/* -*- c++ -*- */

/*

 * Copyright 2004,2006 Free Software Foundation, Inc.

 * 

 * This file is part of GNU Radio

 * 

 * GNU Radio 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, or (at your option)

 * any later version.

 * 

 * GNU Radio 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 GNU Radio; see the file COPYING.  If not, write to

 * the Free Software Foundation, Inc., 51 Franklin Street,

 * Boston, MA 02110-1301, USA.

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include <audio_alsa_source.h>

#include <gr_io_signature.h>

#include <gr_prefs.h>

#include <stdio.h>

#include <iostream>

#include <stdexcept>

#include <gri_alsa.h>

static bool CHATTY_DEBUG = false;

static snd_pcm_format_t acceptable_formats[] = {

  // these are in our preferred order...

  SND_PCM_FORMAT_S32,

  SND_PCM_FORMAT_S16

};

#define NELEMS(x) (sizeof(x)/sizeof(x[0]))

static std::string 

default_device_name ()

{

  return gr_prefs::singleton()->get_string("audio_alsa", "default_input_device", "hw:0,0");

}

static double

default_period_time ()

{

  return std::max(0.001, gr_prefs::singleton()->get_double("audio_alsa", "period_time", 0.010));

}

static int

default_nperiods ()

{

  return std::max(2L, gr_prefs::singleton()->get_long("audio_alsa", "nperiods", 4));

}

// ----------------------------------------------------------------

audio_alsa_source_sptr

audio_alsa_make_source (int sampling_rate, const std::string dev,

                        bool ok_to_block)

{

  return audio_alsa_source_sptr (new audio_alsa_source (sampling_rate, dev,

                                                        ok_to_block));

}

audio_alsa_source::audio_alsa_source (int sampling_rate,

                                      const std::string device_name,

                                      bool ok_to_block)

  : gr_sync_block ("audio_alsa_source",

                   gr_make_io_signature (0, 0, 0),

                   gr_make_io_signature (0, 0, 0)),

    d_sampling_rate (sampling_rate),

    d_device_name (device_name.empty() ? default_device_name() : device_name),

    d_pcm_handle (0),

    d_hw_params ((snd_pcm_hw_params_t *)(new char[snd_pcm_hw_params_sizeof()])),

    d_sw_params ((snd_pcm_sw_params_t *)(new char[snd_pcm_sw_params_sizeof()])),

    d_nperiods (default_nperiods()),

    d_period_time_us ((unsigned int) (default_period_time() * 1e6)),

    d_period_size (0),

    d_buffer_size_bytes (0), d_buffer (0),

    d_worker (0), d_hw_nchan (0),

    d_special_case_stereo_to_mono (false),

    d_noverruns (0), d_nsuspends (0)

{

  CHATTY_DEBUG = gr_prefs::singleton()->get_bool("audio_alsa", "verbose", false);

  int          error;

  int        dir;

  // open the device for capture

  error = snd_pcm_open(&d_pcm_handle, d_device_name.c_str (),

                       SND_PCM_STREAM_CAPTURE, 0);

  if (error < 0){

    fprintf (stderr, "audio_alsa_source[%s]: %s\n",

             d_device_name.c_str(), snd_strerror(error));

    throw std::runtime_error ("audio_alsa_source");

  }

  // Fill params with a full configuration space for a PCM.

  error = snd_pcm_hw_params_any(d_pcm_handle, d_hw_params);

  if (error < 0)

    bail ("broken configuration for playback", error);

  if (CHATTY_DEBUG)

    gri_alsa_dump_hw_params (d_pcm_handle, d_hw_params, stdout);

  // now that we know how many channels the h/w can handle, set output signature

  unsigned int umax_chan;

  unsigned int umin_chan;

  snd_pcm_hw_params_get_channels_min (d_hw_params, &umin_chan);

  snd_pcm_hw_params_get_channels_max (d_hw_params, &umax_chan);

  int min_chan = std::min (umin_chan, 1000U);

  int max_chan = std::min (umax_chan, 1000U);

  // As a special case, if the hw's min_chan is two, we'll accept

  // a single output and handle the demux ourselves.

  if (min_chan == 2){

    min_chan = 1;

    d_special_case_stereo_to_mono = true;

  }

  set_output_signature (gr_make_io_signature (min_chan, max_chan,

                                              sizeof (float)));

  // fill in portions of the d_hw_params that we know now...

