/************************************************************************************************************
 * WavPlay AM824
 * Copyright (C) 2020, Dolby Laboratories Inc.
 * This program is free software: you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation, either version 3 of the License, or
 * (at your option) any later version.
 *
 * This program is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 * You should have received a copy of the GNU General Public License
 * along with this program.  If not, see <http://www.gnu.org/licenses/>
 ************************************************************************************************************/

#define VERSION_STRING "1.0"

#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
#include <string.h>
#include <stdarg.h>
#include "portaudio.h"
#include "am824_framer.h"

#if defined(__linux__)
#include "pa_linux_alsa.h"
#endif

#if defined(_WIN32) || defined(_WIN64)
#include "pa_win_wasapi.h"

#include <windows.h>
#include <Ks.h>
#include <Ksmedia.h>
#include <mmreg.h>

#if PA_USE_ASIO
#include "pa_asio.h"
#endif
#else
#include <sndfile.h>
#endif

typedef enum {
  FF_PCM = 0,
  FF_FLOAT32,
} FileFormat;

typedef enum {
  ERR_FILE_NOT_FOUND = -101,
  ERR_BAD_INPUT_FILE,
  ERR_BAD_WAV_FORMAT,
  ERR_INCOMPLETE_INPUT_FILE,
  ERR_NOT_SUPPORTED,
  ERR_BAD_CMD_OPTION,
  ERR_NO_MEMORY,
  ERR_NO_DEVICE,
  ERR_NO_STREAM,
  ERR_PORTAUDIO,
  ERR_OK = 0
} ErrorCode;

typedef struct {
  unsigned long frameIndex; /* Index into sample array. */
  unsigned long maxFrameIndex;
  unsigned int numChannels;
  unsigned int bytesPerSample;
  unsigned char* audio;
  FileFormat waveFileFormat;
  unsigned int bitsPerSample;
  unsigned int fs;
  unsigned int blockAlign;
  unsigned long totalBytes;
#if defined(_WIN32) || defined(_WIN64)
  HMMIO waveFile;
#else
  SNDFILE* waveFile;
#endif
  unsigned int am824_audio;
  AM824SamplingFrequency am824_fs;
  unsigned int am824_fs_match_wavfile;
  unsigned int am824_professional;
} UserData;

void throwError(ErrorCode err, const char* format, ...) {
  va_list args;
  va_start(args, format);

  fprintf(stderr, "***Error: ");
  vfprintf(stderr, format, args);
  va_end(args);
  fprintf(stderr, "\n");
  fflush(stderr);
  exit(err);
}

/* This is the main callback functioned called by PortAudio. It is registered
 * when the stream is created */

static int playCallback(const void* inputBuffer,
                        void* outputBuffer,
                        unsigned long framesPerBuffer,
                        const PaStreamCallbackTimeInfo* timeInfo,
                        PaStreamCallbackFlags statusFlags,
                        void* userData) {
  UserData* data = (UserData*)userData;
  unsigned char* rptr =
      &data->audio[data->frameIndex * data->numChannels * data->bytesPerSample];
  unsigned char* wptr = (unsigned char*)outputBuffer;
  unsigned int i;
  int finished;
  unsigned int framesLeft = data->maxFrameIndex - data->frameIndex;

  (void)inputBuffer; /* Prevent unused variable warnings. */
  (void)timeInfo;
  (void)statusFlags;
  (void)userData;

  if (framesLeft < framesPerBuffer) {
    /* final buffer... */
    for (i = 0; i < framesLeft * data->numChannels * data->bytesPerSample;
         i++) {
      *wptr++ = *rptr++;
    }
    for (; i < framesPerBuffer * data->numChannels * data->bytesPerSample;
         i++) {
      *wptr++ = 0;
    }
    data->frameIndex += framesLeft;
    finished = paComplete;
  } else {
    for (i = 0; i < framesPerBuffer * data->numChannels * data->bytesPerSample;
         i++) {
      *wptr++ = *rptr++;
    }
    data->frameIndex += framesPerBuffer;
    finished = paContinue;
  }
  return finished;
}

/* Two versions of the following two functions exist, one for Windows and one
   for Linux. This approach was chosen because a platform independent wav file
   library was not used but rather Windows API and libasound directly */

