/**
* Sends the synthesized current frame via decoder_data().
*/
static enum decoder_command
mp3_send_pcm(struct mp3_data *data, unsigned i, unsigned pcm_length)
{
unsigned max_samples;
max_samples = sizeof(data->output_buffer) /
sizeof(data->output_buffer[0]) /
MAD_NCHANNELS(&(data->frame).header);
while (i < pcm_length) {
enum decoder_command cmd;
unsigned int num_samples = pcm_length - i;
if (num_samples > max_samples)
num_samples = max_samples;
i += num_samples;
mad_fixed_to_24_buffer(data->output_buffer,
&data->synth,
i - num_samples, i,
MAD_NCHANNELS(&(data->frame).header));
num_samples *= MAD_NCHANNELS(&(data->frame).header);
cmd = decoder_data(data->decoder, data->input_stream,
data->output_buffer,
sizeof(data->output_buffer[0]) * num_samples,
data->bit_rate / 1000);
if (cmd != DECODE_COMMAND_NONE)
return cmd;
}
return DECODE_COMMAND_NONE;
}
void
mp3_mad_decode (mp3_info_t *info) {
// copy synthesized samples into readbuffer
int idx = info->mad_synth.pcm.length-info->buffer.decode_remaining;
// stereo
if (MAD_NCHANNELS(&info->mad_frame.header) == 2 && info->info.fmt.channels == 2) {
while (info->buffer.decode_remaining > 0 && info->buffer.readsize > 0) {
*((int16_t*)info->buffer.out) = MadFixedToSshort (info->mad_synth.pcm.samples[0][idx]);
info->buffer.readsize -= 2;
info->buffer.out += 2;
*((int16_t*)info->buffer.out) = MadFixedToSshort (info->mad_synth.pcm.samples[1][idx]);
info->buffer.readsize -= 2;
info->buffer.out += 2;
info->buffer.decode_remaining--;
idx++;
}
}
// mono
else if (MAD_NCHANNELS(&info->mad_frame.header) == 1 && info->info.fmt.channels == 1){
while (info->buffer.decode_remaining > 0 && info->buffer.readsize > 0) {
*((int16_t*)info->buffer.out) = MadFixedToSshort (info->mad_synth.pcm.samples[0][idx]);
info->buffer.readsize -= 2;
info->buffer.out += 2;
info->buffer.decode_remaining--;
idx++;
}
}
// workaround for bad mp3s that have both mono and stereo frames
else if (MAD_NCHANNELS(&info->mad_frame.header) == 1 && info->info.fmt.channels == 2) {
while (info->buffer.decode_remaining > 0 && info->buffer.readsize > 0) {
int16_t sample = MadFixedToSshort (info->mad_synth.pcm.samples[0][idx]);
*((int16_t*)info->buffer.out) = sample;
info->buffer.readsize -= 2;
info->buffer.out += 2;
*((int16_t*)info->buffer.out) = sample;
info->buffer.readsize -= 2;
info->buffer.out += 2;
info->buffer.decode_remaining--;
idx++;
}
}
else if (MAD_NCHANNELS(&info->mad_frame.header) == 2 && info->info.fmt.channels == 1) {
while (info->buffer.decode_remaining > 0 && info->buffer.readsize > 0) {
int16_t sample = MadFixedToSshort (info->mad_synth.pcm.samples[0][idx]);
*((int16_t*)info->buffer.out) = sample;
info->buffer.readsize -= 2;
info->buffer.out += 2;
info->buffer.decode_remaining--;
idx++;
}
}
}
Mp3PspStream::Mp3PspStream(Common::SeekableReadStream *inStream, DisposeAfterUse::Flag dispose) :
_inStream(inStream),
_disposeAfterUse(dispose),
_pcmLength(0),
_posInFrame(0),
_state(MP3_STATE_INIT),
_length(0, 1000),
_sampleRate(0),
_totalTime(mad_timer_zero) {
DEBUG_ENTER_FUNC();
assert(_decoderInit); // must be initialized by now
// let's leave the buffer guard -- who knows, it may be good?
memset(_buf, 0, sizeof(_buf));
memset(_codecInBuffer, 0, sizeof(_codecInBuffer));
initStream(); // init needed stuff for the stream
findValidHeader(); // get a first header so we can read basic stuff
_sampleRate = _header.samplerate; // copy it before it gets destroyed
_stereo = (MAD_NCHANNELS(&_header) == 2);
while (_state != MP3_STATE_EOS)
findValidHeader(); // get a first header so we can read basic stuff
_length = Timestamp(mad_timer_count(_totalTime, MAD_UNITS_MILLISECONDS), getRate());
deinitStream();
_state = MP3_STATE_INIT;
}
static int mp3_read (song_s *song, char *buffer)
{
static int i;
static int ret;
static struct audio_dither dither;
static char buffer2[BUF_SIZE];
static char *out_ptr = buffer2;
static char *out_buf_end = buffer2 + BUF_SIZE;
mp3_s *mp3 = song->songlib->mp3;
mad_timer_add (&mp3->timer, mp3->frame.header.duration);
mad_synth_frame (&mp3->synth, &mp3->frame);
mp3->elapsed_time = ((float)mad_timer_count (mp3->timer, MAD_UNITS_MILLISECONDS))/1000.0;
for (i = 0; i < mp3->synth.pcm.length; i++) {
signed int sample;
sample = (signed int)audio_linear_dither (16, mp3->synth.pcm.samples[0][i], &dither);
*(out_ptr++) = sample & 0xff;
*(out_ptr++) = sample >> 8;
if (MAD_NCHANNELS (&(mp3->frame).header) == 2) {
sample = (signed int) audio_linear_dither (16, mp3->synth.pcm.samples[1][i], &dither);
*(out_ptr++) = sample & 0xff;
*(out_ptr++) = sample >> 8;
}
if (out_ptr == out_buf_end) {
memcpy (buffer, buffer2, BUF_SIZE);
bossao_play_chunk (song, buffer, BUF_SIZE);
out_ptr = buffer2;
}
}
void Mp3Decoder::OpenFile()
{
GuardPtr = NULL;
ReadBuffer = (u8 *) memalign(32, SoundBlockSize*SoundBlocks);
if(!ReadBuffer)
{
if(file_fd)
delete file_fd;
file_fd = NULL;
return;
}
u8 dummybuff[4096];
int ret = Read(dummybuff, 4096, 0);
if(ret <= 0)
{
if(file_fd)
delete file_fd;
file_fd = NULL;
return;
}
SampleRate = (u32) Frame.header.samplerate;
Format = ((MAD_NCHANNELS(&Frame.header) == 2) ? VOICE_STEREO_16BIT : VOICE_MONO_16BIT);
Rewind();
Decode();
}
int output_synth_data(mpgedit_madbuf_t *ctx)
{
int i;
int rsts;
for(i=0;i<ctx->Synth.pcm.length;i++) {
signed short Sample;
/* Left channel */
Sample=MadFixedToSshort(ctx->Synth.pcm.samples[0][i]);
*(ctx->OutputPtr++) = Sample & 0xff;
*(ctx->OutputPtr++) = Sample >> 8;
/* Right channel. If the decoded stream is monophonic then
* the right output channel is the same as the left one.
