From 24d30ff306a906b21ad724c07a95caed24659887 Mon Sep 17 00:00:00 2001
From: Josh Blum <josh@joshknows.com>
Date: Wed, 9 Mar 2011 10:29:53 -0800
Subject: audio: work to get alsa support in, lets see if this works before
adding more
---
gr-audio/lib/Makefile.am | 26 ++
gr-audio/lib/audio_alsa_sink.cc | 548 ++++++++++++++++++++++++++++++++++++++
gr-audio/lib/audio_alsa_sink.h | 104 ++++++++
gr-audio/lib/audio_alsa_source.cc | 505 +++++++++++++++++++++++++++++++++++
gr-audio/lib/audio_alsa_source.h | 106 ++++++++
gr-audio/lib/gr-audio-alsa.conf | 11 +
gr-audio/lib/gr-audio.conf | 3 +
gr-audio/lib/gr_audio_registry.cc | 56 +++-
gr-audio/lib/gr_audio_registry.h | 28 +-
gr-audio/lib/gri_alsa.cc | 175 ++++++++++++
gr-audio/lib/gri_alsa.h | 44 +++
11 files changed, 1589 insertions(+), 17 deletions(-)
create mode 100644 gr-audio/lib/audio_alsa_sink.cc
create mode 100644 gr-audio/lib/audio_alsa_sink.h
create mode 100644 gr-audio/lib/audio_alsa_source.cc
create mode 100644 gr-audio/lib/audio_alsa_source.h
create mode 100644 gr-audio/lib/gr-audio-alsa.conf
create mode 100644 gr-audio/lib/gr-audio.conf
create mode 100644 gr-audio/lib/gri_alsa.cc
create mode 100644 gr-audio/lib/gri_alsa.h
(limited to 'gr-audio/lib')
diff --git a/gr-audio/lib/Makefile.am b/gr-audio/lib/Makefile.am
index 8e5915ba29..7d9e419a8b 100644
--- a/gr-audio/lib/Makefile.am
+++ b/gr-audio/lib/Makefile.am
@@ -38,3 +38,29 @@ libgnuradio_audio_la_LIBADD = \
libgnuradio_audio_la_LDFLAGS = $(LTVERSIONFLAGS)
noinst_HEADERS = gr_audio_registry.h
+
+etcdir = $(gr_prefsdir)
+dist_etc_DATA = gr-audio.conf
+
+########################################################################
+## ALSA Support
+########################################################################
+if GR_AUDIO_ALSA_SUPPORT
+
+AM_CPPFLAGS += $(ALSA_CPPFLAGS)
+
+libgnuradio_audio_la_LIBADD += $(ALSA_LIBS)
+
+libgnuradio_audio_la_SOURCES += \
+ gri_alsa.cc \
+ audio_alsa_source.cc \
+ audio_alsa_sink.cc
+
+noinst_HEADERS += \
+ gri_alsa.h \
+ audio_alsa_source.h \
+ audio_alsa_sink.h
+
+dist_etc_DATA += gr-audio-alsa.conf
+
+endif
diff --git a/gr-audio/lib/audio_alsa_sink.cc b/gr-audio/lib/audio_alsa_sink.cc
new file mode 100644
index 0000000000..e02dfc31f9
--- /dev/null
+++ b/gr-audio/lib/audio_alsa_sink.cc
@@ -0,0 +1,548 @@
+/* -*- 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_sink.h>
+#include <gr_io_signature.h>
+#include <gr_prefs.h>
+#include <stdio.h>
+#include <iostream>
+#include <stdexcept>
+#include <gri_alsa.h>
+
+GR_AUDIO_REGISTER_SINK(alsa)(
+ int sampling_rate, const std::string &device_name, bool ok_to_block
+){
+ return gr_audio_sink::sptr(new audio_alsa_sink(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_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::audio_alsa_sink (int sampling_rate,
+ const std::string device_name,
+ bool ok_to_block)
+ : gr_audio_sink ("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), d_ok_to_block(ok_to_block)
+{
+ 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 (ok_to_block == false)
+ snd_pcm_nonblock(d_pcm_handle, !ok_to_block);
+ 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_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_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;
+
+ // 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_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));