#if defined(_WIN32) || defined(_WIN64)
int read_wav_file_header(char* playbackWaveFile, /* Input file name string */
                         UserData* outputData)   /* Output options */
{
  MMCKINFO mmckinfoParent;
  MMCKINFO mmckinfoSubchunk;
  WAVEFORMATEXTENSIBLE* format;

  outputData->waveFile =
      mmioOpenA(playbackWaveFile, 0, MMIO_READ | MMIO_ALLOCBUF);
  if (!outputData->waveFile) {
    throwError(ERR_FILE_NOT_FOUND, "Can't Open %s!", playbackWaveFile);
  }
  mmckinfoParent.fccType = mmioFOURCC('W', 'A', 'V', 'E');
  if (mmioDescend(outputData->waveFile, (LPMMCKINFO)&mmckinfoParent, 0,
                  MMIO_FINDRIFF)) {
    throwError(ERR_BAD_INPUT_FILE, "This file doesn't contain a WAVE!");
  }
  mmckinfoSubchunk.ckid = mmioFOURCC('f', 'm', 't', ' ');
  if (mmioDescend(outputData->waveFile, &mmckinfoSubchunk, &mmckinfoParent,
                  MMIO_FINDCHUNK)) {
    throwError(ERR_BAD_WAV_FORMAT, "Required fmt chunk was not found!");
  }

  format = (WAVEFORMATEXTENSIBLE*)malloc(mmckinfoSubchunk.cksize);

  if (mmioRead(outputData->waveFile, (HPSTR)format, mmckinfoSubchunk.cksize) !=
      (LRESULT)mmckinfoSubchunk.cksize) {
    throwError(ERR_BAD_WAV_FORMAT, "Reading the fmt chunk!");
  }

  if ((format->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) &&
      (format->Samples.wValidBitsPerSample != format->Format.wBitsPerSample)) {
    throwError(ERR_NOT_SUPPORTED,
               "Different container size and bit depth not supported");
  }

  mmioAscend(outputData->waveFile, &mmckinfoSubchunk, 0);
  mmckinfoSubchunk.ckid = mmioFOURCC('d', 'a', 't', 'a');
  if (mmioDescend(outputData->waveFile, &mmckinfoSubchunk, &mmckinfoParent,
                  MMIO_FINDCHUNK)) {
    throwError(ERR_BAD_WAV_FORMAT, "Reading the data chunk!");
  }

  outputData->fs = format->Format.nSamplesPerSec;
  outputData->numChannels = format->Format.nChannels;
  outputData->totalBytes = mmckinfoSubchunk.cksize;
  outputData->bitsPerSample = format->Format.wBitsPerSample;
  outputData->bytesPerSample = outputData->bitsPerSample / 8;
  outputData->blockAlign = format->Format.nBlockAlign;

  if (format->Format.wFormatTag == WAVE_FORMAT_PCM) {
    outputData->waveFileFormat = FF_PCM;
  } else if (format->Format.wFormatTag == WAVE_FORMAT_IEEE_FLOAT) {
    outputData->waveFileFormat = FF_FLOAT32;
  } else if (format->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
    if (format->SubFormat == KSDATAFORMAT_SUBTYPE_PCM) {
      outputData->waveFileFormat = FF_PCM;
    } else if (format->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT) {
      outputData->waveFileFormat == FF_FLOAT32;
    } else {
      throwError(ERR_NOT_SUPPORTED,
                 "Error: Unsupported WAVEFORMAT EXTENSIBLE SUBTYPE!");
    }
  }

  return (0);
}

unsigned long read_entire_wav_file(
    UserData* outputData, /* Input options*/
    void* audioData)      /* Output data read form file */
{
  unsigned long readCount;

  readCount =
      mmioRead(outputData->waveFile, (char*)audioData, outputData->totalBytes);

  mmioClose(outputData->waveFile, 0);

  if (readCount != outputData->totalBytes) {
    throwError(ERR_INCOMPLETE_INPUT_FILE,
               "Failed to read all of audio data in wave file");
  }
  return (readCount);
}

