/**
 * 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");
//.........这里部分代码省略.........

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(&current_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(&current_time);

        mad_timer_add(&time_remaining, current_time);
        mad_timer_negate(&current_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));
        }
//.........这里部分代码省略.........

/*
 * 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];
//.........这里部分代码省略.........