3 * Copyright 2004 Free Software Foundation, Inc.
5 * This file is part of GNU Radio
7 * GNU Radio is free software; you can redistribute it and/or modify
8 * it under the terms of the GNU General Public License as published by
9 * the Free Software Foundation; either version 3, or (at your option)
12 * GNU Radio is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
15 * GNU General Public License for more details.
17 * You should have received a copy of the GNU General Public License
18 * along with GNU Radio; see the file COPYING. If not, write to
19 * the Free Software Foundation, Inc., 51 Franklin Street,
20 * Boston, MA 02110-1301, USA.
27 #include <audio_alsa_sink.h>
28 #include <gr_io_signature.h>
36 static bool CHATTY_DEBUG = false;
39 static snd_pcm_format_t acceptable_formats[] = {
40 // these are in our preferred order...
45 #define NELEMS(x) (sizeof(x)/sizeof(x[0]))
49 default_device_name ()
51 return gr_prefs::singleton()->get_string("audio_alsa", "default_output_device", "hw:0,0");
55 default_period_time ()
57 return std::max(0.001, gr_prefs::singleton()->get_double("audio_alsa", "period_time", 0.010));
63 return std::max(2L, gr_prefs::singleton()->get_long("audio_alsa", "nperiods", 4));
66 // ----------------------------------------------------------------
69 audio_alsa_make_sink (int sampling_rate,
70 const std::string dev,
73 return audio_alsa_sink_sptr (new audio_alsa_sink (sampling_rate, dev,
77 audio_alsa_sink::audio_alsa_sink (int sampling_rate,
78 const std::string device_name,
80 : gr_sync_block ("audio_alsa_sink",
81 gr_make_io_signature (0, 0, 0),
82 gr_make_io_signature (0, 0, 0)),
83 d_sampling_rate (sampling_rate),
84 d_device_name (device_name.empty() ? default_device_name() : device_name),
86 d_hw_params ((snd_pcm_hw_params_t *)(new char[snd_pcm_hw_params_sizeof()])),
87 d_sw_params ((snd_pcm_sw_params_t *)(new char[snd_pcm_sw_params_sizeof()])),
88 d_nperiods (default_nperiods()),
89 d_period_time_us ((unsigned int) (default_period_time() * 1e6)),
91 d_buffer_size_bytes (0), d_buffer (0),
92 d_worker (0), d_special_case_mono_to_stereo (false),
93 d_nunderuns (0), d_nsuspends (0)
95 CHATTY_DEBUG = gr_prefs::singleton()->get_bool("audio_alsa", "verbose", false);
100 // open the device for playback
101 error = snd_pcm_open(&d_pcm_handle, d_device_name.c_str (),
102 SND_PCM_STREAM_PLAYBACK, 0);
104 fprintf (stderr, "audio_alsa_sink[%s]: %s\n",
105 d_device_name.c_str(), snd_strerror(error));
106 throw std::runtime_error ("audio_alsa_sink");
109 // Fill params with a full configuration space for a PCM.
110 error = snd_pcm_hw_params_any(d_pcm_handle, d_hw_params);
112 bail ("broken configuration for playback", error);
116 gri_alsa_dump_hw_params (d_pcm_handle, d_hw_params, stdout);
119 // now that we know how many channels the h/w can handle, set input signature
120 unsigned int umin_chan, umax_chan;
121 snd_pcm_hw_params_get_channels_min (d_hw_params, &umin_chan);
122 snd_pcm_hw_params_get_channels_max (d_hw_params, &umax_chan);
123 int min_chan = std::min (umin_chan, 1000U);
124 int max_chan = std::min (umax_chan, 1000U);
126 // As a special case, if the hw's min_chan is two, we'll accept
127 // a single input and handle the duplication ourselves.
131 d_special_case_mono_to_stereo = true;
133 set_input_signature (gr_make_io_signature (min_chan, max_chan,
136 // fill in portions of the d_hw_params that we know now...