+
+ switch (d_format){
+ case SND_PCM_FORMAT_S16:
+ if (special_case)
+ d_worker = &audio_alsa_sink::work_s16_1x2;
+ else
+ d_worker = &audio_alsa_sink::work_s16;
+ break;
+
+ case SND_PCM_FORMAT_S32:
+ if (special_case)
+ d_worker = &audio_alsa_sink::work_s32_1x2;
+ else
+ d_worker = &audio_alsa_sink::work_s32;
+ break;
+
+ default:
+ assert (0);
+ }
+ return true;
+}
+
+audio_alsa_sink::~audio_alsa_sink ()
+{
+ 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_sink::work (int noutput_items,
+ gr_vector_const_void_star &input_items,
+ gr_vector_void_star &output_items)
+{
+ assert ((noutput_items % d_period_size) == 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_sink::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 = input_items.size ();
+ const float **in = (const float **) &input_items[0];
+ sample_t *buf = (sample_t *) d_buffer;
+ int bi;
+ int n;
+
+ unsigned int sizeof_frame = nchan * sizeof (sample_t);
+ assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
+
+ for (n = 0; n < noutput_items; n += d_period_size){
+
+ // 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++){
+ buf[bi++] = (sample_t) (in[chan][i] * (float) ((1L << (NBITS-1)) - 1));
+ }
+ }
+
+ // update src pointers
+ for (unsigned int chan = 0; chan < nchan; chan++)
+ in[chan] += d_period_size;
+
+ if (!write_buffer (buf, d_period_size, sizeof_frame))
+ return -1; // No fixing this problem. Say we're done.
+ }
+
+ return n;
+}
+
+
+/*
+ * Work function that deals with float to S32 conversion
+ */
+int
+audio_alsa_sink::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 = input_items.size ();
+ const float **in = (const float **) &input_items[0];
+ sample_t *buf = (sample_t *) d_buffer;
+ int bi;
+ int n;
+
+ unsigned int sizeof_frame = nchan * sizeof (sample_t);
+ assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
+
+ for (n = 0; n < noutput_items; n += d_period_size){
+
+ // 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++){
+ buf[bi++] = (sample_t) (in[chan][i] * (float) ((1L << (NBITS-1)) - 1));
+ }
+ }
+
+ // update src pointers
+ for (unsigned int chan = 0; chan < nchan; chan++)
+ in[chan] += d_period_size;
+
+ if (!write_buffer (buf, d_period_size, sizeof_frame))
+ return -1; // No fixing this problem. Say we're done.
+ }
+
+ return n;
+}
+
+/*
+ * Work function that deals with float to S16 conversion and
+ * mono to stereo kludge.
+ */
+int
+audio_alsa_sink::work_s16_1x2 (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
+
+ assert (input_items.size () == 1);
+ static const unsigned int nchan = 2;
+ const float **in = (const float **) &input_items[0];
+ sample_t *buf = (sample_t *) d_buffer;
+ int bi;
+ int n;
+
+ unsigned int sizeof_frame = nchan * sizeof (sample_t);
+ assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
+
+ for (n = 0; n < noutput_items; n += d_period_size){
+
+ // process one period of data
+ bi = 0;
+ for (unsigned int i = 0; i < d_period_size; i++){
+ sample_t t = (sample_t) (in[0][i] * (float) ((1L << (NBITS-1)) - 1));
+ buf[bi++] = t;
+ buf[bi++] = t;
+ }
+
+ // update src pointers
+ in[0] += d_period_size;
+
+ if (!write_buffer (buf, d_period_size, sizeof_frame))
+ return -1; // No fixing this problem. Say we're done.
+ }
+
+ return n;
+}
+
+/*
+ * Work function that deals with float to S32 conversion and
+ * mono to stereo kludge.