*/
if(MAD_NCHANNELS(&ctx->Frame.header)==2) {
Sample=MadFixedToSshort(ctx->Synth.pcm.samples[1][i]);
*(ctx->OutputPtr++) = Sample & 0xff;
*(ctx->OutputPtr++) = Sample >> 8;
}
/* Flush the output buffer if it is full. */
if (ctx->OutputPtr >= ctx->OutputBufferEnd) {
rsts = fwrite(ctx->OutputBuffer, 1, OUTPUT_BUFFER_SIZE, ctx->outfp);
if (rsts != OUTPUT_BUFFER_SIZE) {
fprintf(stderr,"%s: PCM write error (%s).\n",
"main" ,strerror(errno));
return 1;
}
ctx->OutputPtr = ctx->OutputBuffer;
}
}
/*
* NAME: synth->frame()
* DESCRIPTION: perform PCM synthesis of frame subband samples
*/
void mad_synth_frame(struct mad_synth *synth, struct mad_frame const *frame) {
unsigned int nch, ns;
void (*synth_frame)(struct mad_synth *, struct mad_frame const *,
unsigned int, unsigned int);
nch = MAD_NCHANNELS(&frame->header);
ns = MAD_NSBSAMPLES(&frame->header);
synth->pcm.samplerate = frame->header.samplerate;
synth->pcm.channels = nch;
synth->pcm.length = 32 * ns;
synth_frame = synth_full;
if (frame->options & MAD_OPTION_HALFSAMPLERATE) {
synth->pcm.samplerate /= 2;
synth->pcm.length /= 2;
synth_frame = synth_half;
}
synth_frame(synth, frame, nch, ns);
synth->phase = (synth->phase + ns) % 16;
}
static enum mad_flow mad_header_cb(void *blob, const mad_header_t *header) {
state_t *data = (state_t *)blob;
data->sample_rate = header->samplerate;
data->output_channels = MAD_NCHANNELS(header);
data->output_size += mad_timer_count(header->duration, header->samplerate);
return MAD_FLOW_IGNORE;
}
int MP3Stream::readBuffer(int16 *buffer, const int numSamples) {
int samples = 0;
// Keep going as long as we have input available
while (samples < numSamples && _state != MP3_STATE_EOS) {
const int len = MIN(numSamples, samples + (int)(_synth.pcm.length - _posInFrame) * MAD_NCHANNELS(&_frame.header));
while (samples < len) {
*buffer++ = (int16)scale_sample(_synth.pcm.samples[0][_posInFrame]);
samples++;
if (MAD_NCHANNELS(&_frame.header) == 2) {
*buffer++ = (int16)scale_sample(_synth.pcm.samples[1][_posInFrame]);
samples++;
}
_posInFrame++;
}
if (_posInFrame >= _synth.pcm.length) {
// We used up all PCM data in the current frame -- read & decode more
decodeMP3Data();
}
}
return samples;
}
static int put_output (char *buf, int buf_len, struct mad_pcm *pcm,
struct mad_header *header)
{
unsigned int nsamples;
mad_fixed_t const *left_ch, *right_ch;
int olen;
nsamples = pcm->length;
left_ch = pcm->samples[0];
right_ch = pcm->samples[1];
olen = nsamples * MAD_NCHANNELS (header) * 4;
if (olen > buf_len) {
logit ("PCM buffer to small!");
return 0;
}
while (nsamples--) {
long sample0 = round_sample (*left_ch++);
buf[0] = 0;
buf[1] = sample0;
buf[2] = sample0 >> 8;
buf[3] = sample0 >> 16;
buf += 4;
if (MAD_NCHANNELS(header) == 2) {
long sample1;
sample1 = round_sample (*right_ch++);
buf[0] = 0;
buf[1] = sample1;
buf[2] = sample1 >> 8;
buf[3] = sample1 >> 16;
buf += 4;
}
}
int Mp3PspStream::readBuffer(int16 *buffer, const int numSamples) {
DEBUG_ENTER_FUNC();
int samples = 0;
#ifdef PRINT_BUFFERS
int16 *debugBuffer = buffer;
#endif
// Keep going as long as we have input available
while (samples < numSamples && _state != MP3_STATE_EOS) {
const int len = MIN(numSamples, samples + (int)(_pcmLength - _posInFrame) * MAD_NCHANNELS(&_header));
while (samples < len) {
*buffer++ = _pcmSamples[_posInFrame << 1];
samples++;
if (MAD_NCHANNELS(&_header) == 2) {
*buffer++ = _pcmSamples[(_posInFrame << 1) + 1];
samples++;
}
_posInFrame++; // always skip an extra sample since ME always outputs stereo
}
if (_posInFrame >= _pcmLength) {
// We used up all PCM data in the current frame -- read & decode more
decodeMP3Data();
}
}
#ifdef PRINT_BUFFERS
PSP_INFO_PRINT("buffer:\n");
for (int i = 0; i<numSamples; i++)
PSP_INFO_PRINT("%d ", debugBuffer[i]);
PSP_INFO_PRINT("\n\n");
#endif
return samples;
}
RageSoundReader_FileReader::OpenResult RageSoundReader_MP3::Open( RageFileBasic *pFile )
{
m_pFile = pFile;
mad->filesize = m_pFile->GetFileSize();
ASSERT( mad->filesize != -1 );
/* Make sure we can decode at least one frame. This will also fill in header info. */
mad->outpos = 0;
int ret = do_mad_frame_decode();
switch(ret)
{
case 0:
SetError( "Failed to read any data at all" );
return OPEN_UNKNOWN_FILE_FORMAT;
case -1:
SetError( GetError() + " (not an MP3 stream?)" );
return OPEN_UNKNOWN_FILE_FORMAT;
}
/* Store the bitrate of the frame we just got. */
if( mad->bitrate == -1 ) /* might have been set by a Xing tag */
mad->bitrate = mad->Frame.header.bitrate;
SampleRate = mad->Frame.header.samplerate;
mad->framelength = mad->Frame.header.duration;
this->Channels = MAD_NCHANNELS( &mad->Frame.header );
/* Since we've already decoded a frame, just synth it instead of rewinding
* the stream. */
synth_output();
if(mad->length == -1)
{
/* If vbr and !xing, this is just an estimate. */
int bps = mad->bitrate / 8;
float secs = (float)(mad->filesize - mad->header_bytes) / bps;
mad->length = (int)(secs * 1000.f);
}
return OPEN_OK;
}
//extern unsigned char set__;
void mad_synth_frame(struct mad_synth *synth, struct mad_frame /*const*/ *frame)
{
unsigned int nch, ns;
static int samplerate=0;
