From 5748eff26f835bffedb41bd5748ccbaefbe1e02f Mon Sep 17 00:00:00 2001
From: Johnathan Corgan <jcorgan@corganenterprises.com>
Date: Mon, 14 Mar 2011 10:26:56 -0700
Subject: audio: remove obsoleted individual top-level components
---
gr-audio-alsa/src/audio_alsa_source.cc | 507 ---------------------------------
1 file changed, 507 deletions(-)
delete mode 100644 gr-audio-alsa/src/audio_alsa_source.cc
(limited to 'gr-audio-alsa/src/audio_alsa_source.cc')
diff --git a/gr-audio-alsa/src/audio_alsa_source.cc b/gr-audio-alsa/src/audio_alsa_source.cc
deleted file mode 100644
index 7b3d91b9dd..0000000000
--- a/gr-audio-alsa/src/audio_alsa_source.cc
+++ /dev/null
@@ -1,507 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2004,2006,2010 Free Software Foundation, Inc.
- *
- * This file is part of GNU Radio
- *
- * GNU Radio is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3, or (at your option)
- * any later version.
- *
- * GNU Radio is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with GNU Radio; see the file COPYING. If not, write to
- * the Free Software Foundation, Inc., 51 Franklin Street,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <audio_alsa_source.h>
-#include <gr_io_signature.h>
-#include <gr_prefs.h>
-#include <stdio.h>
-#include <iostream>
-#include <stdexcept>
-#include <gri_alsa.h>
-
-
-static bool CHATTY_DEBUG = false;
-
-static snd_pcm_format_t acceptable_formats[] = {
- // these are in our preferred order...
- SND_PCM_FORMAT_S32,
- SND_PCM_FORMAT_S16
-};
-
-#define NELEMS(x) (sizeof(x)/sizeof(x[0]))
-
-
-static std::string
-default_device_name ()
-{
- return gr_prefs::singleton()->get_string("audio_alsa", "default_input_device", "hw:0,0");
-}
-
-static double
-default_period_time ()
-{
- return std::max(0.001, gr_prefs::singleton()->get_double("audio_alsa", "period_time", 0.010));
-}
-
-static int
-default_nperiods ()
-{
- return std::max(2L, gr_prefs::singleton()->get_long("audio_alsa", "nperiods", 4));
-}
-
-// ----------------------------------------------------------------
-
-audio_alsa_source_sptr
-audio_alsa_make_source (int sampling_rate, const std::string dev,
- bool ok_to_block)
-{
- return gnuradio::get_initial_sptr(new audio_alsa_source (sampling_rate, dev,
- ok_to_block));
-}
-
-audio_alsa_source::audio_alsa_source (int sampling_rate,
- const std::string device_name,
- bool ok_to_block)
- : gr_sync_block ("audio_alsa_source",
- gr_make_io_signature (0, 0, 0),
- gr_make_io_signature (0, 0, 0)),
- d_sampling_rate (sampling_rate),
- d_device_name (device_name.empty() ? default_device_name() : device_name),
- d_pcm_handle (0),
- d_hw_params ((snd_pcm_hw_params_t *)(new char[snd_pcm_hw_params_sizeof()])),
- d_sw_params ((snd_pcm_sw_params_t *)(new char[snd_pcm_sw_params_sizeof()])),
- d_nperiods (default_nperiods()),
- d_period_time_us ((unsigned int) (default_period_time() * 1e6)),
- d_period_size (0),
- d_buffer_size_bytes (0), d_buffer (0),
- d_worker (0), d_hw_nchan (0),
- d_special_case_stereo_to_mono (false),
- d_noverruns (0), d_nsuspends (0)
-{
-
- CHATTY_DEBUG = gr_prefs::singleton()->get_bool("audio_alsa", "verbose", false);
-
- int error;
- int dir;
-
- // open the device for capture
- error = snd_pcm_open(&d_pcm_handle, d_device_name.c_str (),
- SND_PCM_STREAM_CAPTURE, 0);
- if (error < 0){
- fprintf (stderr, "audio_alsa_source[%s]: %s\n",
- d_device_name.c_str(), snd_strerror(error));
- throw std::runtime_error ("audio_alsa_source");
- }
-
- // Fill params with a full configuration space for a PCM.