+ */
+int
+audio_alsa_sink::work_s32_1x2 (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
+
+ assert (input_items.size () == 1);
+ static unsigned int nchan = 2;
+ const float **in = (const float **) &input_items[0];
+ sample_t *buf = (sample_t *) d_buffer;
+ int bi;
+ int n;
+
+ unsigned int sizeof_frame = nchan * sizeof (sample_t);
+ assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
+
+ for (n = 0; n < noutput_items; n += d_period_size){
+
+ // process one period of data
+ bi = 0;
+ for (unsigned int i = 0; i < d_period_size; i++){
+ sample_t t = (sample_t) (in[0][i] * (float) ((1L << (NBITS-1)) - 1));
+ buf[bi++] = t;
+ buf[bi++] = t;
+ }
+
+ // update src pointers
+ in[0] += d_period_size;
+
+ if (!write_buffer (buf, d_period_size, sizeof_frame))
+ return -1; // No fixing this problem. Say we're done.
+ }
+
+ return n;
+}
+
+bool
+audio_alsa_sink::write_buffer (const void *vbuffer,
+ unsigned nframes, unsigned sizeof_frame)
+{
+ const unsigned char *buffer = (const unsigned char *) vbuffer;
+
+ while (nframes > 0){
+ int r = snd_pcm_writei (d_pcm_handle, buffer, nframes);
+ if (r == -EAGAIN)
+ {
+ if (d_ok_to_block == true)
+ continue; // try again
+
+ break;
+ }
+
+ else if (r == -EPIPE){ // underrun
+ d_nunderuns++;
+ fputs ("aU", stderr);
+ if ((r = snd_pcm_prepare (d_pcm_handle)) < 0){
+ output_error_msg ("snd_pcm_prepare failed. Can't recover from underrun", 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_writei failed", r);
+ return false;
+ }
+
+ nframes -= r;
+ buffer += r * sizeof_frame;
+ }
+
+ return true;
+}
+
+
+void
+audio_alsa_sink::output_error_msg (const char *msg, int err)
+{
+ fprintf (stderr, "audio_alsa_sink[%s]: %s: %s\n",
+ snd_pcm_name (d_pcm_handle), msg, snd_strerror (err));
+}
+
+void
+audio_alsa_sink::bail (const char *msg, int err) throw (std::runtime_error)
+{
+ output_error_msg (msg, err);
+ throw std::runtime_error ("audio_alsa_sink");
+}
diff --git a/gr-audio/lib/audio_alsa_sink.h b/gr-audio/lib/audio_alsa_sink.h
new file mode 100644
index 0000000000..aa3e88058f
--- /dev/null
+++ b/gr-audio/lib/audio_alsa_sink.h
@@ -0,0 +1,104 @@
+/* -*- 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.
+ */
+
+#ifndef INCLUDED_AUDIO_ALSA_SINK_H
+#define INCLUDED_AUDIO_ALSA_SINK_H
+
+// use new ALSA API
+#define ALSA_PCM_NEW_HW_PARAMS_API
+#define ALSA_PCM_NEW_SW_PARAMS_API
+
+#include <gr_audio_sink.h>
+#include <string>
+#include <alsa/asoundlib.h>
+#include <stdexcept>
+
+/*!
+ * \brief audio sink using ALSA
+ *
+ * The sink has N input streams of floats, where N depends
+ * on the hardware characteristics of the selected device.
+ *
+ * Input samples must be in the range [-1,1].