void (*synth_frame)(struct mad_synth *, struct mad_frame /*const*/ *,
unsigned int, unsigned int);
nch = MAD_NCHANNELS(&frame->header);
ns = MAD_NSBSAMPLES(&frame->header);
synth->pcm.samplerate = frame->header.samplerate;
//--set_dac_sample_rate(synth->pcm.samplerate);
//while(g_ulFlags!=7);
//if(set__)
{
vPortEnterCritical( );
SoundSetFormat(frame->header.samplerate,16,1);
//set__=0;
vPortExitCritical( );
}
synth->pcm.channels = nch;
synth->pcm.length = 32 * ns;
samplerate = synth->pcm.samplerate;
synth_frame = synth_full;
if (frame->options & MAD_OPTION_HALFSAMPLERATE) {
synth->pcm.samplerate /= 2;
synth->pcm.length /= 2;
//--set_dac_sample_rate(synth->pcm.samplerate);
vPortEnterCritical( );
SoundSetFormat(frame->header.samplerate,16,1);
vPortExitCritical( );
synth_frame = synth_half;
}
synth_frame(synth, frame, nch, ns);
synth->phase = (synth->phase + ns) % 16;
}
bool MADDecoder::Open(FileSpecifier &File)
{
if (!File.Open(file)) return false;
file_done = false;
if (DecodeFrame())
{
stereo = (MAD_NCHANNELS(&Frame.header) == 2);
bytes_per_frame = 2 * (stereo ? 2 : 1);
rate = Frame.header.samplerate;
sample = 0;
return true;
}
else
{
return false;
}
}
static void
mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
{
struct mp3_data data;
GError *error = NULL;
struct tag *tag = NULL;
struct audio_format audio_format;
if (!mp3_open(input_stream, &data, decoder, &tag)) {
if (decoder_get_command(decoder) == DECODE_COMMAND_NONE)
g_warning
("Input does not appear to be a mp3 bit stream.\n");
return;
}
if (!audio_format_init_checked(&audio_format,
data.frame.header.samplerate,
SAMPLE_FORMAT_S24_P32,
MAD_NCHANNELS(&data.frame.header),
&error)) {
g_warning("%s", error->message);
g_error_free(error);
if (tag != NULL)
tag_free(tag);
mp3_data_finish(&data);
return;
}
decoder_initialized(decoder, &audio_format,
data.input_stream->seekable, data.total_time);
if (tag != NULL) {
decoder_tag(decoder, input_stream, tag);
tag_free(tag);
}
while (mp3_read(&data)) ;
mp3_data_finish(&data);
}
void RageSoundReader_MP3::synth_output()
{
mad->outleft = mad->outpos = 0;
if( MAD_NCHANNELS(&mad->Frame.header) != this->Channels )
{
/* This frame contains a different number of channels than the first.
* I've never actually seen this--it's just to prevent exploding if
* it happens--and we could continue on easily enough, so if it happens,
* just continue. */
return;
}
mad_synth_frame(&mad->Synth, &mad->Frame);
for(int i=0; i < mad->Synth.pcm.length; i++)
{
for(int chan = 0; chan < this->Channels; ++chan)
{
float Sample = scale(mad->Synth.pcm.samples[chan][i]);
mad->outbuf[mad->outleft] = Sample;
++mad->outleft;
}
}
}
void LibMadWrapper::open()
{
// avoid multiple calls to open()
if (this->infile != nullptr)
{
return;
}
this->infile = fopen(this->Filename.c_str(), "rb");
if (this->infile == nullptr)
{
THROW_RUNTIME_ERROR("fopen failed, errno: " << string(strerror(errno)));
}
int fd = fileno(this->infile);
this->mpeglen = getFileSize(fd);
this->mpegbuf = static_cast<unsigned char *>(mmap(nullptr, this->mpeglen, PROT_READ, MAP_SHARED, fd, 0));
if (this->mpegbuf == MAP_FAILED)
{
THROW_RUNTIME_ERROR("mmap failed for File \"" << this->Filename << ")\"");
}
this->stream = new struct mad_stream;
mad_stream_init(this->stream);
/* load buffer with MPEG audio data */
mad_stream_buffer(this->stream, this->mpegbuf, this->mpeglen);
// we want to know how many pcm frames there are decoded in this file
// therefore decode header of every mpeg frame
// pretty expensive, so only to this once
if (this->numFrames == 0)
{
struct mad_header header;
mad_header_init(&header);
// try to find a valid header
int ret = this->findValidHeader(header);
if (ret != 0)
{
// only free the locally used header here, this->stream and this->mpegbuf are freed in LibMadWrapper::close()
mad_header_finish(&header);
THROW_RUNTIME_ERROR("unable to find a valid frame-header for File \"" + this->Filename + "\"");
}
this->Format.SetVoices(1);
// a first valid header is good, but it may contain garbage
this->Format.VoiceChannels[0] = MAD_NCHANNELS(&header);
this->Format.SampleRate = header.samplerate;
CLOG(LogLevel_t::Debug, "found a first valid header within File \"" << this->Filename << "\"\n\tchannels: " << MAD_NCHANNELS(&header) << "\nsrate: " << header.samplerate);
// no clue what this 32 does
// stolen from mad_synth_frame() in synth.c
this->numFrames += 32 * MAD_NSBSAMPLES(&header);
// try to find a second valid header
ret = this->findValidHeader(header);
if (ret == 0)
{
// better use format infos from this header
this->Format.VoiceChannels[0] = max<int>(MAD_NCHANNELS(&header), this->Format.VoiceChannels[0]);
this->Format.SampleRate = header.samplerate;
CLOG(LogLevel_t::Debug, "found a second valid header within File \"" << this->Filename << "\"\n\tchannels: " << MAD_NCHANNELS(&header) << "\nsrate: " << header.samplerate);
this->numFrames += 32 * MAD_NSBSAMPLES(&header);
// now lets go on and decode rest of file
while (1)
{
if (mad_header_decode(&header, this->stream) != 0)
{
if (MAD_RECOVERABLE(this->stream->error))
{
continue;
}
else
{
break;
}
}
// sanity checks
if (this->Format.Channels() != MAD_NCHANNELS(&header))
