diff options
Diffstat (limited to 'gr-audio/lib/alsa/audio_alsa_source.cc')
| -rw-r--r-- | gr-audio/lib/alsa/audio_alsa_source.cc | 505 |
1 files changed, 505 insertions, 0 deletions
diff --git a/gr-audio/lib/alsa/audio_alsa_source.cc b/gr-audio/lib/alsa/audio_alsa_source.cc new file mode 100644 index 000000000..4f0042b22 --- /dev/null +++ b/gr-audio/lib/alsa/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> + +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; + + // 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"); +} |