/*
 * a.out <audiofile> <sampleskew>
 *
 * This program reads in a raw file (assuming 16bit PCM 44.1kHz mono), adjusts
 * the sample rate, and writes the audio to pulseaudio.
 */

#include <assert.h>
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include <time.h>
#include <pulse/simple.h>
#include "libavcodec/avcodec.h"

#define LENGTH_MS 500       // how many milliseconds of speech to store
#define RATE 44100      // the sampling rate
#define FORMAT PA_SAMPLE_S16NE  // sample size: 8 or 16 bits
#define CHANNELS 1      // 1 = mono 2 = stereo
#define ANGLE_STEP_PER_SEC 15.0 // degrees per second

#define DEG2RAD( angle ) (((angle)/180.0)*3.14159265)

/* Globals */
FILE* infile = 0;
char* filename = 0;
int rate_skew = 0; // Num samples to skew
long file_len = 0;
pa_simple* pulse_conn;
pa_sample_spec spec;
struct AVResampleContext* audio_cntx = 0;
double angle = 0.0;
double prev_time = 0.0;

/* this buffer holds the digitized audio */
char in_buffer[(LENGTH_MS*RATE*16*CHANNELS)/8000];
char out_buffer[ sizeof( in_buffer ) * 4];

/**/
int calc_skew()
{
    // Calculate how much time has elapsed sinse the last call.
    struct timeval now_tv;
    gettimeofday( &now_tv, 0 );
    double now = (double)now_tv.tv_sec + ((double)now_tv.tv_usec / 1000000.0);
    double delta = 0.0;
    if( prev_time )
        delta = now - prev_time;
    prev_time = now;

    // Calculate how much to increase the angle by.
    angle += ANGLE_STEP_PER_SEC * delta; // 'delta' is normalized to seconds so this works.
    if( angle >= 360.0 )
        angle = angle - 360.0;

    return( rate_skew * sin( DEG2RAD(angle) ) );
}

/**/
void play_audio()
{
    size_t bytes_read = 0;
    size_t total_bytes_read = 0;
    // Loop through the file until all data is read.
    for( ; !feof( infile ) && !ferror( infile ); )
    {
        // Read the next chunk of data from the file.
        bytes_read = fread( in_buffer, 1, sizeof( in_buffer ), infile );
        assert( !ferror( infile ) && "Error reading audio file!" );
        total_bytes_read += bytes_read;

        // Resample audio.
        int out_rate = RATE + calc_skew();
        audio_cntx = av_resample_init( out_rate,    // out rate
            RATE,   // in rate
            16, // filter length
            10, // phase count
            0,  // linear FIR filter
            1.0 );  // cutoff frequency
        assert( audio_cntx && "Failed to create resampling context!" );

        int samples_consumed = 0;
        int samples_output = av_resample( audio_cntx,
            (short*)out_buffer,
            (short*)in_buffer,
            &samples_consumed,
            bytes_read / 2,
            sizeof( out_buffer ) / 2, // in samples
            0 );
        assert( samples_output > 0 && "Error calling av_resample()!" );

        av_resample_close( audio_cntx );

        // Play the processed samples.
        pa_simple_write( pulse_conn,
            out_buffer,
            samples_output * 2,
            0 );

        // Display progress text.
        printf( "\rProgress: %2.1f%% Complete, Skew: %2.1f%%, Angle: %2.1f Degrees       ",
            ((float)total_bytes_read/(float)file_len)*100.0,
            ((float)(out_rate)/(float)RATE)*100.0,
            angle );
        fflush( stdout );
    }
    printf( "\nDone!\n" );
}

/**/
int main(int argc, char *argv[])
{
    assert( argc > 2 && "Please specify a file and skew!" );
    filename = argv[1];
    rate_skew = atoi( argv[2] );

    // Initialize ffmpeg resampling.
    avcodec_init();

    // Open audio file.
    infile = fopen( filename, "rb" );
    assert( infile && "Error opening file!" );
    fseek( infile, 0, SEEK_END );
    file_len = ftell( infile );
    fseek( infile, 0, SEEK_SET );

    // Open connection to pulse audio server.
    spec.format = FORMAT;
    spec.channels = CHANNELS;
    spec.rate = RATE;
    pulse_conn = pa_simple_new( NULL, // Use the default server.
        "audio_skew_rate",  // Our application's name.
        PA_STREAM_PLAYBACK,
        NULL,           // Use the default device.
        "Music",        // Description of our stream.
        &spec,          // Our sample format.
        NULL,           // Use default channel map
        NULL,           // Use default buffering attributes.
        NULL );         // Ignore error code.


    // Play audio.
    play_audio();

    // Clean up
    pa_simple_free( pulse_conn );
    fclose( infile );

    return( 0 );
}