{
CLOG(LogLevel_t::Warning, "channelcount varies (now: " << MAD_NCHANNELS(&header) << ") within File \"" << this->Filename << ")\"");
if (!gConfig.MadPermissive)
{
THROW_RUNTIME_ERROR("invalid mp3: channelcount varies");
}
}
if (this->Format.SampleRate != header.samplerate)
{
CLOG(LogLevel_t::Warning, "samplerate varies (now: " << header.samplerate << ") within File \"" << this->Filename << ")\"");
if (!gConfig.MadPermissive)
{
THROW_RUNTIME_ERROR("invalid mp3: samplerate varies");
}
}
this->numFrames += 32 * MAD_NSBSAMPLES(&header);
}
}
else
{
CLOG(LogLevel_t::Warning, "only one valid header found, probably no valid mp3 File \"" << this->Filename << "\"");
}
// somehow reset libmad stream
mad_stream_finish(this->stream);
mad_stream_init(this->stream);
/* load buffer with MPEG audio data */
mad_stream_buffer(this->stream, this->mpegbuf, this->mpeglen);
mad_header_finish(&header);
}
this->frame.hasValue = true;
mad_frame_init(&this->frame.Value);
this->synth.hasValue = true;
mad_synth_init(&this->synth.Value);
}
enum mad_flow output(void *data,
struct mad_header const *header,
struct mad_pcm *pcm)
{
register int nsamples = pcm->length;
mad_fixed_t const *left_ch = pcm->samples[0], *right_ch = pcm->samples[1];
static unsigned char stream[1152*4]; /* 1152 because that's what mad has as a max; *4 because
there are 4 distinct bytes per sample (in 2 channel case) */
static unsigned int rate = 0;
static int channels = 0;
static struct audio_dither dither;
register char * ptr = stream;
register signed int sample;
register mad_fixed_t tempsample;
/* Semaphore operations */
static struct sembuf read_sops = {0, -1, 0};
static struct sembuf write_sops = {1, 1, 0};
if(options.opt & MPG321_ENABLE_BUFFER)
{
semop(semarray,&read_sops,1);
ptr = (Output_Queue+mad_decoder_position)->data;
memcpy(&((Output_Queue+mad_decoder_position)->header),header,sizeof(struct mad_header));
}else{
/* We need to know information about the file before we can open the playdevice
in some cases. So, we do it here. */
if (!playdevice)
{
channels = MAD_NCHANNELS(header);
rate = header->samplerate;
open_ao_playdevice(header);
}
else if ((channels != MAD_NCHANNELS(header) || rate != header->samplerate) && playdevice_is_live())
{
ao_close(playdevice);
channels = MAD_NCHANNELS(header);
rate = header->samplerate;
open_ao_playdevice(header);
}
}
static int peak_rate = 0;
static unsigned short int peak = 0;
if (pcm->channels == 2)
{
while (nsamples--)
{
tempsample = mad_f_mul(*left_ch++, options.volume);
sample = (signed int) audio_linear_dither(16, tempsample, &dither);
peak = abs(sample) > peak ? abs(sample) : peak;
#ifndef WORDS_BIGENDIAN
*ptr++ = (unsigned char) (sample >> 0);
*ptr++ = (unsigned char) (sample >> 8);
#else
*ptr++ = (unsigned char) (sample >> 8);
*ptr++ = (unsigned char) (sample >> 0);
#endif
tempsample = mad_f_mul(*right_ch++, options.volume);
sample = (signed int) audio_linear_dither(16, tempsample, &dither);
peak = abs(sample) > peak ? abs(sample) : peak;
#ifndef WORDS_BIGENDIAN
*ptr++ = (unsigned char) (sample >> 0);
*ptr++ = (unsigned char) (sample >> 8);
#else
*ptr++ = (unsigned char) (sample >> 8);
*ptr++ = (unsigned char) (sample >> 0);
#endif
}
process_fft(stream, pcm->length * 4);
if(options.opt & MPG321_ENABLE_BUFFER)
{
(Output_Queue+mad_decoder_position)->length = pcm->length * 4;
}else
{
ao_play(playdevice, stream, pcm->length * 4);
}
}
else if (options.opt & MPG321_FORCE_STEREO)
enum mad_flow read_header(void *data, struct mad_header const * header)
{
char long_currenttime_str[14]; /* this *will* fill if you're using 100000+ minute mp3s */
char long_remaintime_str[14];
static int frames_played = 0;
buffer *playbuf = (buffer *)data;
mad_timer_t time_remaining;
char *ao_time;
if (stop_playing_file)
{
stop_playing_file = 0;
status = MPG321_STOPPED;
return MAD_FLOW_STOP;
}
if(options.opt & MPG321_REMOTE_PLAY)
{
enum mad_flow mf;
/* We might have to stop if the user inputs something */
if ((mf = remote_get_input_nowait(playbuf)))
return mf;
}
/* Stop playing if -n is used, and we're at the frame specified. */
if ((playbuf->max_frames != -1) && (frames_played++ > playbuf->max_frames))
{
frames_played = 0;
status = MPG321_STOPPED;
if(options.opt & MPG321_ENABLE_BUFFER) Decoded_Frames->done = 1;
//fprintf(stderr,"Total D: %d\n",Decoded_Frames->total_decoded_frames);
return MAD_FLOW_STOP;
}
current_frame++;
mad_timer_add(¤t_time, header->duration);
if(options.opt & MPG321_USE_SCROBBLER && scrobbler_time > 0 && scrobbler_time < current_time.seconds)
{
scrobbler_time = -1;
scrobbler_report();
}
if(options.opt & (MPG321_VERBOSE_PLAY | MPG321_REMOTE_PLAY))
{
mad_timer_string(current_time, long_currenttime_str, "%.2u:%.2u.%.2u", MAD_UNITS_MINUTES,
MAD_UNITS_CENTISECONDS, 0);
if (mad_timer_compare(playbuf->duration, mad_timer_zero) == 0)
time_remaining = current_time;
else
time_remaining = playbuf->duration;
mad_timer_negate(¤t_time);
mad_timer_add(&time_remaining, current_time);
mad_timer_negate(¤t_time);
mad_timer_string(time_remaining, long_remaintime_str, "%.2u:%.2u.%.2u", MAD_UNITS_MINUTES,
MAD_UNITS_CENTISECONDS, 0);