  // Specify the access methods we implement

  // For now, we only handle RW_INTERLEAVED...

  snd_pcm_access_mask_t *access_mask;

  snd_pcm_access_mask_t **access_mask_ptr = &access_mask; // FIXME: workaround for compiler warning

  snd_pcm_access_mask_alloca (access_mask_ptr);

  snd_pcm_access_mask_none (access_mask);

  snd_pcm_access_mask_set (access_mask, SND_PCM_ACCESS_RW_INTERLEAVED);

  // snd_pcm_access_mask_set (access_mask, SND_PCM_ACCESS_RW_NONINTERLEAVED);

  if ((error = snd_pcm_hw_params_set_access_mask (d_pcm_handle,

                                                  d_hw_params, access_mask)) < 0)

    bail ("failed to set access mask", error);

  // set sample format

  if (!gri_alsa_pick_acceptable_format (d_pcm_handle, d_hw_params,

                                        acceptable_formats,

                                        NELEMS (acceptable_formats),

                                        &d_format,

                                        "audio_alsa_source",

                                        CHATTY_DEBUG))

    throw std::runtime_error ("audio_alsa_source");

  // sampling rate

  unsigned int orig_sampling_rate = d_sampling_rate;

  if ((error = snd_pcm_hw_params_set_rate_near (d_pcm_handle, d_hw_params,

                                                &d_sampling_rate, 0)) < 0)

    bail ("failed to set rate near", error);

  if (orig_sampling_rate != d_sampling_rate){

    fprintf (stderr, "audio_alsa_source[%s]: unable to support sampling rate %d\n",

             snd_pcm_name (d_pcm_handle), orig_sampling_rate);

    fprintf (stderr, "  card requested %d instead.\n", d_sampling_rate);

  }

  /*

   * ALSA transfers data in units of "periods".

   * We indirectly determine the underlying buffersize by specifying

   * the number of periods we want (typically 4) and the length of each

   * period in units of time (typically 1ms).

   */

  unsigned int min_nperiods, max_nperiods;

  snd_pcm_hw_params_get_periods_min (d_hw_params, &min_nperiods, &dir);

  snd_pcm_hw_params_get_periods_max (d_hw_params, &max_nperiods, &dir);

  //fprintf (stderr, "alsa_source: min_nperiods = %d, max_nperiods = %d\n",

  // min_nperiods, max_nperiods);

  unsigned int orig_nperiods = d_nperiods;

  d_nperiods = std::min (std::max (min_nperiods, d_nperiods), max_nperiods);

  // adjust period time so that total buffering remains more-or-less constant

  d_period_time_us = (d_period_time_us * orig_nperiods) / d_nperiods;

  error = snd_pcm_hw_params_set_periods (d_pcm_handle, d_hw_params,

                                         d_nperiods, 0);

  if (error < 0)

    bail ("set_periods failed", error);

  dir = 0;

  error = snd_pcm_hw_params_set_period_time_near (d_pcm_handle, d_hw_params,

                                                  &d_period_time_us, &dir);

  if (error < 0)

    bail ("set_period_time_near failed", error);

  dir = 0;

  error = snd_pcm_hw_params_get_period_size (d_hw_params,

                                             &d_period_size, &dir);

  if (error < 0)

    bail ("get_period_size failed", error);

  set_output_multiple (d_period_size);

}

bool

audio_alsa_source::check_topology (int ninputs, int noutputs)

{

  // noutputs is how many channels the user has connected.

  // Now we can finish up setting up the hw params...

  unsigned int nchan = noutputs;

  int err;

  // FIXME check_topology may be called more than once.

  // Ensure that the pcm is in a state where we can still mess with the hw_params

  bool special_case = nchan == 1 && d_special_case_stereo_to_mono;

  if (special_case)

    nchan = 2;

  d_hw_nchan = nchan;

  err = snd_pcm_hw_params_set_channels (d_pcm_handle, d_hw_params, d_hw_nchan);

  if (err < 0){

    output_error_msg ("set_channels failed", err);

    return false;

  }

  // set the parameters into the driver...

  err = snd_pcm_hw_params(d_pcm_handle, d_hw_params);

  if (err < 0){

    output_error_msg ("snd_pcm_hw_params failed", err);

    return false;