138 // Specify the access methods we implement
139 // For now, we only handle RW_INTERLEAVED...
140 snd_pcm_access_mask_t *access_mask;
141 snd_pcm_access_mask_alloca (&access_mask);
142 snd_pcm_access_mask_none (access_mask);
143 snd_pcm_access_mask_set (access_mask, SND_PCM_ACCESS_RW_INTERLEAVED);
144 // snd_pcm_access_mask_set (access_mask, SND_PCM_ACCESS_RW_NONINTERLEAVED);
146 if ((error = snd_pcm_hw_params_set_access_mask (d_pcm_handle,
147 d_hw_params, access_mask)) < 0)
148 bail ("failed to set access mask", error);
152 if (!gri_alsa_pick_acceptable_format (d_pcm_handle, d_hw_params,
154 NELEMS (acceptable_formats),
158 throw std::runtime_error ("audio_alsa_sink");
162 unsigned int orig_sampling_rate = d_sampling_rate;
163 if ((error = snd_pcm_hw_params_set_rate_near (d_pcm_handle, d_hw_params,
164 &d_sampling_rate, 0)) < 0)
165 bail ("failed to set rate near", error);
167 if (orig_sampling_rate != d_sampling_rate){
168 fprintf (stderr, "audio_alsa_sink[%s]: unable to support sampling rate %d\n",
169 snd_pcm_name (d_pcm_handle), orig_sampling_rate);
170 fprintf (stderr, " card requested %d instead.\n", d_sampling_rate);
174 * ALSA transfers data in units of "periods".
175 * We indirectly determine the underlying buffersize by specifying
176 * the number of periods we want (typically 4) and the length of each
177 * period in units of time (typically 1ms).
179 unsigned int min_nperiods, max_nperiods;
180 snd_pcm_hw_params_get_periods_min (d_hw_params, &min_nperiods, &dir);
181 snd_pcm_hw_params_get_periods_max (d_hw_params, &max_nperiods, &dir);
182 //fprintf (stderr, "alsa_sink: min_nperiods = %d, max_nperiods = %d\n",
183 // min_nperiods, max_nperiods);
185 unsigned int orig_nperiods = d_nperiods;
186 d_nperiods = std::min (std::max (min_nperiods, d_nperiods), max_nperiods);
188 // adjust period time so that total buffering remains more-or-less constant
189 d_period_time_us = (d_period_time_us * orig_nperiods) / d_nperiods;
191 error = snd_pcm_hw_params_set_periods (d_pcm_handle, d_hw_params,
194 bail ("set_periods failed", error);
197 error = snd_pcm_hw_params_set_period_time_near (d_pcm_handle, d_hw_params,
198 &d_period_time_us, &dir);
200 bail ("set_period_time_near failed", error);
203 error = snd_pcm_hw_params_get_period_size (d_hw_params,
204 &d_period_size, &dir);
206 bail ("get_period_size failed", error);
208 set_output_multiple (d_period_size);
213 audio_alsa_sink::check_topology (int ninputs, int noutputs)
215 // ninputs is how many channels the user has connected.
216 // Now we can finish up setting up the hw params...
221 // FIXME check_topology may be called more than once.
222 // Ensure that the pcm is in a state where we can still mess with the hw_params
224 bool special_case = nchan == 1 && d_special_case_mono_to_stereo;
228 err = snd_pcm_hw_params_set_channels (d_pcm_handle, d_hw_params, nchan);
231 output_error_msg ("set_channels failed", err);
235 // set the parameters into the driver...
236 err = snd_pcm_hw_params(d_pcm_handle, d_hw_params);
238 output_error_msg ("snd_pcm_hw_params failed", err);
242 // get current s/w params
243 err = snd_pcm_sw_params_current (d_pcm_handle, d_sw_params);
245 bail ("snd_pcm_sw_params_current", err);
247 // Tell the PCM device to wait to start until we've filled
248 // it's buffers half way full. This helps avoid audio underruns.