}
/* update cached table of frames & times */
if (current_frame <= playbuf->num_frames) /* we only allocate enough for our estimate. */
{
playbuf->frames[current_frame] = playbuf->frames[current_frame-1] + (header->bitrate / 8 / 1000)
* mad_timer_count(header->duration, MAD_UNITS_MILLISECONDS);
playbuf->times[current_frame] = current_time;
}
if (file_change)
{
file_change = 0;
if (options.opt & MPG321_REMOTE_PLAY)
{
printf("@S %s %d %d %s %d %ld %d %d %d %d %ld %d\n",versionstring(header->flags), header->layer, header->samplerate,
modestringucase(header->mode), header->mode_extension,
(header->bitrate / 8 / 100) * mad_timer_count(header->duration, MAD_UNITS_CENTISECONDS),
MAD_NCHANNELS(header), header->flags & MAD_FLAG_COPYRIGHT ? 1 : 0,
header->flags & MAD_FLAG_PROTECTION ? 1 : 0, header->emphasis,
header->bitrate/1000, header->mode_extension);
}
else if (options.opt & MPG321_VERBOSE_PLAY)/*zip it good*/
{
fprintf(stderr, "MPEG %s, Layer: %s, Freq: %d, mode: %s, modext: %d, BPF : %ld\n"
"Channels: %d, copyright: %s, original: %s, CRC: %s, emphasis: %d.\n"
"Bitrate: %ld Kbits/s, Extension value: %d\n"
"Audio: 1:1 conversion, rate: %d, encoding: signed 16 bit, channels: %d\n",
versionstring(header->flags),layerstring(header->layer), header->samplerate, modestringucase(header->mode), header->mode_extension,
(header->bitrate / 100) * mad_timer_count(header->duration, MAD_UNITS_CENTISECONDS),
MAD_NCHANNELS(header), header->flags & MAD_FLAG_COPYRIGHT ? "Yes" : "No",
header->flags & MAD_FLAG_ORIGINAL ? "Yes" : "No", header->flags & MAD_FLAG_PROTECTION ? "Yes" : "No",
header->emphasis, header->bitrate/1000, header->mode_extension,
header->samplerate, MAD_NCHANNELS(header));
}
else if (!(options.opt & MPG321_QUIET_PLAY))/*I love Joey*/
{
fprintf(stderr, "MPEG %s layer %s, %ld kbit/s, %d Hz %s\n",
versionstring(header->flags),layerstring(header->layer), header->bitrate/1000, header->samplerate, modestring(header->mode));
}
}
if (status == MPG321_SEEKING && options.seek)
{
if (!--options.seek)
status = MPG321_PLAYING;
return MAD_FLOW_IGNORE;
}
else
{
status = MPG321_PLAYING;
}
if(!(options.opt & MPG321_ENABLE_BUFFER))
{
if(count > 0)
{
count++;
if(count > 40)
{
if(!(options.opt & MPG321_VERBOSE_PLAY))
fprintf(stderr," \r");
count = -1;
fflush(stderr);
}
}
}
if(options.opt & MPG321_ENABLE_BUFFER)
{
ao_time = (Output_Queue+mad_decoder_position)->time;
(Output_Queue+mad_decoder_position)->num_frames = playbuf->num_frames - current_frame;
if(Decoded_Frames->is_file) (Output_Queue+mad_decoder_position)->seconds = time_remaining.seconds;
}
if (options.opt & MPG321_VERBOSE_PLAY)
{
if (!options.skip_printing_frames
|| (options.skip_printing_frames && !(current_frame % options.skip_printing_frames)))
if(count > 0)
{
/* if(options.opt & MPG321_ENABLE_BUFFER)
{
sprintf(ao_time, "Frame# %5lu [%5lu], Time: %s [%s], \r", current_frame,
playbuf->num_frames > 0 ? playbuf->num_frames - current_frame : 0, long_currenttime_str, long_remaintime_str);
//sprintf(ao_time, "Frame# %5lu [%5lu], Time: %s [%s], \r", current_frame,
// playbuf->num_frames > 0 ? playbuf->num_frames - current_frame : 0, long_currenttime_str, long_remaintime_str);
//sprintf(ao_time, "Volume: %d%% Frame# %5lu [%5lu], Time: %s [%s], \r",volume, current_frame,
// playbuf->num_frames > 0 ? playbuf->num_frames - current_frame : 0, long_currenttime_str, long_remaintime_str);
}else*/
{
fprintf(stderr, "Volume: %d%% Frame# %5lu [%5lu], Time: %s [%s], \r",volume, current_frame,
playbuf->num_frames > 0 ? playbuf->num_frames - current_frame : 0, long_currenttime_str, long_remaintime_str);
}
}
else if(count < 0)
{
if(options.opt & MPG321_ENABLE_BUFFER)
{
// sprintf(ao_time, "Frame# %5lu [%5lu], Time: %s [%s], \r", current_frame,
sprintf(ao_time, "Frame# %5lu [%5lu], Time: %s [%s], \r", current_frame,
playbuf->num_frames > 0 ? playbuf->num_frames - current_frame : 0, long_currenttime_str, long_remaintime_str);
}
else
{
fprintf(stderr, "Frame# %5lu [%5lu], Time: %s [%s], \r", current_frame,
playbuf->num_frames > 0 ? playbuf->num_frames - current_frame : 0, long_currenttime_str, long_remaintime_str);
}
}
}
else if (options.opt & MPG321_REMOTE_PLAY)
{
if (!options.skip_printing_frames
|| (options.skip_printing_frames && !(current_frame % options.skip_printing_frames)))
{
if(options.opt & MPG321_ENABLE_BUFFER)
{
sprintf(ao_time,"@F %ld %ld %.2f %.2f\n", current_frame, playbuf->num_frames - current_frame,
((double)mad_timer_count(current_time, MAD_UNITS_CENTISECONDS)/100.0),
((double)mad_timer_count(time_remaining, MAD_UNITS_CENTISECONDS)/100.0));
}
else
{
fprintf(stderr,"@F %ld %ld %.2f %.2f\n", current_frame, playbuf->num_frames - current_frame,
((double)mad_timer_count(current_time, MAD_UNITS_CENTISECONDS)/100.0),
((double)mad_timer_count(time_remaining, MAD_UNITS_CENTISECONDS)/100.0));
}
}
}
return MAD_FLOW_CONTINUE;
}
/*
* NAME: layer->II()
* DESCRIPTION: decode a single Layer II frame
*/
int mad_layer_II(struct mad_stream *stream, struct mad_frame *frame)
{
struct mad_header *header = &frame->header;
struct mad_bitptr start;
unsigned int index, sblimit, nbal, nch, bound, gr, ch, s, sb;