- error = snd_pcm_hw_params_any(d_pcm_handle, d_hw_params);
- if (error < 0)
- bail ("broken configuration for playback", error);
-
- if (CHATTY_DEBUG)
- gri_alsa_dump_hw_params (d_pcm_handle, d_hw_params, stdout);
-
- // now that we know how many channels the h/w can handle, set output signature
- unsigned int umax_chan;
- unsigned int umin_chan;
- snd_pcm_hw_params_get_channels_min (d_hw_params, &umin_chan);
- snd_pcm_hw_params_get_channels_max (d_hw_params, &umax_chan);
- int min_chan = std::min (umin_chan, 1000U);
- int max_chan = std::min (umax_chan, 1000U);
-
- // As a special case, if the hw's min_chan is two, we'll accept
- // a single output and handle the demux ourselves.
-
- if (min_chan == 2){
- min_chan = 1;
- d_special_case_stereo_to_mono = true;
- }
-
- set_output_signature (gr_make_io_signature (min_chan, max_chan,
- sizeof (float)));
-
- // fill in portions of the d_hw_params that we know now...
-
- // Specify the access methods we implement
- // For now, we only handle RW_INTERLEAVED...
- snd_pcm_access_mask_t *access_mask;
- snd_pcm_access_mask_t **access_mask_ptr = &access_mask; // FIXME: workaround for compiler warning
- snd_pcm_access_mask_alloca (access_mask_ptr);
- snd_pcm_access_mask_none (access_mask);
- snd_pcm_access_mask_set (access_mask, SND_PCM_ACCESS_RW_INTERLEAVED);
- // snd_pcm_access_mask_set (access_mask, SND_PCM_ACCESS_RW_NONINTERLEAVED);
-
- if ((error = snd_pcm_hw_params_set_access_mask (d_pcm_handle,
- d_hw_params, access_mask)) < 0)
- bail ("failed to set access mask", error);
-
-
- // set sample format
- if (!gri_alsa_pick_acceptable_format (d_pcm_handle, d_hw_params,
- acceptable_formats,
- NELEMS (acceptable_formats),
- &d_format,
- "audio_alsa_source",
- CHATTY_DEBUG))
- throw std::runtime_error ("audio_alsa_source");
-
-
- // sampling rate
- unsigned int orig_sampling_rate = d_sampling_rate;
- if ((error = snd_pcm_hw_params_set_rate_near (d_pcm_handle, d_hw_params,
- &d_sampling_rate, 0)) < 0)
- bail ("failed to set rate near", error);
-
- if (orig_sampling_rate != d_sampling_rate){
- fprintf (stderr, "audio_alsa_source[%s]: unable to support sampling rate %d\n",
- snd_pcm_name (d_pcm_handle), orig_sampling_rate);
- fprintf (stderr, " card requested %d instead.\n", d_sampling_rate);
- }
-
- /*
- * ALSA transfers data in units of "periods".
- * We indirectly determine the underlying buffersize by specifying
- * the number of periods we want (typically 4) and the length of each
- * period in units of time (typically 1ms).
- */
- unsigned int min_nperiods, max_nperiods;
- snd_pcm_hw_params_get_periods_min (d_hw_params, &min_nperiods, &dir);
- snd_pcm_hw_params_get_periods_max (d_hw_params, &max_nperiods, &dir);
- //fprintf (stderr, "alsa_source: min_nperiods = %d, max_nperiods = %d\n",
- // min_nperiods, max_nperiods);
-
-
- unsigned int orig_nperiods = d_nperiods;
- d_nperiods = std::min (std::max (min_nperiods, d_nperiods), max_nperiods);
-
- // adjust period time so that total buffering remains more-or-less constant
- d_period_time_us = (d_period_time_us * orig_nperiods) / d_nperiods;
-
- error = snd_pcm_hw_params_set_periods (d_pcm_handle, d_hw_params,
- d_nperiods, 0);
- if (error < 0)
- bail ("set_periods failed", error);
-
- dir = 0;
- error = snd_pcm_hw_params_set_period_time_near (d_pcm_handle, d_hw_params,
- &d_period_time_us, &dir);
- if (error < 0)
- bail ("set_period_time_near failed", error);
-
- dir = 0;
- error = snd_pcm_hw_params_get_period_size (d_hw_params,
- &d_period_size, &dir);
- if (error < 0)
- bail ("get_period_size failed", error);
-
- set_output_multiple (d_period_size);
-}
-
-bool
-audio_alsa_source::check_topology (int ninputs, int noutputs)
-{
- // noutputs is how many channels the user has connected.
- // Now we can finish up setting up the hw params...
-
- unsigned int nchan = noutputs;
- int err;
-
- // FIXME check_topology may be called more than once.
