path: root/main.c

/* This is a program that can read in a text file and convert it to a
   sound file.  Since this is going to depend on ffmpeg's libraries,
   otherwise I am only able to produce raw audio files, which are too
   large and not very convenient to use, a makefile is included to
   make the compiling process easier. */

/* include stuff */
#include <stdlib.h>
#include <stdio.h>
#include <math.h>

/* include FFMPEG stuff */
#include <libavcodec/avcodec.h>

#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/frame.h>
#include <libavutil/samplefmt.h>

/* Self-Dependencies */
#include "util.h"
#include "instrument.h"
#include "parser.h"

/* #include <libavformat/avformat.h> */

/* helper functions */

/* FFMPEG helpers */

/* check that a given sample format is supported by the encoder */
UNUSED
static int
check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt)
{
    const enum AVSampleFormat *p = codec->sample_fmts;

    /* the array is terminated by -1 */
    for (int i = 0; *(p+i) != -1; i++)
      fprintf(stderr, "supports %s\n", av_get_sample_fmt_name(*(p+i)));

    while (*p != AV_SAMPLE_FMT_NONE) {
        if (*p == sample_fmt)
            return 1;
        p++;
    }
    return 0;
}

__attribute__((__hot__, __unused__))
static void
/* encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
 *        FILE *output) */
encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt, FILE *output)
{
  int ret = 0;

  /* send the frame for encoding */
  ret = avcodec_send_frame(ctx, frame);
  if (ret < 0) {
    fprintf(stderr, "Error sending the frame to the encoder\n");
    exit(1);
  }

  /* read all the available output packets (in general there may be any
   * number of them */
  while (ret >= 0) {
    ret = avcodec_receive_packet(ctx, pkt);
    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
      return;
    else if (ret < 0) {
      fprintf(stderr, "Error encoding audio frame\n");
      exit(1);
    }

    fwrite(pkt->data, 1, pkt->size, output);
    /* av_write_frame(afc, pkt); */
    av_packet_unref(pkt);
  }
}

/* select layout with the highest channel count */
UNUSED
static int
select_channel_layout(const AVCodec *codec)
{
    const uint64_t *p;
    uint64_t best_ch_layout = 0;
    int best_nb_channels = 0;

    if (!codec->channel_layouts)
        return AV_CH_LAYOUT_STEREO;

    p = codec->channel_layouts;
    while (*p) {
        int nb_channels = av_get_channel_layout_nb_channels(*p);

        if (nb_channels > best_nb_channels) {
            best_ch_layout    = *p;
            best_nb_channels = nb_channels;
        }
        p++;
    }
    return best_ch_layout;
}

UHA_ATTR
static inline Pulse
square_wave(float counter)
{
  return (fmod(counter, 2*M_PI) < M_PI) ? 1 : -1;
}

UHA_ATTR
static inline Pulse
synthesize_wave(float theta)
{
  
  /* return -1.0 + fmod(theta, 2*M_PI)/M_PI; */
  
  /* return sin(theta); */

  /* return asin(sin(theta)) * 2/M_PI; */

  /* return pow(theta/2*M_PI, 2); */

  /* return sin(theta)+0.442*sin(2.0*theta)+0.315*sin(3.0*theta)
   *   +0.083*sin(4.0*theta)+cos(theta)+0.442*cos(2.0*theta); */
  
  /* piano like */
  
  /* return sin(theta)+1.869*sin(2.0*theta)+0.042*sin(3.0*theta)
   *   +0.022*sin(4.0*theta)+cos(theta); */

  /* excellent piano like */
  
  return ((0.65*sin(theta)+0.5*sin(2.0*theta))*exp(-0.004*theta));
  
}

/* This is replaced by make_sound function. */
/* UH_ATTR
 * static inline WaveFrag
 * oscillator(Volume v, Hertz h, Seconds s, Instrument *ins)
 * {
 *   
 *   return 0.0;
 * } */

UHA_ATTR
static inline float
frequency_modulation(float theta, Hertz h, float fm)
{
  return theta+FM_AMP*theta*sin(fm);
}

UH_ATTR
static WaveFrag
hstow(Volume v, Hertz h, Seconds s)
{
  LENT sample_num = (LENT) floor(SAMPLE_RATE * s);
  float attack_step = 1.0 / (float) (sample_num * ATTACK_P);
  float decay_step = 1.0 / (float) (sample_num * DECAY_P);
  float sustain_step = 1.0 / (float) (sample_num * SUSTAIN_P);
  float release_step = 1.0 / (float) (sample_num * RELEASE_P);

  float step = (float) (h * 2 * M_PI) / SAMPLE_RATE;

  float fm_step = (float) ((FM_FREQUENCY)*2*M_PI) / SAMPLE_RATE;
  
  unsigned char phase = 0, sustain_flag = 0;