  }

  d_buffer_size_bytes =

    d_period_size * d_hw_nchan * snd_pcm_format_size (d_format, 1);

  d_buffer = new char [d_buffer_size_bytes];

  if (CHATTY_DEBUG)

    fprintf (stdout, "audio_alsa_source[%s]: sample resolution = %d bits\n",

             snd_pcm_name (d_pcm_handle),

             snd_pcm_hw_params_get_sbits (d_hw_params));

  switch (d_format){

  case SND_PCM_FORMAT_S16:

    if (special_case)

      d_worker = &audio_alsa_source::work_s16_2x1;

    else

      d_worker = &audio_alsa_source::work_s16;

    break;

  case SND_PCM_FORMAT_S32:

    if (special_case)

      d_worker = &audio_alsa_source::work_s32_2x1;

    else

      d_worker = &audio_alsa_source::work_s32;

    break;

  default:

    assert (0);

  }

  return true;

}

audio_alsa_source::~audio_alsa_source ()

{

  if (snd_pcm_state (d_pcm_handle) == SND_PCM_STATE_RUNNING)

    snd_pcm_drop (d_pcm_handle);

  snd_pcm_close(d_pcm_handle);

  delete [] ((char *) d_hw_params);

  delete [] ((char *) d_sw_params);

  delete [] d_buffer;

}

int

audio_alsa_source::work (int noutput_items,

                         gr_vector_const_void_star &input_items,

                         gr_vector_void_star &output_items)

{

  assert ((noutput_items % d_period_size) == 0);

  assert (noutput_items != 0);

  // this is a call through a pointer to a method...

  return (this->*d_worker)(noutput_items, input_items, output_items);

}

/*

 * Work function that deals with float to S16 conversion

 */

int

audio_alsa_source::work_s16 (int noutput_items,

                             gr_vector_const_void_star &input_items,

                             gr_vector_void_star &output_items)

{

  typedef gr_int16        sample_t;        // the type of samples we're creating

  static const int NBITS = 16;                // # of bits in a sample

  unsigned int nchan = output_items.size ();

  float **out = (float **) &output_items[0];

  sample_t *buf = (sample_t *) d_buffer;

  int bi;

  unsigned int sizeof_frame = d_hw_nchan * sizeof (sample_t);

  assert (d_buffer_size_bytes == d_period_size * sizeof_frame);

  // To minimize latency, return at most a single period's worth of samples.

  // [We could also read the first one in a blocking mode and subsequent

  //  ones in non-blocking mode, but we'll leave that for later (or never).]

  if (!read_buffer (buf, d_period_size, sizeof_frame))

    return -1;                // No fixing this problem.  Say we're done.

  // process one period of data

  bi = 0;

  for (unsigned int i = 0; i < d_period_size; i++){

    for (unsigned int chan = 0; chan < nchan; chan++){

      out[chan][i] = (float) buf[bi++] * (1.0 / (float) ((1L << (NBITS-1)) - 1));

    }

  }

  return d_period_size;

}

/*

 * Work function that deals with float to S16 conversion

 * and stereo to mono kludge...

 */

int

audio_alsa_source::work_s16_2x1 (int noutput_items,

                                 gr_vector_const_void_star &input_items,

                                 gr_vector_void_star &output_items)

{

  typedef gr_int16        sample_t;        // the type of samples we're creating

  static const int NBITS = 16;                // # of bits in a sample

  unsigned int nchan = output_items.size ();

  float **out = (float **) &output_items[0];

  sample_t *buf = (sample_t *) d_buffer;

  int bi;

  assert (nchan == 1);

  unsigned int sizeof_frame = d_hw_nchan * sizeof (sample_t);

  assert (d_buffer_size_bytes == d_period_size * sizeof_frame);

  // To minimize latency, return at most a single period's worth of samples.

  // [We could also read the first one in a blocking mode and subsequent

  //  ones in non-blocking mode, but we'll leave that for later (or never).]

  if (!read_buffer (buf, d_period_size, sizeof_frame))

    return -1;                // No fixing this problem.  Say we're done.