#else
int read_wav_file_header(char* playbackWaveFile, /* Input filename string */
                         UserData* outputData)   /* Output options */
{
  uint32_t waveFormat;
  SF_INFO fileInfo;

  outputData->waveFile = sf_open(playbackWaveFile, SFM_READ, &fileInfo);
  if (!outputData->waveFile) {
    throwError(ERR_FILE_NOT_FOUND, "File %s not found\n", playbackWaveFile);
  }

  outputData->fs = fileInfo.samplerate;
  outputData->numChannels = fileInfo.channels;

  waveFormat = fileInfo.format;
  if ((waveFormat & SF_FORMAT_TYPEMASK) == SF_FORMAT_RF64) {
    throwError(ERR_NOT_SUPPORTED, "RF64 format not yet supported");
  }

  if (((waveFormat & SF_FORMAT_TYPEMASK) != SF_FORMAT_WAV) &&
      ((waveFormat & SF_FORMAT_TYPEMASK) != SF_FORMAT_WAVEX)) {
    throwError(ERR_BAD_INPUT_FILE, "Input file is not a wavefile");
  }

  switch (waveFormat & SF_FORMAT_SUBMASK) {
    case SF_FORMAT_PCM_16:
      outputData->waveFileFormat = FF_PCM;
      outputData->bitsPerSample = 16;
      break;
    case SF_FORMAT_PCM_24:
      outputData->waveFileFormat = FF_PCM;
      outputData->bitsPerSample = 24;
      break;
    case SF_FORMAT_PCM_32:
      outputData->waveFileFormat = FF_PCM;
      outputData->bitsPerSample = 32;
      break;
    case SF_FORMAT_FLOAT:
      outputData->waveFileFormat = FF_FLOAT32;
      outputData->bitsPerSample = 32;
      break;
    default:
      throwError(ERR_NOT_SUPPORTED, "Unsupported wavefile format supported");
  }

  outputData->bytesPerSample = outputData->bitsPerSample / 8;
  outputData->blockAlign = outputData->numChannels * outputData->bytesPerSample;
  outputData->totalBytes = outputData->blockAlign * fileInfo.frames;

  return (0);
}

unsigned long read_entire_wav_file(
    UserData* outputData, /* Input Options */
    void* audioData)      /* Output Audio data from file */
{
  unsigned long readCount;

  readCount =
      sf_read_raw(outputData->waveFile, audioData, outputData->totalBytes);
  if (readCount != outputData->totalBytes) {
    throwError(ERR_INCOMPLETE_INPUT_FILE,
               "Failed to read all of audio data in wave file");
  }
  return (readCount);
}

#endif

unsigned int countBits(unsigned int a) {
  unsigned int count = 0;
  while (a) {
    count += (a & 0x1);
    a >>= 1;
  }
  return (count);
}

/* This function takes standard PCM audio plus channel status options and
 * creates the samples for AM824 format */

void am824Convert(UserData* userData, /* Input options */
                  void* audio,        /* Input PCM samples */
                  void** am824audio)  /* Output AM824 samples, always 32bit */
{
  unsigned long bytesConverted = 0;
  uint32_t inputSample32;
  uint16_t inputSample16;
  uint8_t* inputPtr;
  uint32_t* outputPtr;
  unsigned int channel;
  unsigned long outputMemSize;
  AM824ErrorCode err;

  if ((userData->bytesPerSample != 3) && (userData->bytesPerSample != 2)) {
    throwError(ERR_NOT_SUPPORTED,
               "Only 2 or 3 bytes per sample supported for AM824 mode");
  }

  AM824Framer framer(userData->numChannels, userData->bytesPerSample * 8,
                     AM824_LITTLE_ENDIAN, err);
  if (err != AM824_ERR_OK) {
    if (err == AM824_ERR_UNSUPPORTED_BITDEPTH) {
      throwError(ERR_NOT_SUPPORTED,
                 "AM824 framer reports bitdepth %d not supported",
                 userData->bytesPerSample * 8);
    }
  }