  Wave w = MYALLOC(Pulse, sizeof(*w) * sample_num);
  
  float adsrcounter = 0.0, counter = 0.0, fm_counter = 0.0;
  float theta = 0.0, temp = 0.0;

  for (LENT i = 0; i < sample_num; i++, counter += step, fm_counter += fm_step) {
    /* theta = frequency_modulation(counter, (float) h*i/((float) SAMPLE_RATE), fm_counter); */
    theta=counter;
    temp = synthesize_wave(theta);
    temp += temp*temp*temp;
    temp = synthesize_wave(temp);
    temp += theta*exp(-2.0*theta);
    *(w+i) = (float) v * temp;
      /* (synthesize_wave(temp)/\*  +
       *   * 0.01*(2.0*(float)rand()/(float) RAND_MAX - 1.0) *\/) *//*  *
       * ((sustain_flag) ? SUSTAIN_LEVEL : adsrcounter) ;*/
    switch (phase) {
    case 0:                     /* attack phase */
      adsrcounter += attack_step;
      if (adsrcounter >= 1.0) {
        adsrcounter = 1.0;
        phase++;
      }
      break;
    case 1:                     /* decay phase */
      adsrcounter -= decay_step;
      if (adsrcounter <= SUSTAIN_LEVEL) {
        adsrcounter = 0.0;
        sustain_flag = 1;
        phase++;
      }
      break;
    case 2:                     /* sustain phase */
      adsrcounter += sustain_step;
      if (adsrcounter >= 1.0) {
        adsrcounter = SUSTAIN_LEVEL;
        sustain_flag = 0;
        phase++;
      }
      break;
    default:                    /* release phase */
      adsrcounter -= release_step;
      if (adsrcounter <= 0.0) adsrcounter = 0.0;
      break;
    }
  }
  
  return (WaveFrag) { w, sample_num };
}

H_ATTR
static WaveFrag
sbtow(Volume v, Semitones st, Beats b)
{
  return hstow(v, stoh(st), (b * BEAT_DUR));
}

/* compose */

D_ATTR("play_sheet")
static WaveFrag
compose(Volume *vs, Semitones *sts, Beats *bs, LENT len)
{
  WaveFrag wf, *temp = MYALLOC(WaveFrag, len);
  
  LENT total_len = 0;
  
  for (LENT i = 0; i < len; i++) {
    if (*(sts+i) >= -50)
      *(temp+i) = sbtow(*(vs+i), *(sts+i), *(bs+i));
    else
      *(temp+i) = sbtow(*(vs+i), *(sts+i), *(bs+i));
    total_len += (temp+i)->n;
  }

  wf.n = total_len;

  wf.w = MYALLOC(Pulse, total_len);
  
  LENT counter = 0;
  
  for (int i = 0; i < len; i++) {
    LENT local = (temp+i)->n;

    for (LENT j = 0; j < local; j++)
      *(wf.w+counter++) = *((temp+i)->w+j);
    
    free((temp+i)->w);
  }
  
  free(temp);
  
  return wf;
}

/* Reading */

/* The file should contain an even number of numbers.  For any natural
   number N, the 2*N th number specifies the Semitone of the Nth note,
   and the 2*N+1 number specifies the beats of the Nth note. */

U_ATTR
static void
read_sb(const char *str, int str_len,
        int *len, Semitones **sts, Beats **bs)
{
  int str_counter = 0;

  float temp = 0;

  int semi_len = 0, b_len = 0;
  Node_t semis = EMPTY_NODE, beats = EMPTY_NODE;
  Node_t sn = EMPTY_NODE, bn = EMPTY_NODE;
  Node_t *sp = NULL, *bp = NULL;

  LENT nth = 0;

  for (;str_counter < str_len; nth++) {
    int number_chars_read = 0;
    int result = sscanf(str+str_counter, "%f %n", &temp, &number_chars_read);
    if (result < 1) break;
    
    str_counter += number_chars_read;
    
    if (nth % 2) {
      /* odd */
      PUSH_NODE(b_len, bp, beats, temp, n);
    } else {
      /* even */
      PUSH_NODE(semi_len, sp, semis, temp, n);
    }
  }

  if (nth % 2) {
    fprintf(stderr, "There should be an even number of floats in "
            "the file, but found %lu.\n",
            nth);
    exit(1);
  }

  *len = floor(nth/2.0);
  *sts = realloc(*sts, sizeof(float) * (*len));
  *bs = realloc(*bs, sizeof(float) * (*len));
  
  for (int i = 0; i < *len; i++) {
    POP_NODE(semi_len, temp, semis, sn, n);
    *(*sts+*len-1-i) = temp;
    