250 err = snd_pcm_sw_params_set_start_threshold(d_pcm_handle,
252 d_nperiods * d_period_size / 2);
254 bail ("snd_pcm_sw_params_set_start_threshold", err);
256 // store the s/w params
257 err = snd_pcm_sw_params (d_pcm_handle, d_sw_params);
259 bail ("snd_pcm_sw_params", err);
261 d_buffer_size_bytes =
262 d_period_size * nchan * snd_pcm_format_size (d_format, 1);
264 d_buffer = new char [d_buffer_size_bytes];
267 fprintf (stdout, "audio_alsa_sink[%s]: sample resolution = %d bits\n",
268 snd_pcm_name (d_pcm_handle),
269 snd_pcm_hw_params_get_sbits (d_hw_params));
272 case SND_PCM_FORMAT_S16:
274 d_worker = &audio_alsa_sink::work_s16_1x2;
276 d_worker = &audio_alsa_sink::work_s16;
279 case SND_PCM_FORMAT_S32:
281 d_worker = &audio_alsa_sink::work_s32_1x2;
283 d_worker = &audio_alsa_sink::work_s32;
293 audio_alsa_sink::~audio_alsa_sink ()
295 if (snd_pcm_state (d_pcm_handle) == SND_PCM_STATE_RUNNING)
296 snd_pcm_drop (d_pcm_handle);
298 snd_pcm_close(d_pcm_handle);
299 delete [] ((char *) d_hw_params);
300 delete [] ((char *) d_sw_params);
305 audio_alsa_sink::work (int noutput_items,
306 gr_vector_const_void_star &input_items,
307 gr_vector_void_star &output_items)
309 assert ((noutput_items % d_period_size) == 0);
311 // this is a call through a pointer to a method...
312 return (this->*d_worker)(noutput_items, input_items, output_items);
316 * Work function that deals with float to S16 conversion
319 audio_alsa_sink::work_s16 (int noutput_items,
320 gr_vector_const_void_star &input_items,
321 gr_vector_void_star &output_items)
323 typedef gr_int16 sample_t; // the type of samples we're creating
324 static const int NBITS = 16; // # of bits in a sample
326 unsigned int nchan = input_items.size ();
327 const float **in = (const float **) &input_items[0];
328 sample_t *buf = (sample_t *) d_buffer;
332 unsigned int sizeof_frame = nchan * sizeof (sample_t);
333 assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
335 for (n = 0; n < noutput_items; n += d_period_size){
337 // process one period of data
339 for (unsigned int i = 0; i < d_period_size; i++){
340 for (unsigned int chan = 0; chan < nchan; chan++){
341 buf[bi++] = (sample_t) (in[chan][i] * (float) ((1L << (NBITS-1)) - 1));
345 // update src pointers
346 for (unsigned int chan = 0; chan < nchan; chan++)
347 in[chan] += d_period_size;
349 if (!write_buffer (buf, d_period_size, sizeof_frame))
350 return -1; // No fixing this problem. Say we're done.
358 * Work function that deals with float to S32 conversion
361 audio_alsa_sink::work_s32 (int noutput_items,
362 gr_vector_const_void_star &input_items,
363 gr_vector_void_star &output_items)
365 typedef gr_int32 sample_t; // the type of samples we're creating
366 static const int NBITS = 32; // # of bits in a sample
368 unsigned int nchan = input_items.size ();
369 const float **in = (const float **) &input_items[0];
370 sample_t *buf = (sample_t *) d_buffer;
374 unsigned int sizeof_frame = nchan * sizeof (sample_t);
375 assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
377 for (n = 0; n < noutput_items; n += d_period_size){
379 // process one period of data
381 for (unsigned int i = 0; i < d_period_size; i++){
382 for (unsigned int chan = 0; chan < nchan; chan++){
383 buf[bi++] = (sample_t) (in[chan][i] * (float) ((1L << (NBITS-1)) - 1));
387 // update src pointers
388 for (unsigned int chan = 0; chan < nchan; chan++)
389 in[chan] += d_period_size;
391 if (!write_buffer (buf, d_period_size, sizeof_frame))
392 return -1; // No fixing this problem. Say we're done.