- // Ensure that the pcm is in a state where we can still mess with the hw_params
-
- bool special_case = nchan == 1 && d_special_case_stereo_to_mono;
- if (special_case)
- nchan = 2;
-
- d_hw_nchan = nchan;
- err = snd_pcm_hw_params_set_channels (d_pcm_handle, d_hw_params, d_hw_nchan);
- if (err < 0){
- output_error_msg ("set_channels failed", err);
- return false;
- }
-
- // set the parameters into the driver...
- err = snd_pcm_hw_params(d_pcm_handle, d_hw_params);
- if (err < 0){
- output_error_msg ("snd_pcm_hw_params failed", err);
- return false;
- }
-
- d_buffer_size_bytes =
- d_period_size * d_hw_nchan * snd_pcm_format_size (d_format, 1);
-
- d_buffer = new char [d_buffer_size_bytes];
-
- if (CHATTY_DEBUG)
- fprintf (stdout, "audio_alsa_source[%s]: sample resolution = %d bits\n",
- snd_pcm_name (d_pcm_handle),
- snd_pcm_hw_params_get_sbits (d_hw_params));
-
- switch (d_format){
- case SND_PCM_FORMAT_S16:
- if (special_case)
- d_worker = &audio_alsa_source::work_s16_2x1;
- else
- d_worker = &audio_alsa_source::work_s16;
- break;
-
- case SND_PCM_FORMAT_S32:
- if (special_case)
- d_worker = &audio_alsa_source::work_s32_2x1;
- else
- d_worker = &audio_alsa_source::work_s32;
- break;
-
- default:
- assert (0);
- }
-
- return true;
-}
-
-audio_alsa_source::~audio_alsa_source ()
-{
- if (snd_pcm_state (d_pcm_handle) == SND_PCM_STATE_RUNNING)
- snd_pcm_drop (d_pcm_handle);
-
- snd_pcm_close(d_pcm_handle);
- delete [] ((char *) d_hw_params);
- delete [] ((char *) d_sw_params);
- delete [] d_buffer;
-}
-
-int
-audio_alsa_source::work (int noutput_items,
- gr_vector_const_void_star &input_items,
- gr_vector_void_star &output_items)
-{
- assert ((noutput_items % d_period_size) == 0);
- assert (noutput_items != 0);
-
- // this is a call through a pointer to a method...
- return (this->*d_worker)(noutput_items, input_items, output_items);
-}
-
-/*
- * Work function that deals with float to S16 conversion
- */
-int
-audio_alsa_source::work_s16 (int noutput_items,
- gr_vector_const_void_star &input_items,
- gr_vector_void_star &output_items)
-{
- typedef gr_int16 sample_t; // the type of samples we're creating
- static const int NBITS = 16; // # of bits in a sample
-
- unsigned int nchan = output_items.size ();
- float **out = (float **) &output_items[0];
- sample_t *buf = (sample_t *) d_buffer;
- int bi;
-
- unsigned int sizeof_frame = d_hw_nchan * sizeof (sample_t);
- assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
-
- // To minimize latency, return at most a single period's worth of samples.
- // [We could also read the first one in a blocking mode and subsequent
- // ones in non-blocking mode, but we'll leave that for later (or never).]
-
- if (!read_buffer (buf, d_period_size, sizeof_frame))
- return -1; // No fixing this problem. Say we're done.
-
- // process one period of data
- bi = 0;
- for (unsigned int i = 0; i < d_period_size; i++){
- for (unsigned int chan = 0; chan < nchan; chan++){
- out[chan][i] = (float) buf[bi++] * (1.0 / (float) ((1L << (NBITS-1)) - 1));
- }
- }
-
- return d_period_size;
-}
-
-/*
- * Work function that deals with float to S16 conversion
- * and stereo to mono kludge...
- */
-int
-audio_alsa_source::work_s16_2x1 (int noutput_items,
- gr_vector_const_void_star &input_items,
- gr_vector_void_star &output_items)
-{
- typedef gr_int16 sample_t; // the type of samples we're creating
- static const int NBITS = 16; // # of bits in a sample
-
- unsigned int nchan = output_items.size ();
- float **out = (float **) &output_items[0];
- sample_t *buf = (sample_t *) d_buffer;
- int bi;
-
- assert (nchan == 1);
-
- unsigned int sizeof_frame = d_hw_nchan * sizeof (sample_t);
- assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
-
- // To minimize latency, return at most a single period's worth of samples.
- // [We could also read the first one in a blocking mode and subsequent
- // ones in non-blocking mode, but we'll leave that for later (or never).]
-
- if (!read_buffer (buf, d_period_size, sizeof_frame))
- return -1; // No fixing this problem. Say we're done.