  // framer.testCRC();

  if (userData->am824_audio) {
    framer.setAudioMode();
  } else {
    framer.setDataMode();
  }
  if (userData->am824_fs_match_wavfile) {
    switch (userData->fs) {
      case 32000:
        framer.setSamplingFrequency(FS_32000_HZ);
        break;
      case 44100:
        framer.setSamplingFrequency(FS_44100_HZ);
        break;
      case 48000:
        framer.setSamplingFrequency(FS_48000_HZ);
        break;
      default:
        framer.setSamplingFrequency(FS_NOT_INDICATED);
    }
  } else {
    framer.setSamplingFrequency(userData->am824_fs);
  }
  if (userData->am824_professional) {
    framer.setProfessionalMode();
  } else {
    framer.setConsumerMode();
  }
  outputMemSize =
      (userData->totalBytes * sizeof(uint32_t)) / userData->bytesPerSample;
  *am824audio = malloc(outputMemSize);
  outputPtr = (uint32_t*)*am824audio;
  inputPtr = (uint8_t*)audio;

  while (bytesConverted < userData->totalBytes) {
    for (channel = 0; channel < userData->numChannels; channel++) {
      if (userData->bytesPerSample == 3) {
        inputSample32 = *((uint32_t*)inputPtr) & 0xffffff;
        framer.getAM824Sample(inputSample32, (uint8_t*)outputPtr);
      } else {
        inputSample16 = *((uint16_t*)inputPtr);
        framer.getAM824Sample(inputSample16, (uint8_t*)outputPtr);
      }
      outputPtr += 1;
      inputPtr += userData->bytesPerSample;
      bytesConverted += userData->bytesPerSample;
    }
  }
  userData->totalBytes =
      (userData->totalBytes * sizeof(uint32_t)) / userData->bytesPerSample;
  userData->bytesPerSample = sizeof(uint32_t);
  userData->bitsPerSample = 32;
  userData->blockAlign = userData->numChannels * userData->bytesPerSample;
}

void list_devices(void) {
  int i, numDevices, defaultDisplayed;
  const PaDeviceInfo* deviceInfo;

  printf("PortAudio version: 0x%08X\n", Pa_GetVersion());
  printf("Version text: '%s'\n", Pa_GetVersionText());

  numDevices = Pa_GetDeviceCount();
  if (numDevices < 0) {
    throwError(ERR_NO_DEVICE, "Pa_GetDeviceCount returned 0x%x\n", numDevices);
  }

  printf("Number of devices = %d\n", numDevices);
  for (i = 0; i < numDevices; i++) {
    deviceInfo = Pa_GetDeviceInfo(i);
    if (deviceInfo->maxOutputChannels > 0) {
      printf("--------------------------------------- device #%d\n", i);

      /* Mark global and API specific default devices */
      defaultDisplayed = 0;
      if (i == Pa_GetDefaultOutputDevice()) {
        printf("[ Default Output");
        defaultDisplayed = 1;
      } else if (i ==
                 Pa_GetHostApiInfo(deviceInfo->hostApi)->defaultOutputDevice) {
        const PaHostApiInfo* hostInfo = Pa_GetHostApiInfo(deviceInfo->hostApi);
        printf("[ Default %s Output", hostInfo->name);
        defaultDisplayed = 1;
      }

      if (defaultDisplayed)
        printf(" ]\n");

        /* print device info fields */
#ifdef WIN32
      { /* Use wide char on windows, so we can show UTF-8 encoded device names
         */
        wchar_t wideName[MAX_PATH];
        MultiByteToWideChar(CP_UTF8, 0, deviceInfo->name, -1, wideName,
                            MAX_PATH - 1);
        wprintf(L"Name                        = %s\n", wideName);
      }
#else
      printf("Name                        = %s\n", deviceInfo->name);
#endif
      printf("Host API                    = %s\n",
             Pa_GetHostApiInfo(deviceInfo->hostApi)->name);
      printf("Max output channels          = %d\n",
             deviceInfo->maxOutputChannels);

      printf("Default low output latency   = %4.4f\n",
             deviceInfo->defaultLowOutputLatency);
      printf("Default high output latency  = %4.4f\n",
             deviceInfo->defaultHighOutputLatency);