unsigned char const *offsets;
unsigned char allocation[2][32], scfsi[2][32], scalefactor[2][32][3];
mad_fixed_t samples[3];
nch = MAD_NCHANNELS(header);
if (header->flags & MAD_FLAG_LSF_EXT)
index = 4;
else {
switch (nch == 2 ? header->bitrate / 2 : header->bitrate) {
case 32000:
case 48000:
index = (header->samplerate == 32000) ? 3 : 2;
break;
case 56000:
case 64000:
case 80000:
index = 0;
break;
default:
index = (header->samplerate == 48000) ? 0 : 1;
}
}
sblimit = sbquant_table[index].sblimit;
offsets = sbquant_table[index].offsets;
bound = 32;
if (header->mode == MAD_MODE_JOINT_STEREO) {
header->flags |= MAD_FLAG_I_STEREO;
bound = 4 + header->mode_extension * 4;
}
if (bound > sblimit)
bound = sblimit;
start = stream->ptr;
/* decode bit allocations */
for (sb = 0; sb < bound; ++sb) {
nbal = bitalloc_table[offsets[sb]].nbal;
for (ch = 0; ch < nch; ++ch)
allocation[ch][sb] = mad_bit_read(&stream->ptr, nbal);
}
for (sb = bound; sb < sblimit; ++sb) {
nbal = bitalloc_table[offsets[sb]].nbal;
allocation[0][sb] =
allocation[1][sb] = mad_bit_read(&stream->ptr, nbal);
}
/* decode scalefactor selection info */
for (sb = 0; sb < sblimit; ++sb) {
for (ch = 0; ch < nch; ++ch) {
if (allocation[ch][sb])
scfsi[ch][sb] = mad_bit_read(&stream->ptr, 2);
}
}
/* check CRC word */
if (header->flags & MAD_FLAG_PROTECTION) {
header->crc_check =
mad_bit_crc(start, mad_bit_length(&start, &stream->ptr),
header->crc_check);
if (header->crc_check != header->crc_target &&
!(frame->options & MAD_OPTION_IGNORECRC)) {
stream->error = MAD_ERROR_BADCRC;
return -1;
}
}
/* decode scalefactors */
for (sb = 0; sb < sblimit; ++sb) {
for (ch = 0; ch < nch; ++ch) {
if (allocation[ch][sb]) {
scalefactor[ch][sb][0] = mad_bit_read(&stream->ptr, 6);
switch (scfsi[ch][sb]) {
case 2:
scalefactor[ch][sb][2] =
scalefactor[ch][sb][1] =
scalefactor[ch][sb][0];
break;
case 0:
scalefactor[ch][sb][1] = mad_bit_read(&stream->ptr, 6);
/* fall through */
case 1:
case 3:
scalefactor[ch][sb][2] = mad_bit_read(&stream->ptr, 6);
}
if (scfsi[ch][sb] & 1)
scalefactor[ch][sb][1] = scalefactor[ch][sb][scfsi[ch][sb] - 1];
# if defined(OPT_STRICT)
/*
* Scalefactor index 63 does not appear in Table B.1 of
* ISO/IEC 11172-3. Nonetheless, other implementations accept it,
* so we only reject it if OPT_STRICT is defined.
*/
if (scalefactor[ch][sb][0] == 63 ||
scalefactor[ch][sb][1] == 63 ||
scalefactor[ch][sb][2] == 63) {
stream->error = MAD_ERROR_BADSCALEFACTOR;
return -1;
}
# endif
}
}
}
/* decode samples */
for (gr = 0; gr < 12; ++gr) {
for (sb = 0; sb < bound; ++sb) {
for (ch = 0; ch < nch; ++ch) {
if ((index = allocation[ch][sb])) {
index = offset_table[bitalloc_table[offsets[sb]].offset][index - 1];
II_samples(&stream->ptr, &qc_table[index], samples);
for (s = 0; s < 3; ++s) {
frame->sbsample[ch][3 * gr + s][sb] =
mad_f_mul(samples[s], sf_table[scalefactor[ch][sb][gr / 4]]);
}
}
else {
for (s = 0; s < 3; ++s)
frame->sbsample[ch][3 * gr + s][sb] = 0;
}
}
}
for (sb = bound; sb < sblimit; ++sb) {
if ((index = allocation[0][sb])) {
index = offset_table[bitalloc_table[offsets[sb]].offset][index - 1];
II_samples(&stream->ptr, &qc_table[index], samples);
for (ch = 0; ch < nch; ++ch) {
for (s = 0; s < 3; ++s) {
frame->sbsample[ch][3 * gr + s][sb] =
mad_f_mul(samples[s], sf_table[scalefactor[ch][sb][gr / 4]]);
}
}
}
else {
for (ch = 0; ch < nch; ++ch) {
for (s = 0; s < 3; ++s)
frame->sbsample[ch][3 * gr + s][sb] = 0;
}
}
}
for (ch = 0; ch < nch; ++ch) {
for (s = 0; s < 3; ++s) {
for (sb = sblimit; sb < 32; ++sb)
frame->sbsample[ch][3 * gr + s][sb] = 0;
}
}
}
return 0;
}
/*
* NAME: layer->I()
* DESCRIPTION: decode a single Layer I frame
*/
int mad_layer_I(struct mad_stream *stream, struct mad_frame *frame)
{
struct mad_header *header = &frame->header;
unsigned int nch, bound, ch, s, sb, nb;
unsigned char allocation[2][32], scalefactor[2][32];
nch = MAD_NCHANNELS(header);
bound = 32;
if (header->mode == MAD_MODE_JOINT_STEREO) {
header->flags |= MAD_FLAG_I_STEREO;
bound = 4 + header->mode_extension * 4;
}
/* check CRC word */
if (header->flags & MAD_FLAG_PROTECTION) {
header->crc_check =
mad_bit_crc(stream->ptr, 4 * (bound * nch + (32 - bound)),
header->crc_check);
if (header->crc_check != header->crc_target &&
!(frame->options & MAD_OPTION_IGNORECRC)) {
stream->error = MAD_ERROR_BADCRC;
return -1;
}
}
/* decode bit allocations */
for (sb = 0; sb < bound; ++sb) {
for (ch = 0; ch < nch; ++ch) {
nb = mad_bit_read(&stream->ptr, 4);
if (nb == 15) {
stream->error = MAD_ERROR_BADBITALLOC;
return -1;
}
allocation[ch][sb] = nb ? nb + 1 : 0;
}
}
for (sb = bound; sb < 32; ++sb) {
nb = mad_bit_read(&stream->ptr, 4);
if (nb == 15) {
stream->error = MAD_ERROR_BADBITALLOC;
return -1;
}
allocation[0][sb] =
allocation[1][sb] = nb ? nb + 1 : 0;
}
/* decode scalefactors */
for (sb = 0; sb < 32; ++sb) {
for (ch = 0; ch < nch; ++ch) {
if (allocation[ch][sb]) {
scalefactor[ch][sb] = mad_bit_read(&stream->ptr, 6);
# if defined(OPT_STRICT)
/*
* Scalefactor index 63 does not appear in Table B.1 of
* ISO/IEC 11172-3. Nonetheless, other implementations accept it,
* so we only reject it if OPT_STRICT is defined.
