = stream.bufend - stream.next_frame;
memmove(g_input_buff, stream.next_frame, remaining);
read_start = g_input_buff + remaining;
read_size = BUFF_SIZE - remaining;
} else {
read_size = BUFF_SIZE;
read_start = g_input_buff;
remaining = 0;
}
if (mp3_data.use_buffer)
bufsize =
buffered_reader_read(mp3_data.r, read_start, read_size);
else
bufsize = xrIoRead(mp3_data.fd, read_start, read_size);
if (bufsize <= 0) {
__end();
return -1;
}
if (bufsize < read_size) {
uint8_t *guard = read_start + read_size;
memset(guard, 0, MAD_BUFFER_GUARD);
read_size += MAD_BUFFER_GUARD;
}
mad_stream_buffer(&stream, g_input_buff, read_size + remaining);
stream.error = 0;
}
ret = mad_frame_decode(&frame, &stream);
if (ret == -1) {
if (MAD_RECOVERABLE(stream.error)
|| stream.error == MAD_ERROR_BUFLEN) {
if (stream.error == MAD_ERROR_LOSTSYNC) {
long tagsize = id3_tag_query(stream.this_frame,
stream.bufend -
stream.this_frame);
if (tagsize > 0) {
mad_stream_skip(&stream, tagsize);
}
if (mad_header_decode(&frame.header, &stream) == -1) {
if (stream.error != MAD_ERROR_BUFLEN) {
if (!MAD_RECOVERABLE(stream.error)) {
__end();
return -1;
}
}
} else {
stream.error = MAD_ERROR_NONE;
}
}
g_buff_frame_size = 0;
g_buff_frame_start = 0;
continue;
} else {
__end();
return -1;
}
}
output = &g_buff[0];
if (stream.error != MAD_ERROR_NONE) {
continue;
}
mad_synth_frame(&synth, &frame);
for (i = 0; i < synth.pcm.length; i++) {
signed short sample;
if (MAD_NCHANNELS(&frame.header) == 2) {
/* Left channel */
sample = MadFixedToSshort(synth.pcm.samples[0][i]);
*(output++) = sample;
sample = MadFixedToSshort(synth.pcm.samples[1][i]);
*(output++) = sample;
} else {
sample = MadFixedToSshort(synth.pcm.samples[0][i]);
*(output++) = sample;
*(output++) = sample;
}
}
g_buff_frame_size = synth.pcm.length;
g_buff_frame_start = 0;
incr = frame.header.duration.seconds;
incr +=
mad_timer_fraction(frame.header.duration,
MAD_UNITS_MILLISECONDS) / 1000.0;
g_play_time += incr;
add_bitrate(&g_inst_br, frame.header.bitrate, incr);
}
}
return 0;
}
static gavl_source_status_t decode_frame_mad(bgav_stream_t * s)
{
mad_priv_t * priv;
int i, j;
gavl_source_status_t st;
int got_frame;
int flush = 0;
priv = s->decoder_priv;
if(priv->eof)
return GAVL_SOURCE_EOF;
while(1)
{
st = get_data(s);
switch(st)
{
case GAVL_SOURCE_AGAIN:
return st;
break;
case GAVL_SOURCE_EOF:
flush = 1;
break;
case GAVL_SOURCE_OK:
break;
}
got_frame = 1;
mad_stream_buffer(&priv->stream, priv->buf.buffer,
priv->buf.size + flush * MAD_BUFFER_GUARD);
if(priv->do_init)
{
get_format(s);
priv->do_init = 0;
}
if(mad_frame_decode(&priv->frame, &priv->stream) == -1)
{
got_frame = 0;
if(priv->stream.error != MAD_ERROR_BUFLEN)
{
/* This is often a spurious error, which ALWAYS occurs for ripped radio stations */
if(priv->stream.error != MAD_ERROR_BADDATAPTR)
bgav_log(s->opt, BGAV_LOG_ERROR, LOG_DOMAIN, "Decode failed %s\n",
mad_stream_errorstr(&priv->stream));
break;
}
}
else
break;
if(!got_frame && flush)
return GAVL_SOURCE_EOF;
}
if(got_frame)
{
// fprintf(stderr, "Decodes %ld bytes\n", priv->stream.next_frame - priv->stream.buffer);
mad_synth_frame(&priv->synth, &priv->frame);
for(i = 0; i < s->data.audio.format.num_channels; i++)
{
for(j = 0; j < s->data.audio.format.samples_per_frame; j++)
{
if (priv->synth.pcm.samples[i][j] >= MAD_F_ONE)
priv->synth.pcm.samples[i][j] = MAD_F_ONE - 1;
else if (priv->synth.pcm.samples[i][j] < -MAD_F_ONE)
priv->synth.pcm.samples[i][j] = -MAD_F_ONE;
priv->audio_frame->channels.f[i][j] =
(float)(priv->synth.pcm.samples[i][j]) /
(float)MAD_F_ONE;
}
}
priv->audio_frame->valid_samples = s->data.audio.format.samples_per_frame;
}
else
gavl_audio_frame_mute(priv->audio_frame, &s->data.audio.format);
if(flush && priv->last_duration &&
(priv->last_duration < priv->audio_frame->valid_samples))
priv->audio_frame->valid_samples = priv->last_duration;
gavl_audio_frame_copy_ptrs(&s->data.audio.format,
s->data.audio.frame, priv->audio_frame);
#if 0
fprintf(stderr, "Done decode %ld %ld\n",
priv->stream.this_frame - priv->stream.buffer,
priv->stream.next_frame - priv->stream.this_frame);
#endif
s->flags |= STREAM_HAVE_FRAME;
bgav_bytebuffer_remove(&priv->buf,
priv->stream.next_frame - priv->stream.buffer);
if(flush)
priv->eof = 1;
return GAVL_SOURCE_OK;
}