#ifdef WIN32
#if PA_USE_ASIO
      /* ASIO specific latency information */
      if (Pa_GetHostApiInfo(deviceInfo->hostApi)->type == paASIO) {
        long minLatency, maxLatency, preferredLatency, granularity;

        err = PaAsio_GetAvailableLatencyValues(i, &minLatency, &maxLatency,
                                               &preferredLatency, &granularity);

        printf("ASIO minimum buffer size    = %ld\n", minLatency);
        printf("ASIO maximum buffer size    = %ld\n", maxLatency);
        printf("ASIO preferred buffer size  = %ld\n", preferredLatency);

        if (granularity == -1)
          printf("ASIO buffer granularity     = power of 2\n");
        else
          printf("ASIO buffer granularity     = %ld\n", granularity);
      }
#endif /* PA_USE_ASIO */
#endif /* WIN32 */

      printf("Default sample rate         = %8.2f\n",
             deviceInfo->defaultSampleRate);
    }
  }
  return;
}

void print_usage(void) {
  fprintf(stderr, "wavplay_am824 [OPTION]... <OUTPUT FILE> v%s\n",
          VERSION_STRING);
  fprintf(stderr,
          "Copyright Dolby Laboratories Inc., 2020. All rights reserved.\n\n");
  fprintf(stderr, "-h                   Display this messgage\n");
  fprintf(stderr, "-ld                  List playback devices\n");
  fprintf(stderr,
          "-l                   Preferred playout latency in seconds\n");
  fprintf(stderr,
          "-buf <samples>       Playout buffer size in frames (samples x "
          "channels)\n");
  fprintf(stderr, "-d <index>           Device index to use for playback\n");
  fprintf(stderr,
          "-am824               Using virtual sound card feeding an "
          "AM824/2110-31 stream\n");
  fprintf(stderr,
          "                     The following keywords can follow the '-am824' "
          "switch to modify channel status:\n");
  fprintf(stderr,
          "    audio, nonaudio, fs_not_indicated, fs_48k, fs_441k, fs_32k, "
          "professional, consumer\n");
#if defined(_WIN32) || defined(_WIN64)
  fprintf(stderr,
          "-e <index>           Uses WASPI exclusive mode if selected device "
          "is a WASAPI device\n");
#endif
  fprintf(stderr, "\n");
}

/*******************************************************************/
int main(int argc, char* argv[]) {
  PaStreamParameters outputParameters;
  PaStream* stream;
  PaError err = paNoError;
  UserData data;
  int i;
  unsigned long bytesRead;
  unsigned int framesPerBuffer = 128;
  float userLatency = 0.0;
  char playbackWavFileName[256] = "";
  double sampleRate = 48000.0;
  double startTime/*,finishTime*/;
  void* audio;
#if defined(_WIN32) || defined(_WIN64)
  struct PaWasapiStreamInfo wasapiInfo;
  int waspiExclusiveMode = 0;
#endif
  unsigned int am824Mode = 0;
  void* am824audio;

  outputParameters.device = paNoDevice;
  audio = NULL;

  err = Pa_Initialize();

  if (err != paNoError) {
    throwError(ERR_PORTAUDIO, "Pa_Initialize returned %d, %s", err,
               Pa_GetErrorText(err));
  }