*/
if (scalefactor[ch][sb] == 63) {
stream->error = MAD_ERROR_BADSCALEFACTOR;
return -1;
}
# endif
}
}
}
/* decode samples */
for (s = 0; s < 12; ++s) {
for (sb = 0; sb < bound; ++sb) {
for (ch = 0; ch < nch; ++ch) {
nb = allocation[ch][sb];
frame->sbsample[ch][s][sb] = nb ?
mad_f_mul(I_sample(&stream->ptr, nb),
sf_table[scalefactor[ch][sb]]) : 0;
}
}
for (sb = bound; sb < 32; ++sb) {
if ((nb = allocation[0][sb])) {
mad_fixed_t sample;
sample = I_sample(&stream->ptr, nb);
for (ch = 0; ch < nch; ++ch) {
frame->sbsample[ch][s][sb] =
mad_f_mul(sample, sf_table[scalefactor[ch][sb]]);
}
}
else {
for (ch = 0; ch < nch; ++ch)
frame->sbsample[ch][s][sb] = 0;
}
}
}
return 0;
}
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;
}
SoundSource::OpenResult SoundSourceMp3::tryOpen(const AudioSourceConfig& /*audioSrcCfg*/) {
DEBUG_ASSERT(!hasValidChannelCount());
DEBUG_ASSERT(!hasValidSamplingRate());
DEBUG_ASSERT(!m_file.isOpen());
if (!m_file.open(QIODevice::ReadOnly)) {
qWarning() << "Failed to open file:" << m_file.fileName();
return OpenResult::FAILED;
}
// Get a pointer to the file using memory mapped IO
m_fileSize = m_file.size();
m_pFileData = m_file.map(0, m_fileSize);
// NOTE(uklotzde): If the file disappears unexpectedly while mapped
// a SIGBUS error might occur that is not handled and will terminate
// Mixxx immediately. This behavior is documented in the manpage of
// mmap(). It has already appeared due to hardware errors and is
// described in the following bug report:
// https://bugs.launchpad.net/mixxx/+bug/1452005
// Transfer it to the mad stream-buffer:
mad_stream_options(&m_madStream, MAD_OPTION_IGNORECRC);
mad_stream_buffer(&m_madStream, m_pFileData, m_fileSize);
DEBUG_ASSERT(m_pFileData == m_madStream.this_frame);
DEBUG_ASSERT(m_seekFrameList.empty());
m_avgSeekFrameCount = 0;
m_curFrameIndex = getMinFrameIndex();
int headerPerSamplingRate[kSamplingRateCount];
for (int i = 0; i < kSamplingRateCount; ++i) {
headerPerSamplingRate[i] = 0;
}
// Decode all the headers and calculate audio properties
unsigned long sumBitrate = 0;
mad_header madHeader;
mad_header_init(&madHeader);
SINT maxChannelCount = getChannelCount();
do {
if (!decodeFrameHeader(&madHeader, &m_madStream, true)) {
if (isStreamValid(m_madStream)) {
// Skip frame
continue;
} else {
// Abort decoding
break;
}
}
// Grab data from madHeader
const unsigned int madSampleRate = madHeader.samplerate;
// TODO(XXX): Replace DEBUG_ASSERT with static_assert
// MAD must not change its enum values!
DEBUG_ASSERT(MAD_UNITS_8000_HZ == 8000);
const mad_units madUnits = static_cast<mad_units>(madSampleRate);
const long madFrameLength = mad_timer_count(madHeader.duration, madUnits);
if (0 >= madFrameLength) {
qWarning() << "Skipping MP3 frame with invalid length"
<< madFrameLength
<< "in:" << m_file.fileName();
// Skip frame
continue;
}
const SINT madChannelCount = MAD_NCHANNELS(&madHeader);
if (isValidChannelCount(maxChannelCount) && (madChannelCount != maxChannelCount)) {
qWarning() << "Differing number of channels"
<< madChannelCount << "<>" << maxChannelCount
<< "in some MP3 frame headers:"
<< m_file.fileName();
}
maxChannelCount = math_max(madChannelCount, maxChannelCount);
const int samplingRateIndex = getIndexBySamplingRate(madSampleRate);
if (samplingRateIndex >= kSamplingRateCount) {
qWarning() << "Invalid sample rate:" << m_file.fileName()
<< madSampleRate;
// Abort
mad_header_finish(&madHeader);
return OpenResult::FAILED;
}
// Count valid frames separated by its sampling rate
headerPerSamplingRate[samplingRateIndex]++;
addSeekFrame(m_curFrameIndex, m_madStream.this_frame);
// Accumulate data from the header
sumBitrate += madHeader.bitrate;
// Update current stream position
m_curFrameIndex += madFrameLength;
DEBUG_ASSERT(m_madStream.this_frame);
DEBUG_ASSERT(0 <= (m_madStream.this_frame - m_pFileData));
} while (quint64(m_madStream.this_frame - m_pFileData) < m_fileSize);
mad_header_finish(&madHeader);
if (MAD_ERROR_NONE != m_madStream.error) {
// Unreachable code for recoverable errors
DEBUG_ASSERT(!MAD_RECOVERABLE(m_madStream.error));
if (MAD_ERROR_BUFLEN != m_madStream.error) {
qWarning() << "Unrecoverable MP3 header error:"
<< mad_stream_errorstr(&m_madStream);
// Abort
return OpenResult::FAILED;
}
}
if (m_seekFrameList.empty()) {
// This is not a working MP3 file.
