audio_alsa_sink.cc

这是用python语言写的一个数字广播的信号处理工具包。利用它

/* -*- c++ -*- */
/*
 * Copyright 2004 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_sink.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_output_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_sink_sptr
audio_alsa_make_sink (int sampling_rate,
		      const std::string dev,
		      bool ok_to_block)
{
  return audio_alsa_sink_sptr (new audio_alsa_sink (sampling_rate, dev,
						    ok_to_block));
}

audio_alsa_sink::audio_alsa_sink (int sampling_rate,
				  const std::string device_name,
				  bool ok_to_block)
  : gr_sync_block ("audio_alsa_sink",
		   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_special_case_mono_to_stereo (false),
    d_nunderuns (0), d_nsuspends (0)
{
  CHATTY_DEBUG = gr_prefs::singleton()->get_bool("audio_alsa", "verbose", false);

  int  	error;
  int	dir;

  // open the device for playback
  error = snd_pcm_open(&d_pcm_handle, d_device_name.c_str (),
		       SND_PCM_STREAM_PLAYBACK, 0);
  if (error < 0){
    fprintf (stderr, "audio_alsa_sink[%s]: %s\n",
	     d_device_name.c_str(), snd_strerror(error));
    throw std::runtime_error ("audio_alsa_sink");
  }

  // 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 input signature
  unsigned int umin_chan, umax_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 input and handle the duplication ourselves.

  if (min_chan == 2){
    min_chan = 1;
    d_special_case_mono_to_stereo = true;
  }
  set_input_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_alloca (&access_mask);
  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_sink",
					CHATTY_DEBUG))
    throw std::runtime_error ("audio_alsa_sink");
  

  // 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_sink[%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_sink: 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_sink::check_topology (int ninputs, int noutputs)
{
  // ninputs is how many channels the user has connected.
  // Now we can finish up setting up the hw params...

  int nchan = ninputs;
  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_mono_to_stereo;
  if (special_case)
    nchan = 2;
  
  err = snd_pcm_hw_params_set_channels (d_pcm_handle, d_hw_params, 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;
  }

  // get current s/w params
  err = snd_pcm_sw_params_current (d_pcm_handle, d_sw_params);
  if (err < 0)
    bail ("snd_pcm_sw_params_current", err);
  
  // Tell the PCM device to wait to start until we've filled
  // it's buffers half way full.  This helps avoid audio underruns.

  err = snd_pcm_sw_params_set_start_threshold(d_pcm_handle,
					      d_sw_params,
					      d_nperiods * d_period_size / 2);
  if (err < 0)
    bail ("snd_pcm_sw_params_set_start_threshold", err);

  // store the s/w params
  err = snd_pcm_sw_params (d_pcm_handle, d_sw_params);
  if (err < 0)
    bail ("snd_pcm_sw_params", err);

  d_buffer_size_bytes =
    d_period_size * nchan * snd_pcm_format_size (d_format, 1);

  d_buffer = new char [d_buffer_size_bytes];

  if (CHATTY_DEBUG)
    fprintf (stdout, "audio_alsa_sink[%s]: sample resolution = %d bits\n",
	     snd_pcm_name (d_pcm_handle),
	     snd_pcm_hw_params_get_sbits (d_hw_params));