+ */
+class audio_alsa_sink : public gr_audio_sink {
+ // typedef for pointer to class work method
+ typedef int (audio_alsa_sink::*work_t)(int noutput_items,
+ gr_vector_const_void_star &input_items,
+ gr_vector_void_star &output_items);
+
+ unsigned int d_sampling_rate;
+ std::string d_device_name;
+ snd_pcm_t *d_pcm_handle;
+ snd_pcm_hw_params_t *d_hw_params;
+ snd_pcm_sw_params_t *d_sw_params;
+ snd_pcm_format_t d_format;
+ unsigned int d_nperiods;
+ unsigned int d_period_time_us; // microseconds
+ snd_pcm_uframes_t d_period_size; // in frames
+ unsigned int d_buffer_size_bytes; // sizeof of d_buffer
+ char *d_buffer;
+ work_t d_worker; // the work method to use
+ bool d_special_case_mono_to_stereo;
+
+ // random stats
+ int d_nunderuns; // count of underruns
+ int d_nsuspends; // count of suspends
+ bool d_ok_to_block; // defaults to "true", controls blocking/non-block I/O
+
+ void output_error_msg (const char *msg, int err);
+ void bail (const char *msg, int err) throw (std::runtime_error);
+
+public:
+ audio_alsa_sink (int sampling_rate, const std::string device_name,
+ bool ok_to_block);
+
+ ~audio_alsa_sink ();
+
+ bool check_topology (int ninputs, int noutputs);
+
+ int work (int noutput_items,
+ gr_vector_const_void_star &input_items,
+ gr_vector_void_star &output_items);
+
+
+protected:
+ bool write_buffer (const void *buffer, unsigned nframes, unsigned sizeof_frame);
+
+ int work_s16 (int noutput_items,
+ gr_vector_const_void_star &input_items,
+ gr_vector_void_star &output_items);
+
+ int work_s16_1x2 (int noutput_items,
+ gr_vector_const_void_star &input_items,
+ gr_vector_void_star &output_items);
+
+ int work_s32 (int noutput_items,
+ gr_vector_const_void_star &input_items,
+ gr_vector_void_star &output_items);
+
+ int work_s32_1x2 (int noutput_items,
+ gr_vector_const_void_star &input_items,
+ gr_vector_void_star &output_items);
+};
+
+#endif /* INCLUDED_AUDIO_ALSA_SINK_H */
diff --git a/gr-audio/lib/audio_alsa_source.cc b/gr-audio/lib/audio_alsa_source.cc
new file mode 100644
index 0000000000..33c2c39327
--- /dev/null
+++ b/gr-audio/lib/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>
+
+GR_AUDIO_REGISTER_SOURCE(alsa)(
+ int sampling_rate, const std::string &device_name, bool ok_to_block
+){
+ return gr_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)
+ : gr_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");
+}
diff --git a/gr-audio/lib/audio_alsa_source.h b/gr-audio/lib/audio_alsa_source.h
new file mode 100644
index 0000000000..8ddc00bd2f
--- /dev/null
+++ b/gr-audio/lib/audio_alsa_source.h
@@ -0,0 +1,106 @@
+/* -*- 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.
+ */
+
+#ifndef INCLUDED_AUDIO_ALSA_SOURCE_H
+#define INCLUDED_AUDIO_ALSA_SOURCE_H
+
+// use new ALSA API
+#define ALSA_PCM_NEW_HW_PARAMS_API
+#define ALSA_PCM_NEW_SW_PARAMS_API
+
+#include <gr_audio_source.h>
+#include <string>
+#include <alsa/asoundlib.h>
+#include <stdexcept>
+
+class audio_alsa_source;
+typedef boost::shared_ptr<audio_alsa_source> audio_alsa_source_sptr;
+
+/*!
+ * \brief audio source using ALSA
+ *
+ * The source has between 1 and N input streams of floats, where N is
+ * depends on the hardware characteristics of the selected device.
+ *
+ * Output samples will be in the range [-1,1].
