diff options
Diffstat (limited to 'gr-audio-alsa/src/audio_alsa_source.cc')
| -rw-r--r-- | gr-audio-alsa/src/audio_alsa_source.cc | 507 |
1 files changed, 0 insertions, 507 deletions
diff --git a/gr-audio-alsa/src/audio_alsa_source.cc b/gr-audio-alsa/src/audio_alsa_source.cc deleted file mode 100644 index 7b3d91b9d..000000000 --- a/gr-audio-alsa/src/audio_alsa_source.cc +++ /dev/null @@ -1,507 +0,0 @@ -/* -*- c++ -*- */ -/* - * Copyright 2004,2006,2010 Free Software Foundation, Inc. - * - * This file is part of GNU Radio - * - * GNU Radio is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 3, or (at your option) - * any later version. - * - * GNU Radio is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with GNU Radio; see the file COPYING. If not, write to - * the Free Software Foundation, Inc., 51 Franklin Street, - * Boston, MA 02110-1301, USA. - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include <audio_alsa_source.h> -#include <gr_io_signature.h> -#include <gr_prefs.h> -#include <stdio.h> -#include <iostream> -#include <stdexcept> -#include <gri_alsa.h> - - -static bool CHATTY_DEBUG = false; - -static snd_pcm_format_t acceptable_formats[] = { - // these are in our preferred order... - SND_PCM_FORMAT_S32, - SND_PCM_FORMAT_S16 -}; - -#define NELEMS(x) (sizeof(x)/sizeof(x[0])) - - -static std::string -default_device_name () -{ - return gr_prefs::singleton()->get_string("audio_alsa", "default_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_sptr -audio_alsa_make_source (int sampling_rate, const std::string dev, - bool ok_to_block) -{ - return gnuradio::get_initial_sptr(new audio_alsa_source (sampling_rate, dev, - ok_to_block)); -} - -audio_alsa_source::audio_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 -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"); -} |