    POP_NODE(b_len, temp, beats, bn, n);
    *(*bs+*len-1-i) = temp;
  }
}

/* main */

int main(int argc, char **argv)
{
  const AVCodec *codec;
  AVCodecContext *c = NULL;
  /* AVOutputFormat *of;
   * AVFormatContext *oc; */
  AVFrame *frame;
  AVPacket *pkt;
  int ret = 0;
  float *samples;

  /* output format setup */
  
  /* if (!(of = av_guess_format(NULL, DEFAULT_OUTPUT_NAME, "audio/aac"))) {
   *   fprintf(stderr, "cannot guess output format\n");
   *   exit(1);
   * } */
  
  /* avformat_alloc_output_context2(&oc, of, NULL, NULL); */
  /* oc = avformat_alloc_context();
   * 
   * if (!oc) {
   *   fprintf(stderr, "cannot allocate output context\n");
   *   exit(1);
   * } */

  /* oc->oformat = of; */
  /* oc->url = MYALLOC(char, sizeof(DEFAULT_OUTPUT_NAME));
   * for (int i = 0; i < sizeof(DEFAULT_OUTPUT_NAME); i++)
   *   oc->url[i] = DEFAULT_OUTPUT_NAME[i]; */
  
  /* ret = avio_open(&oc->pb, DEFAULT_OUTPUT_NAME, AVIO_FLAG_READ_WRITE);
   * 
   * if (ret<0) {
   *   fprintf(stderr, "cannot open %s\n", DEFAULT_OUTPUT_NAME);
   *   exit(1);
   * } */
  
  /* find the encoder */
  codec = avcodec_find_encoder(AV_CODEC_ID_MP3);
  if (!codec) {
    fprintf(stderr, "Codec not found\n");
    exit(1);
  }

  /* set an option to place MOOV atom first. */
  
  /* AVDictionary *opt = NULL;
   * av_dict_set(&opt, "movflags", "faststart", 0);
   * 
   * printf("of->audio_codec = %d\n", of->audio_codec);
   * 
   * if (oc->oformat->flags & AVFMT_GLOBALHEADER)
   *   c->flags |= AVFMT_GLOBALHEADER; */

  /* return 0; */
  
  /* STD_BASE = (float) pow(2, 1.0/12.0); */

  UNUSED int notes_len = 0;

  /* UNUSED Volume *vs = NULL;
   * Semitones *sts = MYALLOC(Semitones, 1);
   * Beats *bs = MYALLOC(Beats, 1); */

  char *notes_content = MYALLOC(char, 500);

  LENT notes_str_len = 0;

  notes_str_len = (LENT) read_entire_file("notes.txt", &notes_content);

#ifdef DEBUG
  fprintf(stderr, "notes content = %s\n", notes_content);
#endif

  PSheet psh = read_sheet(notes_content, notes_str_len);
  
  if (argc >= 2 && **(argv+1) == 'p')
    print_sheet(psh);
  
#ifdef DEBUG
  fprintf(stderr, "after printing\n");
#endif

  /* read_sb(notes_content, notes_str_len, &notes_len,
   *         &sts, &bs); */
  
#ifdef DEBUG
  fprintf(stderr, "notes len = %d\nfirst bs = %.2f\n",
          notes_len, *bs);
#endif

  /* vs = MYALLOC(Volume, notes_len);
   * for (int i = 0; i < notes_len;) *(vs+i++) = 0.3; */

  /* WaveFrag wf = compose(vs, sts, bs, notes_len); */
  WaveFrag wf = play_sheet(psh, 0.3);
  
  destroy_sheet(psh);

  printf("wf n = %lu\n", wf.n);

  /* AVStream *os = avformat_new_stream(oc, NULL); */

  c = avcodec_alloc_context3(codec);
  if (!c) {
    fprintf(stderr, "Could not allocate audio codec context\n");
    exit(1);
  }

  /* check that the encoder supports FLTP pcm input */
  /* The P marks that this is "planar", i.e. the input is not
     interleaved between channels. */
  c->sample_fmt = AV_SAMPLE_FMT_FLTP;
  if (!check_sample_fmt(codec, c->sample_fmt)) {
    fprintf(stderr, "Encoder does not support sample format %s",
            av_get_sample_fmt_name(c->sample_fmt));
    exit(1);
  }
  
  /* set other audio parameters supported by the encoder */
  c->sample_rate = (int) SAMPLE_RATE;
  c->channel_layout = select_channel_layout(codec);
  c->channels = av_get_channel_layout_nb_channels(c->channel_layout);