399 * Work function that deals with float to S16 conversion and
400 * mono to stereo kludge.
403 audio_alsa_sink::work_s16_1x2 (int noutput_items,
404 gr_vector_const_void_star &input_items,
405 gr_vector_void_star &output_items)
407 typedef gr_int16 sample_t; // the type of samples we're creating
408 static const int NBITS = 16; // # of bits in a sample
410 assert (input_items.size () == 1);
411 static const unsigned int nchan = 2;
412 const float **in = (const float **) &input_items[0];
413 sample_t *buf = (sample_t *) d_buffer;
417 unsigned int sizeof_frame = nchan * sizeof (sample_t);
418 assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
420 for (n = 0; n < noutput_items; n += d_period_size){
422 // process one period of data
424 for (unsigned int i = 0; i < d_period_size; i++){
425 sample_t t = (sample_t) (in[0][i] * (float) ((1L << (NBITS-1)) - 1));
430 // update src pointers
431 in[0] += d_period_size;
433 if (!write_buffer (buf, d_period_size, sizeof_frame))
434 return -1; // No fixing this problem. Say we're done.
441 * Work function that deals with float to S32 conversion and
442 * mono to stereo kludge.
445 audio_alsa_sink::work_s32_1x2 (int noutput_items,
446 gr_vector_const_void_star &input_items,
447 gr_vector_void_star &output_items)
449 typedef gr_int32 sample_t; // the type of samples we're creating
450 static const int NBITS = 32; // # of bits in a sample
452 assert (input_items.size () == 1);
453 static unsigned int nchan = 2;
454 const float **in = (const float **) &input_items[0];
455 sample_t *buf = (sample_t *) d_buffer;
459 unsigned int sizeof_frame = nchan * sizeof (sample_t);
460 assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
462 for (n = 0; n < noutput_items; n += d_period_size){
464 // process one period of data
466 for (unsigned int i = 0; i < d_period_size; i++){
467 sample_t t = (sample_t) (in[0][i] * (float) ((1L << (NBITS-1)) - 1));
472 // update src pointers
473 in[0] += d_period_size;
475 if (!write_buffer (buf, d_period_size, sizeof_frame))
476 return -1; // No fixing this problem. Say we're done.
483 audio_alsa_sink::write_buffer (const void *vbuffer,
484 unsigned nframes, unsigned sizeof_frame)
486 const unsigned char *buffer = (const unsigned char *) vbuffer;
489 int r = snd_pcm_writei (d_pcm_handle, buffer, nframes);
491 continue; // try again
493 else if (r == -EPIPE){ // underrun
495 fputs ("aU", stderr);
496 if ((r = snd_pcm_prepare (d_pcm_handle)) < 0){
497 output_error_msg ("snd_pcm_prepare failed. Can't recover from underrun", r);
500 continue; // try again
503 else if (r == -ESTRPIPE){ // h/w is suspended (whatever that means)
504 // This is apparently related to power management
506 if ((r = snd_pcm_resume (d_pcm_handle)) < 0){
507 output_error_msg ("failed to resume from suspend", r);
510 continue; // try again
514 output_error_msg ("snd_pcm_writei failed", r);
519 buffer += r * sizeof_frame;
527 audio_alsa_sink::output_error_msg (const char *msg, int err)
529 fprintf (stderr, "audio_alsa_sink[%s]: %s: %s\n",
530 snd_pcm_name (d_pcm_handle), msg, snd_strerror (err));
534 audio_alsa_sink::bail (const char *msg, int err) throw (std::runtime_error)
536 output_error_msg (msg, err);
537 throw std::runtime_error ("audio_alsa_sink");