qWarning() << "SSMP3: This is not a working MP3 file:"
<< m_file.fileName();
// Abort
return OpenResult::FAILED;
}
int mostCommonSamplingRateIndex = kSamplingRateCount; // invalid
int mostCommonSamplingRateCount = 0;
int differentRates = 0;
for (int i = 0; i < kSamplingRateCount; ++i) {
// Find most common sampling rate
if (mostCommonSamplingRateCount < headerPerSamplingRate[i]) {
mostCommonSamplingRateCount = headerPerSamplingRate[i];
mostCommonSamplingRateIndex = i;
differentRates++;
}
}
if (differentRates > 1) {
qWarning() << "Differing sampling rate in some headers:"
<< m_file.fileName();
for (int i = 0; i < kSamplingRateCount; ++i) {
if (0 < headerPerSamplingRate[i]) {
qWarning() << headerPerSamplingRate[i] << "MP3 headers with sampling rate" << getSamplingRateByIndex(i);
}
}
qWarning() << "MP3 files with varying sample rate are not supported!";
qWarning() << "Since this happens most likely due to a corrupt file";
qWarning() << "Mixxx tries to plays it with the most common sample rate for this file";
}
if (mostCommonSamplingRateIndex < kSamplingRateCount) {
setSamplingRate(getSamplingRateByIndex(mostCommonSamplingRateIndex));
} else {
qWarning() << "No single valid sampling rate in header";
// Abort
return OpenResult::FAILED;
}
// Initialize the AudioSource
setChannelCount(maxChannelCount);
setFrameCount(m_curFrameIndex);
// Calculate average values
m_avgSeekFrameCount = getFrameCount() / m_seekFrameList.size();
const unsigned long avgBitrate = sumBitrate / m_seekFrameList.size();
setBitrate(avgBitrate / 1000);
// Terminate m_seekFrameList
addSeekFrame(m_curFrameIndex, 0);
// Reset positions
m_curFrameIndex = getMinFrameIndex();
// Restart decoding at the beginning of the audio stream
m_curFrameIndex = restartDecoding(m_seekFrameList.front());
if (m_curFrameIndex != m_seekFrameList.front().frameIndex) {
qWarning() << "Failed to start decoding:" << m_file.fileName();
// Abort
return OpenResult::FAILED;
}
return OpenResult::SUCCEEDED;
}
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;
}
static int startread(sox_format_t * ft)
{
priv_t *p = (priv_t *) ft->priv;
size_t ReadSize;
sox_bool ignore_length = ft->signal.length == SOX_IGNORE_LENGTH;
int open_library_result;
LSX_DLLIBRARY_OPEN(
p,
mad_dl,
MAD_FUNC_ENTRIES,
"MAD decoder library",
mad_library_names,
open_library_result);
if (open_library_result)
return SOX_EOF;
p->mp3_buffer_size = sox_globals.bufsiz;
p->mp3_buffer = lsx_malloc(p->mp3_buffer_size);
ft->signal.length = SOX_UNSPEC;
if (ft->seekable) {
#ifdef USING_ID3TAG
read_comments(ft);
rewind((FILE*)ft->fp);
if (!ft->signal.length)
#endif
if (!ignore_length)
ft->signal.length = mp3_duration_ms(ft);
}
p->mad_stream_init(&p->Stream);
p->mad_frame_init(&p->Frame);
p->mad_synth_init(&p->Synth);
mad_timer_reset(&p->Timer);
ft->encoding.encoding = SOX_ENCODING_MP3;
/* 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.
*/
ReadSize = lsx_readbuf(ft, p->mp3_buffer, p->mp3_buffer_size);
if (ReadSize != p->mp3_buffer_size && ferror((FILE*)ft->fp))
return SOX_EOF;
p->mad_stream_buffer(&p->Stream, p->mp3_buffer, ReadSize);
/* Find a valid frame before starting up. This makes sure
* that we have a valid MP3 and also skips past ID3v2 tags
* at the beginning of the audio file.
*/
p->Stream.error = 0;
while (p->mad_frame_decode(&p->Frame,&p->Stream))
{
/* check whether input buffer needs a refill */
if (p->Stream.error == MAD_ERROR_BUFLEN)
{
if (sox_mp3_input(ft) == SOX_EOF)
return SOX_EOF;
continue;
}
/* Consume any ID3 tags */
sox_mp3_inputtag(ft);
/* 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.
*/
p->Stream.error = 0;
}
if (p->Stream.error)
{
lsx_fail_errno(ft,SOX_EOF,"No valid MP3 frame found");
return SOX_EOF;
}
switch(p->Frame.header.mode)
{
case MAD_MODE_SINGLE_CHANNEL:
case MAD_MODE_DUAL_CHANNEL:
case MAD_MODE_JOINT_STEREO:
case MAD_MODE_STEREO:
ft->signal.channels = MAD_NCHANNELS(&p->Frame.header);
break;
default:
lsx_fail_errno(ft, SOX_EFMT, "Cannot determine number of channels");
return SOX_EOF;
}
p->FrameCount=1;
p->mad_timer_add(&p->Timer,p->Frame.header.duration);
p->mad_synth_frame(&p->Synth,&p->Frame);
ft->signal.precision = MP3_MAD_PRECISION;
ft->signal.rate=p->Synth.pcm.samplerate;
if (ignore_length)
ft->signal.length = SOX_UNSPEC;
else {
ft->signal.length = (uint64_t)(ft->signal.length * .001 * ft->signal.rate + .5);
ft->signal.length *= ft->signal.channels; /* Keep separate from line above! */
}
p->cursamp = 0;
return SOX_SUCCESS;
}
enum mad_flow output(void *data,
struct mad_header const *header,
struct mad_pcm *pcm)
{
register int nsamples = pcm->length;
mad_fixed_t const *left_ch = pcm->samples[0], *right_ch = pcm->samples[1];
static unsigned char stream[1152*4]; /* 1152 because that's what mad has as a max; *4 because
there are 4 distinct bytes per sample (in 2 channel case) */
static unsigned int rate = 0;
static int channels = 0;
static struct audio_dither dither;
register char * ptr = stream;
register signed int sample;
register mad_fixed_t tempsample;
/* We need to know information about the file before we can open the playdevice
in some cases. So, we do it here. */
if (!playdevice)
{
channels = MAD_NCHANNELS(header);
rate = header->samplerate;
open_ao_playdevice(header);
}
else if ((channels != MAD_NCHANNELS(header) || rate != header->samplerate) && playdevice_is_live())
{
ao_close(playdevice);
channels = MAD_NCHANNELS(header);
rate = header->samplerate;
open_ao_playdevice(header);
}
if (pcm->channels == 2)
{
while (nsamples--)
{
tempsample = (mad_fixed_t)((*left_ch++ * (double)options.volume)/MAD_F_ONE);
sample = (signed int) audio_linear_dither(16, tempsample, &dither);
#ifndef WORDS_BIGENDIAN
*ptr++ = (unsigned char) (sample >> 0);
*ptr++ = (unsigned char) (sample >> 8);
#else
*ptr++ = (unsigned char) (sample >> 8);
*ptr++ = (unsigned char) (sample >> 0);
#endif
tempsample = (mad_fixed_t)((*right_ch++ * (double)options.volume)/MAD_F_ONE);
sample = (signed int) audio_linear_dither(16, tempsample, &dither);
#ifndef WORDS_BIGENDIAN
*ptr++ = (unsigned char) (sample >> 0);
*ptr++ = (unsigned char) (sample >> 8);
#else
*ptr++ = (unsigned char) (sample >> 8);
*ptr++ = (unsigned char) (sample >> 0);
#endif
}
ao_play(playdevice, stream, pcm->length * 4);
}
else if (options.opt & MPG321_FORCE_STEREO)
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 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;
}
/**
* 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 void mp3_audio_format(struct mp3_data *data, struct audio_format *af)
{
af->bits = 24;
af->sample_rate = (data->frame).header.samplerate;
af->channels = MAD_NCHANNELS(&(data->frame).header);
}