  /* this should be sizeof(float) * 8 * nb_channels * SAMPLE_RATE 

     We divide by 1000 since the unit is KB.*/
  c->bit_rate = sizeof(float) * 8 * SAMPLE_RATE * c->channels / 1000;

  /* os->time_base = (AVRational) { 1, (int) SAMPLE_RATE }; */

  /* open it */
  if (avcodec_open2(c, codec, NULL) < 0) {
    fprintf(stderr, "Could not open codec\n");
    exit(1);
  }

  /* /\* set stream data *\/
   * os->codecpar->sample_rate = SAMPLE_RATE;
   * os->codecpar->format = c->sample_fmt;
   * os->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
   * os->codecpar->codec_id = of->audio_codec;
   * os->codecpar->bit_rate = c->bit_rate;
   * os->codecpar->channels = c->channels;
   * os->codecpar->channel_layout = c->channel_layout;
   * os->codecpar->frame_size = c->frame_size;
   * 
   * if (avformat_write_header(oc, &opt) < 0) {
   *   fprintf(stderr, "cannot write header\n");
   *   exit(1);
   * }
   * 
   * av_dump_format(oc, 0, DEFAULT_OUTPUT_NAME, 1); */
  
  /* return 0; */

  /* audio_st.st->time_base = (AVRational){ 1, c->sample_rate };
   * 
   * if (oc->oformat->flags & AVFMT_GLOBALHEADER)
   *   c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER; */

  /* av_dump_format(oc, 0, DEFAULT_OUTPUT_NAME, 1); */

  /* ret = avio_open(&oc->pb, DEFAULT_OUTPUT_NAME, AVIO_FLAG_WRITE);
   * if (ret < 0) {
   *   fprintf(stderr, "Could not open '%s': %s\n", DEFAULT_OUTPUT_NAME,
   *           av_err2str(ret));
   *   return 1;
   * }
   * 
   * ret = avformat_write_header(oc, &opt);
   * if (ret < 0) {
   *   fprintf(stderr, "Error occurred when opening output file: %s\n",
   *           av_err2str(ret));
   *   return 1;
   * } */

  printf("sample rate = %d, channels = %d, bit rate = %lu\n"
         "frame size = %d\n",
         c->sample_rate, c->channels, (long) c->bit_rate,
         c->frame_size);
  /* return 0; */

  FILE *file = fopen(DEFAULT_OUTPUT_NAME, "wb");
  if (!file) {
    fprintf(stderr, "Could not open %s\n", DEFAULT_OUTPUT_NAME);
    exit(1);
  }

  /* packet for holding encoded output */
  pkt = av_packet_alloc();
  if (!pkt) {
    fprintf(stderr, "could not allocate the packet\n");
    exit(1);
  }

  /* pkt->stream_index = os->index; */

  /* frame containing input raw audio */
  frame = av_frame_alloc();
  if (!frame) {
    fprintf(stderr, "Could not allocate audio frame\n");
    exit(1);
  }
  
  frame->nb_samples     = c->frame_size;
  frame->format         = c->sample_fmt;
  frame->channel_layout = c->channel_layout;
  
  /* allocate the data buffers */
  ret = av_frame_get_buffer(frame, 0);
  if (ret < 0) {
    fprintf(stderr, "Could not allocate audio data buffers\n");
    fprintf(stderr, "nbs = %d, format = %s, layout = %d\n",
            frame->nb_samples, av_get_sample_fmt_name(frame->format),
            av_get_channel_layout_nb_channels(frame->channel_layout));
    exit(1);
  }

  /* write to the frame */

  long frame_num = floor((float) wf.n/(float) c->frame_size) + 1;
  long w_counter = 0;
  
  for (int i = 0; i < frame_num; i++, w_counter += c->frame_size) {
    ret = av_frame_make_writable(frame);
    if (ret < 0) {
      fprintf(stderr, "cannot make frame writable\n");
      exit(1);
    }

    for (int j = 0; j < c->channels; j++) {
      samples = (float *)frame->data[j];
      
      for (int k = 0; k < c->frame_size && k+w_counter < wf.n; k++)
        samples[k] = *(wf.w+w_counter+k);
    }

    encode(c, frame, pkt, file);
  }

  /* fwrite(wf.w, sizeof(float), wf.n, file); */

  /* flush the encoder */
  encode(c, NULL, pkt, file);

  /* free(vs); */
  /* free(sts);
   * free(bs); */
  free(wf.w);
  free(notes_content);
  fclose(file);
  
  av_frame_free(&frame);
  av_packet_free(&pkt);
  avcodec_free_context(&c);
  /* av_dict_free(&opt); */

  return 0;
}