audio: moved alsa support files into subdirectory

1 files changed, 505 insertions, 0 deletions

diff --git a/gr-audio/lib/alsa/audio_alsa_source.cc b/gr-audio/lib/alsa/audio_alsa_source.cc
new file mode 100644
index 000000000..a8667361e
--- /dev/null
+++ b/gr-audio/lib/alsa/audio_alsa_source.cc
@@ -0,0 +1,505 @@
+/* -*- c++ -*- */
+/*
+ * Copyright 2004-2011 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 "gr_audio_registry.h"
+#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>
+
+AUDIO_REGISTER_SOURCE(alsa)(
+    int sampling_rate, const std::string &device_name, bool ok_to_block
+){
+    return audio_source::sptr(new audio_alsa_source(sampling_rate, device_name, ok_to_block));
+}
+
+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::audio_alsa_source (int sampling_rate,
+				      const std::string device_name,
+				      bool ok_to_block)
+  : audio_source ("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");
+}