+ */
+class audio_alsa_source : public gr_audio_source {
+ // typedef for pointer to class work method
+ typedef int (audio_alsa_source::*work_t)(int noutput_items,
+ gr_vector_const_void_star &input_items,
+ gr_vector_void_star &output_items);
+
+ unsigned int d_sampling_rate;
+ std::string d_device_name;
+ snd_pcm_t *d_pcm_handle;
+ snd_pcm_hw_params_t *d_hw_params;
+ snd_pcm_sw_params_t *d_sw_params;
+ snd_pcm_format_t d_format;
+ unsigned int d_nperiods;
+ unsigned int d_period_time_us; // microseconds
+ snd_pcm_uframes_t d_period_size; // in frames
+ unsigned int d_buffer_size_bytes; // sizeof of d_buffer
+ char *d_buffer;
+ work_t d_worker; // the work method to use
+ unsigned int d_hw_nchan; // # of configured h/w channels
+ bool d_special_case_stereo_to_mono;
+
+ // random stats
+ int d_noverruns; // count of overruns
+ int d_nsuspends; // count of suspends
+
+ void output_error_msg (const char *msg, int err);
+ void bail (const char *msg, int err) throw (std::runtime_error);
+
+public:
+ audio_alsa_source (int sampling_rate, const std::string device_name,
+ bool ok_to_block);
+
+ ~audio_alsa_source ();
+
+ bool check_topology (int ninputs, int noutputs);
+
+ int work (int noutput_items,
+ gr_vector_const_void_star &input_items,
+ gr_vector_void_star &output_items);
+
+protected:
+ bool read_buffer (void *buffer, unsigned nframes, unsigned sizeof_frame);
+
+ int work_s16 (int noutput_items,
+ gr_vector_const_void_star &input_items,
+ gr_vector_void_star &output_items);
+
+ int work_s16_2x1 (int noutput_items,
+ gr_vector_const_void_star &input_items,
+ gr_vector_void_star &output_items);
+
+ int work_s32 (int noutput_items,
+ gr_vector_const_void_star &input_items,
+ gr_vector_void_star &output_items);
+
+ int work_s32_2x1 (int noutput_items,
+ gr_vector_const_void_star &input_items,
+ gr_vector_void_star &output_items);
+};
+
+#endif /* INCLUDED_AUDIO_ALSA_SOURCE_H */
diff --git a/gr-audio/lib/gr-audio-alsa.conf b/gr-audio/lib/gr-audio-alsa.conf
new file mode 100644
index 0000000000..5cec63e7ab
--- /dev/null
+++ b/gr-audio/lib/gr-audio-alsa.conf
@@ -0,0 +1,11 @@
+# This file contains system wide configuration data for GNU Radio.
+# You may override any setting on a per-user basis by editing
+# ~/.gnuradio/config.conf
+
+[audio_alsa]
+
+default_input_device = hw:0,0
+default_output_device = hw:0,0
+period_time = 0.010 # in seconds
+nperiods = 4 # total buffering = period_time * nperiods
+verbose = false
diff --git a/gr-audio/lib/gr-audio.conf b/gr-audio/lib/gr-audio.conf
new file mode 100644
index 0000000000..a79413d1a1
--- /dev/null
+++ b/gr-audio/lib/gr-audio.conf
@@ -0,0 +1,3 @@
+# This file contains system wide configuration data for GNU Radio.
+# You may override any setting on a per-user basis by editing
+# ~/.gnuradio/config.conf
diff --git a/gr-audio/lib/gr_audio_registry.cc b/gr-audio/lib/gr_audio_registry.cc
index e40d1ff0f5..708601489a 100644
--- a/gr-audio/lib/gr_audio_registry.cc
+++ b/gr-audio/lib/gr_audio_registry.cc
@@ -22,26 +22,45 @@
#include "gr_audio_registry.h"
#include <stdexcept>
#include <vector>
+#include <utility>
-static std::vector<gr_audio_registry_entry> &get_registry(void){
- static std::vector<gr_audio_registry_entry> _registry;
+/***********************************************************************
+ * Create registries
+ **********************************************************************/
+static std::vector<std::pair<std::string, source_factory_t> > &get_source_registry(void){