  if (argc < 2) {
    print_usage();
    list_devices();
    exit(0);
  }

  for (i = 1; i < argc; i++) {
    // check to see if its a filename first
    if ((argv[i][0] != '-') && (strlen(playbackWavFileName) == 0)) {
      strcpy(playbackWavFileName, argv[i]);
    } else if (!strcmp(argv[i], "-d")) {
      if (i == (argc - 1)) {
        print_usage();
        throwError(ERR_NO_DEVICE, "Can't find soundcard index");
      }
      outputParameters.device = atoi(argv[i + 1]);
      // We increment i here to step over the next parameter
      // which has been parsed as the value
      i++;
    }
#if defined(_WIN32) || defined(_WIN64)
    else if (!strcmp(argv[i], "-e")) {
      waspiExclusiveMode = 1;
    }
#endif
    else if (!strcmp(argv[i], "-h") || !strcmp(argv[i], "-help") ||
             !strcmp(argv[i], "--help")) {
      print_usage();
      exit(0);
    } else if (!strcmp(argv[i], "-buf")) {
      if (i == (argc - 1)) {
        print_usage();
        throwError(ERR_NO_DEVICE, "Can't find buffer size");
      }
      framesPerBuffer = atoi(argv[i + 1]);
      // We increment i here to step over the next parameter
      // which has been parsed as the value
      i++;
    } else if (!strcmp(argv[i], "-l")) {
      if (i == (argc - 1)) {
        print_usage();
        throwError(ERR_NO_DEVICE, "Can't find latency");
      }
      userLatency = atof(argv[i + 1]);
      // We increment i here to step over the next parameter
      // which has been parsed as the value
      i++;
    } else if (!strcmp(argv[i], "-ld")) {
      list_devices();
      exit(0);
    } else if (!strcmp(argv[i], "-am824")) {
      am824Mode = 1;
      data.am824_audio = 1;
      data.am824_professional = 0;
      data.am824_fs_match_wavfile = 1;
      i++;
      while ((i < argc) && (argv[i][0] != '-')) {
        if (!strcmp(argv[i], "audio")) {
          data.am824_audio = 1;
        } else if (!strcmp(argv[i], "nonaudio")) {
          data.am824_audio = 0;
        } else if (!strcmp(argv[i], "fs_not_indicated")) {
          data.am824_fs = FS_NOT_INDICATED;
          data.am824_fs_match_wavfile = 0;
        } else if (!strcmp(argv[i], "fs_48k")) {
          data.am824_fs = FS_48000_HZ;
          data.am824_fs_match_wavfile = 0;

        } else if (!strcmp(argv[i], "fs_441k")) {
          data.am824_fs = FS_44100_HZ;
          data.am824_fs_match_wavfile = 0;
        } else if (!strcmp(argv[i], "fs_32k")) {
          data.am824_fs = FS_32000_HZ;
          data.am824_fs_match_wavfile = 0;
        } else if (!strcmp(argv[i], "professional")) {
          data.am824_professional = 1;
        } else if (!strcmp(argv[i], "consumer")) {
          data.am824_professional = 0;
        } else {
          break;
        }
        i++;
      }
      i--;
    } else {
      print_usage();
      throwError(ERR_BAD_CMD_OPTION, "Option %s not recognized", argv[i]);
    }
  }

  if (read_wav_file_header(playbackWavFileName, &data)) {
    throwError(ERR_BAD_WAV_FORMAT, "Bad wav header");
  }

  audio = malloc(data.totalBytes);
  if (!audio) {
    throwError(ERR_NO_MEMORY, "Memory allocation failed");
  }

  bytesRead = read_entire_wav_file(&data, audio);
  if (bytesRead != data.totalBytes) {
    throwError(ERR_INCOMPLETE_INPUT_FILE, "Couldn't read all input data");
  }

  if (am824Mode) {
    printf("AM824 output mode selected\n");
    am824Convert(&data, audio, &am824audio);
    free(audio);
    audio = am824audio;
  }

  if (outputParameters.device == paNoDevice) {
    outputParameters.device = Pa_GetDefaultOutputDevice();
  }

  if (outputParameters.device == paNoDevice) {
    throwError(ERR_NO_DEVICE, "No default output device");
  }

#if defined(_WIN32) || defined(_WIN64)
  if ((waspiExclusiveMode) &&
      (Pa_GetHostApiInfo(Pa_GetDeviceInfo(outputParameters.device)->hostApi)
           ->type == paWASAPI)) {
    wasapiInfo.size = sizeof(PaWasapiStreamInfo);
    wasapiInfo.hostApiType = paWASAPI;
    wasapiInfo.version = 1;
    wasapiInfo.flags = (paWinWasapiExclusive | paWinWasapiThreadPriority);
    wasapiInfo.channelMask = 0;
    wasapiInfo.hostProcessorOutput = NULL;
    wasapiInfo.hostProcessorInput = NULL;
    wasapiInfo.threadPriority = eThreadPriorityProAudio;
    outputParameters.hostApiSpecificStreamInfo = (&wasapiInfo);
    printf("Detected WASAPI device and setting exclusive mode\n");
  } else {
#endif
    outputParameters.hostApiSpecificStreamInfo = NULL;
#if defined(_WIN32) || defined(_WIN64)
  }
#endif

  if (userLatency > 0.0) {
    outputParameters.suggestedLatency = userLatency;
  } else {
    outputParameters.suggestedLatency =
        Pa_GetDeviceInfo(outputParameters.device)->defaultHighOutputLatency;
  }

