/* -*- 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 */