+ static std::vector<std::pair<std::string, source_factory_t> > _registry;
return _registry;
}
-void gr_audio_register(const gr_audio_registry_entry &entry){
- get_registry().push_back(entry);
+static std::vector<std::pair<std::string, sink_factory_t> > &get_sink_registry(void){
+ static std::vector<std::pair<std::string, sink_factory_t> > _registry;
+ return _registry;
+}
+
+/***********************************************************************
+ * Register functions
+ **********************************************************************/
+void gr_audio_register_source(const std::string &name, source_factory_t source){
+ get_source_registry().push_back(std::make_pair(name, source));
+}
+
+void gr_audio_register_sink(const std::string &name, sink_factory_t sink){
+ get_sink_registry().push_back(std::make_pair(name, sink));
}
+/***********************************************************************
+ * Factory functions
+ **********************************************************************/
gr_audio_source::sptr gr_make_audio_source(
int sampling_rate,
const std::string device_name,
bool ok_to_block
){
- if (get_registry().empty()){
+ if (get_source_registry().empty()){
throw std::runtime_error("no available audio factories");
}
//TODO we may prefer to use a specific entry in the registry
- return get_registry().front().source(sampling_rate, device_name, ok_to_block);
+ return get_source_registry().front().second(sampling_rate, device_name, ok_to_block);
}
gr_audio_sink::sptr gr_make_audio_sink(
@@ -49,9 +68,30 @@ gr_audio_sink::sptr gr_make_audio_sink(
const std::string device_name,
bool ok_to_block
){
- if (get_registry().empty()){
+ if (get_sink_registry().empty()){
throw std::runtime_error("no available audio factories");
}
//TODO we may prefer to use a specific entry in the registry
- return get_registry().front().sink(sampling_rate, device_name, ok_to_block);
+ return get_sink_registry().front().second(sampling_rate, device_name, ok_to_block);
}
+
+/***********************************************************************
+ * Default constructors
+ **********************************************************************/
+#include <gr_io_signature.h>
+
+gr_audio_sink::gr_audio_sink(
+ const std::string &name,
+ gr_io_signature_sptr insig,
+ gr_io_signature_sptr outsig
+):
+ gr_sync_block(name, insig, outsig)
+{}
+
+gr_audio_source::gr_audio_source(
+ const std::string &name,
+ gr_io_signature_sptr insig,
+ gr_io_signature_sptr outsig
+):
+ gr_sync_block(name, insig, outsig)
+{}
diff --git a/gr-audio/lib/gr_audio_registry.h b/gr-audio/lib/gr_audio_registry.h
index 4b43a408b0..b846650174 100644
--- a/gr-audio/lib/gr_audio_registry.h
+++ b/gr-audio/lib/gr_audio_registry.h
@@ -26,14 +26,24 @@
#include <gr_audio_source.h>
#include <string>
-struct gr_audio_registry_entry{
- typedef gr_audio_source::sptr(*source_factory_t)(int, const std::string &, bool);
- typedef gr_audio_sink::sptr(*sink_factory_t)(int, const std::string &, bool);
- std::string name;
- source_factory_t source;
- sink_factory_t sink;
-};
-
-void gr_audio_register(const gr_audio_registry_entry &entry);
+typedef gr_audio_source::sptr(*source_factory_t)(int, const std::string &, bool);
+typedef gr_audio_sink::sptr(*sink_factory_t)(int, const std::string &, bool);
+
+void gr_audio_register_source(const std::string &name, source_factory_t source);
+void gr_audio_register_sink(const std::string &name, sink_factory_t sink);
+
+#define GR_AUDIO_REGISTER_FIXTURE(x) static struct x{x();}x;x::x()
+
