summaryrefslogtreecommitdiff
path: root/gr-audio/lib/alsa/audio_alsa_sink.cc
diff options
context:
space:
mode:
Diffstat (limited to 'gr-audio/lib/alsa/audio_alsa_sink.cc')
-rw-r--r--gr-audio/lib/alsa/audio_alsa_sink.cc548
1 files changed, 548 insertions, 0 deletions
diff --git a/gr-audio/lib/alsa/audio_alsa_sink.cc b/gr-audio/lib/alsa/audio_alsa_sink.cc
new file mode 100644
index 000000000..0728f421c
--- /dev/null
+++ b/gr-audio/lib/alsa/audio_alsa_sink.cc
@@ -0,0 +1,548 @@
+/* -*- 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)
+ : audio_sink ("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 int NBITS = 16; // # of bits in a sample
+
+ 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] * (float) ((1L << (NBITS-1)) - 1));
+ }
+ }
+
+ // 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 int NBITS = 32; // # of bits in a sample
+
+ 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] * (float) ((1L << (NBITS-1)) - 1));
+ }
+ }
+
+ // 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 int NBITS = 16; // # of bits in a sample
+
+ 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] * (float) ((1L << (NBITS-1)) - 1));
+ 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 int NBITS = 32; // # of bits in a sample
+
+ 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] * (float) ((1L << (NBITS-1)) - 1));
+ 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");
+}