diff options
Diffstat (limited to 'gr-audio/lib/alsa')
| -rw-r--r-- | gr-audio/lib/alsa/audio_alsa_sink.cc | 548 | ||||
| -rw-r--r-- | gr-audio/lib/alsa/audio_alsa_sink.h | 105 | ||||
| -rw-r--r-- | gr-audio/lib/alsa/audio_alsa_source.cc | 509 | ||||
| -rw-r--r-- | gr-audio/lib/alsa/audio_alsa_source.h | 107 | ||||
| -rw-r--r-- | gr-audio/lib/alsa/gr-audio-alsa.conf | 11 | ||||
| -rw-r--r-- | gr-audio/lib/alsa/gri_alsa.cc | 175 | ||||
| -rw-r--r-- | gr-audio/lib/alsa/gri_alsa.h | 44 |
7 files changed, 0 insertions, 1499 deletions
diff --git a/gr-audio/lib/alsa/audio_alsa_sink.cc b/gr-audio/lib/alsa/audio_alsa_sink.cc deleted file mode 100644 index 687f24bde..000000000 --- a/gr-audio/lib/alsa/audio_alsa_sink.cc +++ /dev/null @@ -1,548 +0,0 @@ -/* -*- 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> - -AUDIO_REGISTER_SINK(REG_PRIO_HIGH, alsa)( - int sampling_rate, const std::string &device_name, bool ok_to_block -){ - return 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_sync_block ("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 float scale_factor = std::pow(2.0f, 16-1) - 1; - - 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] * scale_factor); - } - } - - // 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 float scale_factor = std::pow(2.0f, 32-1) - 1; - - 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] * scale_factor); - } - } - - // 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 float scale_factor = std::pow(2.0f, 16-1) - 1; - - 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] * scale_factor); - 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 float scale_factor = std::pow(2.0f, 32-1) - 1; - - 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] * scale_factor); - 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/alsa/audio_alsa_sink.h b/gr-audio/lib/alsa/audio_alsa_sink.h deleted file mode 100644 index d456e53de..000000000 --- a/gr-audio/lib/alsa/audio_alsa_sink.h +++ /dev/null @@ -1,105 +0,0 @@ -/* -*- 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 - * \ingroup audio_blk - * - * 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 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/alsa/audio_alsa_source.cc b/gr-audio/lib/alsa/audio_alsa_source.cc deleted file mode 100644 index 9fdf80b43..000000000 --- a/gr-audio/lib/alsa/audio_alsa_source.cc +++ /dev/null @@ -1,509 +0,0 @@ -/* -*- c++ -*- */ -/* - * Copyright 2004-2011 Free Software Foundation, Inc. - * - * This file is part of GNU Radio - * - * GNU Radio is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 3, or (at your option) - * any later version. - * - * GNU Radio is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with GNU Radio; see the file COPYING. If not, write to - * the Free Software Foundation, Inc., 51 Franklin Street, - * Boston, MA 02110-1301, USA. - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include "gr_audio_registry.h" -#include <audio_alsa_source.h> -#include <gr_io_signature.h> -#include <gr_prefs.h> -#include <stdio.h> -#include <iostream> -#include <stdexcept> -#include <gri_alsa.h> - -AUDIO_REGISTER_SOURCE(REG_PRIO_HIGH, alsa)( - int sampling_rate, const std::string &device_name, bool ok_to_block -){ - return audio_source::sptr(new audio_alsa_source(sampling_rate, device_name, ok_to_block)); -} - -static bool CHATTY_DEBUG = false; - -static snd_pcm_format_t acceptable_formats[] = { - // these are in our preferred order... - SND_PCM_FORMAT_S32, - SND_PCM_FORMAT_S16 -}; - -#define NELEMS(x) (sizeof(x)/sizeof(x[0])) - - -static std::string -default_device_name () -{ - return gr_prefs::singleton()->get_string("audio_alsa", "default_input_device", "hw:0,0"); -} - -static double -default_period_time () -{ - return std::max(0.001, gr_prefs::singleton()->get_double("audio_alsa", "period_time", 0.010)); -} - -static int -default_nperiods () -{ - return std::max(2L, gr_prefs::singleton()->get_long("audio_alsa", "nperiods", 4)); -} - -// ---------------------------------------------------------------- - -audio_alsa_source::audio_alsa_source (int sampling_rate, - const std::string device_name, - bool ok_to_block) - : 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; - - // 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 = &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 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 -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 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 -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 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 -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 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 -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/alsa/audio_alsa_source.h b/gr-audio/lib/alsa/audio_alsa_source.h deleted file mode 100644 index 320d49bd2..000000000 --- a/gr-audio/lib/alsa/audio_alsa_source.h +++ /dev/null @@ -1,107 +0,0 @@ -/* -*- 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 - * \ingroup audio_blk - * - * 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 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/alsa/gr-audio-alsa.conf b/gr-audio/lib/alsa/gr-audio-alsa.conf deleted file mode 100644 index 5cec63e7a..000000000 --- a/gr-audio/lib/alsa/gr-audio-alsa.conf +++ /dev/null @@ -1,11 +0,0 @@ -# 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/alsa/gri_alsa.cc b/gr-audio/lib/alsa/gri_alsa.cc deleted file mode 100644 index 7bae0937d..000000000 --- a/gr-audio/lib/alsa/gri_alsa.cc +++ /dev/null @@ -1,175 +0,0 @@ -/* -*- 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/alsa/gri_alsa.h b/gr-audio/lib/alsa/gri_alsa.h deleted file mode 100644 index 9c64e2c36..000000000 --- a/gr-audio/lib/alsa/gri_alsa.h +++ /dev/null @@ -1,44 +0,0 @@ -/* -*- 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 */ |