+#define GR_AUDIO_REGISTER_SOURCE(name) \
+ static gr_audio_source::sptr name##_source_fcn(int, const std::string &, bool); \
+ GR_AUDIO_REGISTER_FIXTURE(name##_source_reg){ \
+ gr_audio_register_source(#name, &name##_source_fcn); \
+ } static gr_audio_source::sptr name##_source_fcn
+
+#define GR_AUDIO_REGISTER_SINK(name) \
+ static gr_audio_sink::sptr name##_sink_fcn(int, const std::string &, bool); \
+ GR_AUDIO_REGISTER_FIXTURE(name##_sink_reg){ \
+ gr_audio_register_sink(#name, &name##_sink_fcn); \
+ } static gr_audio_sink::sptr name##_sink_fcn
#endif /* INCLUDED_GR_AUDIO_REGISTRY_H */
diff --git a/gr-audio/lib/gri_alsa.cc b/gr-audio/lib/gri_alsa.cc
new file mode 100644
index 0000000000..d9fda0f7d4
--- /dev/null
+++ b/gr-audio/lib/gri_alsa.cc
@@ -0,0 +1,175 @@
+/* -*- 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 <gri_alsa.h>
+#include <algorithm>
+
+static snd_pcm_access_t access_types[] = {
+ SND_PCM_ACCESS_MMAP_INTERLEAVED,
+ SND_PCM_ACCESS_MMAP_NONINTERLEAVED,
+ SND_PCM_ACCESS_MMAP_COMPLEX,
+ SND_PCM_ACCESS_RW_INTERLEAVED,
+ SND_PCM_ACCESS_RW_NONINTERLEAVED
+};
+
+static snd_pcm_format_t format_types[] = {
+ // SND_PCM_FORMAT_UNKNOWN,
+ SND_PCM_FORMAT_S8,
+ SND_PCM_FORMAT_U8,
+ SND_PCM_FORMAT_S16_LE,
+ SND_PCM_FORMAT_S16_BE,
+ SND_PCM_FORMAT_U16_LE,
+ SND_PCM_FORMAT_U16_BE,
+ SND_PCM_FORMAT_S24_LE,
+ SND_PCM_FORMAT_S24_BE,
+ SND_PCM_FORMAT_U24_LE,
+ SND_PCM_FORMAT_U24_BE,
+ SND_PCM_FORMAT_S32_LE,
+ SND_PCM_FORMAT_S32_BE,
+ SND_PCM_FORMAT_U32_LE,
+ SND_PCM_FORMAT_U32_BE,
+ SND_PCM_FORMAT_FLOAT_LE,
+ SND_PCM_FORMAT_FLOAT_BE,
+ SND_PCM_FORMAT_FLOAT64_LE,
+ SND_PCM_FORMAT_FLOAT64_BE,
+ SND_PCM_FORMAT_IEC958_SUBFRAME_LE,
+ SND_PCM_FORMAT_IEC958_SUBFRAME_BE,
+ SND_PCM_FORMAT_MU_LAW,
+ SND_PCM_FORMAT_A_LAW,
+ SND_PCM_FORMAT_IMA_ADPCM,
+ SND_PCM_FORMAT_MPEG,
+ SND_PCM_FORMAT_GSM,
+ SND_PCM_FORMAT_SPECIAL,
+ SND_PCM_FORMAT_S24_3LE,
+ SND_PCM_FORMAT_S24_3BE,
+ SND_PCM_FORMAT_U24_3LE,
+ SND_PCM_FORMAT_U24_3BE,
+ SND_PCM_FORMAT_S20_3LE,
+ SND_PCM_FORMAT_S20_3BE,
+ SND_PCM_FORMAT_U20_3LE,
+ SND_PCM_FORMAT_U20_3BE,
+ SND_PCM_FORMAT_S18_3LE,
+ SND_PCM_FORMAT_S18_3BE,
+ SND_PCM_FORMAT_U18_3LE,
+ SND_PCM_FORMAT_U18_3BE
+};
+
+static unsigned int test_rates[] = {
+ 8000, 16000, 22050, 32000, 44100, 48000, 96000, 192000
+};
+
+#define NELEMS(x) (sizeof(x)/sizeof(x[0]))
+
+void
+gri_alsa_dump_hw_params (snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams, FILE *fp)
+{
+ fprintf (fp, "PCM name: %s\n", snd_pcm_name (pcm));
+
+ fprintf (fp, "Access types:\n");
+ for (unsigned i = 0; i < NELEMS (access_types); i++){
+ snd_pcm_access_t at = access_types[i];
+ fprintf (fp, " %-20s %s\n",
+ snd_pcm_access_name (at),
+ snd_pcm_hw_params_test_access (pcm, hwparams, at) == 0 ? "YES" : "NO");
+ }
+
+ fprintf (fp, "Formats:\n");
+ for (unsigned i = 0; i < NELEMS (format_types); i++){
+ snd_pcm_format_t ft = format_types[i];
+ if (0)
+ fprintf (fp, " %-20s %s\n",
+ snd_pcm_format_name (ft),
+ snd_pcm_hw_params_test_format (pcm, hwparams, ft) == 0 ? "YES" : "NO");
+ else {
+ if (snd_pcm_hw_params_test_format (pcm, hwparams, ft) == 0)
+ fprintf (fp, " %-20s YES\n", snd_pcm_format_name (ft));
+ }
+ }
+
+ fprintf (fp, "Number of channels\n");
+ unsigned int min_chan, max_chan;
+ snd_pcm_hw_params_get_channels_min (hwparams, &min_chan);
+ snd_pcm_hw_params_get_channels_max (hwparams, &max_chan);
+ fprintf (fp, " min channels: %d\n", min_chan);
+ fprintf (fp, " max channels: %d\n", max_chan);
+ unsigned int chan;
+ max_chan = std::min (max_chan, 16U); // truncate display...