  switch (data.waveFileFormat) {
    case FF_PCM:

      if ((data.bitsPerSample != 8) && (data.bitsPerSample != 16) &&
          (data.bitsPerSample != 24) && (data.bitsPerSample != 32)) {
        throwError(ERR_NOT_SUPPORTED, "Unsupported WAVE_FORMAT_PCM bitdepth");
      }
      break;
    case FF_FLOAT32:
      if (data.bitsPerSample != 32) {
        throwError(
            ERR_BAD_WAV_FORMAT,
            "Wavefile indicated floating point but bits per sample is not 32");
      }
      break;
    default:
      throwError(ERR_NOT_SUPPORTED,
                 "Unsupported WAV format tag (WAVE_FORMAT_PCM & "
                 "WAVE_FORMAT_IEEE_FLOAT supported)");
  }

  if (data.bitsPerSample == 16) {
    outputParameters.sampleFormat = paInt16;
  } else if (data.bitsPerSample == 8) {
    outputParameters.sampleFormat = paInt8;
  } else if (data.bitsPerSample == 24) {
    outputParameters.sampleFormat = paInt24;
  } else if (data.bitsPerSample == 32) {
    if (data.waveFileFormat == FF_FLOAT32) {
      outputParameters.sampleFormat = paFloat32;
    } else {
      outputParameters.sampleFormat = paInt32;
    }
  } else {
    throwError(ERR_NOT_SUPPORTED, "Unsupported bitdepth %d",
               data.bitsPerSample);
  }

  outputParameters.channelCount = data.numChannels;
  sampleRate = (double)data.fs;
  err = Pa_IsFormatSupported(NULL, &outputParameters, sampleRate);
  if (err != paNoError) {
    throwError(ERR_NOT_SUPPORTED, "Pa_IsFormatSupported returned %d, %s", err,
               Pa_GetErrorText(err));
  }

  printf("Pa_IsFormatSupported succeeded\n");

  printf(
      "device: %u\nchannels: %u\nsampleFormat: %lu\nlatency: %f\nsampleRate: "
      "%u\n",
      outputParameters.device, outputParameters.channelCount,
      outputParameters.sampleFormat, outputParameters.suggestedLatency,
      data.fs);

  // Set callback parameters
  data.frameIndex = 0; /* Index into sample array. */
  data.maxFrameIndex = (unsigned long)data.totalBytes / data.blockAlign;
  data.audio = (unsigned char*)audio;

  printf("\n=== Now playing back. ===\n");
  fflush(stdout);
  err = Pa_OpenStream(&stream, NULL, /* no input */
                      &outputParameters, data.fs, framesPerBuffer,
                      paClipOff, /* we won't output out of range samples so
                                    don't bother clipping them */
                      playCallback, &data);
  if (err != paNoError) {
    Pa_Terminate();
    throwError(ERR_NO_STREAM, "Pa_OpenStream returned %d, %s", err,
               Pa_GetErrorText(err));
  }

  if (stream) {
#if defined(__linux__)
    PaAlsa_EnableRealtimeScheduling(stream, 1);
    printf("RealTime Scheduling enabled\n");
#endif
    err = Pa_StartStream(stream);
    if (err != paNoError) {
      Pa_Terminate();
      throwError(ERR_NO_STREAM, "Pa_StartStream returned %d, %s", err,
                 Pa_GetErrorText(err));
    }
    printf("Waiting for playback to start.\n");

    do {
      startTime = Pa_GetStreamTime(stream);
    } while (startTime == 0.0);

    printf("Waiting for playback to finish.\n");
    fflush(stdout);
    //finishTime = data.maxFrameIndex / data.fs;
    while ((err = Pa_IsStreamActive(stream)) == 1) {
      Pa_Sleep(1000);
    }
    printf("\n");

    if (err != paNoError) {
      Pa_Terminate();
      throwError(ERR_NO_STREAM, "Pa error %d received during playback: %s", err,
                 Pa_GetErrorText(err));
    }

    err = Pa_CloseStream(stream);
    if (err != paNoError)

      printf("Done.\n");
    fflush(stdout);
  }

  if (audio) {
    free(audio);
  }
  Pa_Terminate();
  exit(0);
}