  // process one period of data

  bi = 0;

  for (unsigned int i = 0; i < d_period_size; i++){

    int t = (buf[bi] + buf[bi+1]) / 2;

    bi += 2;

    out[0][i] = (float) t * (1.0 / (float) ((1L << (NBITS-1)) - 1));

  }

  return d_period_size;

}

/*

 * Work function that deals with float to S32 conversion

 */

int

audio_alsa_source::work_s32 (int noutput_items,

                             gr_vector_const_void_star &input_items,

                             gr_vector_void_star &output_items)

{

  typedef gr_int32        sample_t;        // the type of samples we're creating

  static const int NBITS = 32;                // # of bits in a sample

  unsigned int nchan = output_items.size ();

  float **out = (float **) &output_items[0];

  sample_t *buf = (sample_t *) d_buffer;

  int bi;

  unsigned int sizeof_frame = d_hw_nchan * sizeof (sample_t);

  assert (d_buffer_size_bytes == d_period_size * sizeof_frame);

  // To minimize latency, return at most a single period's worth of samples.

  // [We could also read the first one in a blocking mode and subsequent

  //  ones in non-blocking mode, but we'll leave that for later (or never).]

  if (!read_buffer (buf, d_period_size, sizeof_frame))

    return -1;                // No fixing this problem.  Say we're done.

  // process one period of data

  bi = 0;

  for (unsigned int i = 0; i < d_period_size; i++){

    for (unsigned int chan = 0; chan < nchan; chan++){

      out[chan][i] = (float) buf[bi++] * (1.0 / (float) ((1L << (NBITS-1)) - 1));

    }

  }

  return d_period_size;

}

/*

 * Work function that deals with float to S32 conversion

 * and stereo to mono kludge...

 */

int

audio_alsa_source::work_s32_2x1 (int noutput_items,

                                 gr_vector_const_void_star &input_items,

                                 gr_vector_void_star &output_items)

{

  typedef gr_int32        sample_t;        // the type of samples we're creating

  static const int NBITS = 32;                // # of bits in a sample

  unsigned int nchan = output_items.size ();

  float **out = (float **) &output_items[0];

  sample_t *buf = (sample_t *) d_buffer;

  int bi;

  assert (nchan == 1);

  unsigned int sizeof_frame = d_hw_nchan * sizeof (sample_t);

  assert (d_buffer_size_bytes == d_period_size * sizeof_frame);

  // To minimize latency, return at most a single period's worth of samples.

  // [We could also read the first one in a blocking mode and subsequent

  //  ones in non-blocking mode, but we'll leave that for later (or never).]

  if (!read_buffer (buf, d_period_size, sizeof_frame))

    return -1;                // No fixing this problem.  Say we're done.

  // process one period of data

  bi = 0;

  for (unsigned int i = 0; i < d_period_size; i++){

    int t = (buf[bi] + buf[bi+1]) / 2;

    bi += 2;

    out[0][i] = (float) t * (1.0 / (float) ((1L << (NBITS-1)) - 1));

  }

  return d_period_size;

}

bool

audio_alsa_source::read_buffer (void *vbuffer, unsigned nframes, unsigned sizeof_frame)

{

  unsigned char *buffer = (unsigned char *) vbuffer;

  while (nframes > 0){

    int r = snd_pcm_readi (d_pcm_handle, buffer, nframes);

    if (r == -EAGAIN)

      continue;                        // try again

    else if (r == -EPIPE){        // overrun

      d_noverruns++;

      fputs ("aO", stderr);

      if ((r = snd_pcm_prepare (d_pcm_handle)) < 0){

        output_error_msg ("snd_pcm_prepare failed. Can't recover from overrun", r);

        return false;

      }

      continue;                        // try again

    }

    else if (r == -ESTRPIPE){        // h/w is suspended (whatever that means)

                                // This is apparently related to power management

      d_nsuspends++;

      if ((r = snd_pcm_resume (d_pcm_handle)) < 0){

        output_error_msg ("failed to resume from suspend", r);

        return false;

      }

      continue;                        // try again

    }

    else if (r < 0){

      output_error_msg ("snd_pcm_readi failed", r);

      return false;

    }

    nframes -= r;

    buffer += r * sizeof_frame;

  }

  return true;

}

void

audio_alsa_source::output_error_msg (const char *msg, int err)

{

  fprintf (stderr, "audio_alsa_source[%s]: %s: %s\n",

           snd_pcm_name (d_pcm_handle), msg,  snd_strerror (err));

}

void

audio_alsa_source::bail (const char *msg, int err) throw (std::runtime_error)

{

  output_error_msg (msg, err);

  throw std::runtime_error ("audio_alsa_source");

}