+ for (chan = min_chan; chan <= max_chan; chan++){
+ fprintf (fp, " %d channels\t%s\n", chan,
+ snd_pcm_hw_params_test_channels (pcm, hwparams, chan) == 0 ? "YES" : "NO");
+ }
+
+ fprintf (fp, "Sample Rates:\n");
+ unsigned int min_rate, max_rate;
+ int min_dir, max_dir;
+
+ snd_pcm_hw_params_get_rate_min (hwparams, &min_rate, &min_dir);
+ snd_pcm_hw_params_get_rate_max (hwparams, &max_rate, &max_dir);
+ fprintf (fp, " min rate: %7d (dir = %d)\n", min_rate, min_dir);
+ fprintf (fp, " max rate: %7d (dir = %d)\n", max_rate, max_dir);
+ for (unsigned i = 0; i < NELEMS (test_rates); i++){
+ unsigned int rate = test_rates[i];
+ fprintf (fp, " %6u %s\n", rate,
+ snd_pcm_hw_params_test_rate (pcm, hwparams, rate, 0) == 0 ? "YES" : "NO");
+ }
+
+ fflush (fp);
+}
+
+bool
+gri_alsa_pick_acceptable_format (snd_pcm_t *pcm,
+ snd_pcm_hw_params_t *hwparams,
+ snd_pcm_format_t acceptable_formats[],
+ unsigned nacceptable_formats,
+ snd_pcm_format_t *selected_format,
+ const char *error_msg_tag,
+ bool verbose)
+{
+ int err;
+
+ // pick a format that we like...
+ for (unsigned i = 0; i < nacceptable_formats; i++){
+ if (snd_pcm_hw_params_test_format (pcm, hwparams,
+ acceptable_formats[i]) == 0){
+ err = snd_pcm_hw_params_set_format (pcm, hwparams, acceptable_formats[i]);
+ if (err < 0){
+ fprintf (stderr, "%s[%s]: failed to set format: %s\n",
+ error_msg_tag, snd_pcm_name (pcm), snd_strerror (err));
+ return false;
+ }
+ if (verbose)
+ fprintf (stdout, "%s[%s]: using %s\n",
+ error_msg_tag, snd_pcm_name (pcm),
+ snd_pcm_format_name (acceptable_formats[i]));
+ *selected_format = acceptable_formats[i];
+ return true;
+ }
+ }
+
+ fprintf (stderr, "%s[%s]: failed to find acceptable format",
+ error_msg_tag, snd_pcm_name (pcm));
+ return false;
+}
diff --git a/gr-audio/lib/gri_alsa.h b/gr-audio/lib/gri_alsa.h
new file mode 100644
index 0000000000..3d72fd950e
--- /dev/null
+++ b/gr-audio/lib/gri_alsa.h
@@ -0,0 +1,44 @@
+/* -*- 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.
+ */
+
+#ifndef INCLUDED_GRI_ALSA_H
+#define INCLUDED_GRI_ALSA_H
+
+#include <stdio.h>
+#include <alsa/asoundlib.h>
+
+void
+gri_alsa_dump_hw_params (snd_pcm_t *pcm,
+ snd_pcm_hw_params_t *hwparams,
+ FILE *fp);
+
+bool
+gri_alsa_pick_acceptable_format (snd_pcm_t *pcm,
+ snd_pcm_hw_params_t *hwparams,
+ snd_pcm_format_t acceptable_formats[],
+ unsigned nacceptable_formats,
+ snd_pcm_format_t *selected_format,
+ const char *error_msg_tag,
+ bool verbose);
+
+
+#endif /* INCLUDED_GRI_ALSA_H */
--
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