-
- // process one period of data
- bi = 0;
- for (unsigned int i = 0; i < d_period_size; i++){
- int t = (buf[bi] + buf[bi+1]) / 2;
- bi += 2;
- out[0][i] = (float) t * (1.0 / (float) ((1L << (NBITS-1)) - 1));
- }
-
- return d_period_size;
-}
-
-/*
- * Work function that deals with float to S32 conversion
- */
-int
-audio_alsa_source::work_s32 (int noutput_items,
- gr_vector_const_void_star &input_items,
- gr_vector_void_star &output_items)
-{
- typedef gr_int32 sample_t; // the type of samples we're creating
- static const int NBITS = 32; // # of bits in a sample
-
- unsigned int nchan = output_items.size ();
- float **out = (float **) &output_items[0];
- sample_t *buf = (sample_t *) d_buffer;
- int bi;
-
- unsigned int sizeof_frame = d_hw_nchan * sizeof (sample_t);
- assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
-
- // To minimize latency, return at most a single period's worth of samples.
- // [We could also read the first one in a blocking mode and subsequent
- // ones in non-blocking mode, but we'll leave that for later (or never).]
-
- if (!read_buffer (buf, d_period_size, sizeof_frame))
- return -1; // No fixing this problem. Say we're done.
-
- // process one period of data
- bi = 0;
- for (unsigned int i = 0; i < d_period_size; i++){
- for (unsigned int chan = 0; chan < nchan; chan++){
- out[chan][i] = (float) buf[bi++] * (1.0 / (float) ((1L << (NBITS-1)) - 1));
- }
- }
-
- return d_period_size;
-}
-
-/*
- * Work function that deals with float to S32 conversion
- * and stereo to mono kludge...
- */
-int
-audio_alsa_source::work_s32_2x1 (int noutput_items,
- gr_vector_const_void_star &input_items,
- gr_vector_void_star &output_items)
-{
- typedef gr_int32 sample_t; // the type of samples we're creating
- static const int NBITS = 32; // # of bits in a sample
-
- unsigned int nchan = output_items.size ();
- float **out = (float **) &output_items[0];
- sample_t *buf = (sample_t *) d_buffer;
- int bi;
-
- assert (nchan == 1);
-
- unsigned int sizeof_frame = d_hw_nchan * sizeof (sample_t);
- assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
-
- // To minimize latency, return at most a single period's worth of samples.
- // [We could also read the first one in a blocking mode and subsequent
- // ones in non-blocking mode, but we'll leave that for later (or never).]
-
- if (!read_buffer (buf, d_period_size, sizeof_frame))
- return -1; // No fixing this problem. Say we're done.
-
- // process one period of data
- bi = 0;
- for (unsigned int i = 0; i < d_period_size; i++){
- int t = (buf[bi] + buf[bi+1]) / 2;
- bi += 2;
- out[0][i] = (float) t * (1.0 / (float) ((1L << (NBITS-1)) - 1));
- }
-
- return d_period_size;
-}
-
-bool
-audio_alsa_source::read_buffer (void *vbuffer, unsigned nframes, unsigned sizeof_frame)
-{
- unsigned char *buffer = (unsigned char *) vbuffer;
-
- while (nframes > 0){
- int r = snd_pcm_readi (d_pcm_handle, buffer, nframes);
- if (r == -EAGAIN)
- continue; // try again
-
- else if (r == -EPIPE){ // overrun
- d_noverruns++;
- fputs ("aO", stderr);
- if ((r = snd_pcm_prepare (d_pcm_handle)) < 0){
- output_error_msg ("snd_pcm_prepare failed. Can't recover from overrun", r);
- return false;
- }
- continue; // try again
- }
-
- else if (r == -ESTRPIPE){ // h/w is suspended (whatever that means)
- // This is apparently related to power management
- d_nsuspends++;
- if ((r = snd_pcm_resume (d_pcm_handle)) < 0){
- output_error_msg ("failed to resume from suspend", r);
- return false;
- }
- continue; // try again
- }
-
- else if (r < 0){
- output_error_msg ("snd_pcm_readi failed", r);
- return false;
- }
-
- nframes -= r;
- buffer += r * sizeof_frame;
- }
-
- return true;
-}
-
-
-void
-audio_alsa_source::output_error_msg (const char *msg, int err)
-{
- fprintf (stderr, "audio_alsa_source[%s]: %s: %s\n",
- snd_pcm_name (d_pcm_handle), msg, snd_strerror (err));
-}
-
-void
-audio_alsa_source::bail (const char *msg, int err) throw (std::runtime_error)
-{
- output_error_msg (msg, err);
- throw std::runtime_error ("audio_alsa_source");
-}
--
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