1 files changed, 514 insertions, 0 deletions
diff --git a/gr-audio/lib/alsa/alsa_source.cc b/gr-audio/lib/alsa/alsa_source.cc
new file mode 100644
index 0000000000..838da60af3
--- /dev/null
+++ b/gr-audio/lib/alsa/alsa_source.cc
@@ -0,0 +1,514 @@
+/* -*- c++ -*- */
+/*
+ * Copyright 2004-2011,2013 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_registry.h"
+#include <alsa_source.h>
+#include <alsa_impl.h>
+#include <gr_io_signature.h>
+#include <gr_prefs.h>
+#include <stdio.h>
+#include <iostream>
+#include <stdexcept>
+
+namespace gr {
+  namespace audio {
+
+    AUDIO_REGISTER_SOURCE(REG_PRIO_HIGH, alsa)(int sampling_rate,
+                                               const std::string &device_name,
+                                               bool ok_to_block)
+    {
+      return source::sptr
+        (new 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));
+    }
+
+    // ----------------------------------------------------------------
+
+    alsa_source::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
+    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;
+
+      // Check the state of the stream
+      // Ensure that the pcm is in a state where we can still mess with the hw_params
+      snd_pcm_state_t state;
+      state=snd_pcm_state(d_pcm_handle);
+      if(state== SND_PCM_STATE_RUNNING)
+        return true;  // If stream is running, don't change any parameters
+      else if(state == SND_PCM_STATE_XRUN)
+        snd_pcm_prepare(d_pcm_handle); // Prepare stream on underrun, and we can set parameters;
+
+      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 = &alsa_source::work_s16_2x1;
+        else
+          d_worker = &alsa_source::work_s16;
+        break;
+
+      case SND_PCM_FORMAT_S32:
+        if(special_case)
+          d_worker = &alsa_source::work_s32_2x1;
+        else
+          d_worker = &alsa_source::work_s32;
+        break;
+
+      default:
+        assert(0);
+      }
+
+      return true;
+    }
+
+    alsa_source::~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
+    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
+    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 float scale_factor = 1.0 / std::pow(2.0f, 16-1);
+
+      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++] * scale_factor;
+        }
+      }
+
+      return d_period_size;
+    }
+
+    /*
+     * Work function that deals with float to S16 conversion
+     * and stereo to mono kludge...
+     */
+    int
+    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 float scale_factor = 1.0 / std::pow(2.0f, 16-1);
+
+      float **out = (float**)&output_items[0];
+      sample_t *buf = (sample_t*)d_buffer;
+      int bi;
+
+      assert(output_items.size () == 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 * scale_factor;
+      }
+
+      return d_period_size;
+    }
+
+    /*
+     * Work function that deals with float to S32 conversion
+     */
+    int
+    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 float scale_factor = 1.0 / std::pow(2.0f, 32-1);
+
+      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++] * scale_factor;
+        }
+      }
+
+      return d_period_size;
+    }
+
+    /*
+     * Work function that deals with float to S32 conversion
+     * and stereo to mono kludge...
+     */
+    int
+    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 float scale_factor = 1.0 / std::pow(2.0f, 32-1);
+
+      float **out = (float**)&output_items[0];
+      sample_t *buf = (sample_t*)d_buffer;
+      int bi;
+
+      assert(output_items.size () == 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 * scale_factor;
+      }
+
+      return d_period_size;
+    }
+
+    bool
+    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
+    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
+    alsa_source::bail(const char *msg, int err) throw (std::runtime_error)
+    {
+      output_error_msg(msg, err);
+      throw std::runtime_error("audio_alsa_source");
+    }
+
+  } /* namespace audio */
+} /* namespace gr */