diff options
Diffstat (limited to 'gr-audio/lib/alsa/audio_alsa_sink.cc')
| -rw-r--r-- | gr-audio/lib/alsa/audio_alsa_sink.cc | 42 |
1 files changed, 21 insertions, 21 deletions
diff --git a/gr-audio/lib/alsa/audio_alsa_sink.cc b/gr-audio/lib/alsa/audio_alsa_sink.cc index 0bda42470..687f24bde 100644 --- a/gr-audio/lib/alsa/audio_alsa_sink.cc +++ b/gr-audio/lib/alsa/audio_alsa_sink.cc @@ -1,19 +1,19 @@ /* -*- 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, @@ -51,7 +51,7 @@ static snd_pcm_format_t acceptable_formats[] = { #define NELEMS(x) (sizeof(x)/sizeof(x[0])) -static std::string +static std::string default_device_name () { return gr_prefs::singleton()->get_string("audio_alsa", "default_output_device", "hw:0,0"); @@ -131,7 +131,7 @@ audio_alsa_sink::audio_alsa_sink (int sampling_rate, } 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 @@ -156,14 +156,14 @@ audio_alsa_sink::audio_alsa_sink (int sampling_rate, "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); @@ -204,7 +204,7 @@ audio_alsa_sink::audio_alsa_sink (int sampling_rate, &d_period_size, &dir); if (error < 0) bail ("get_period_size failed", error); - + set_output_multiple (d_period_size); } @@ -226,11 +226,11 @@ audio_alsa_sink::check_topology (int ninputs, int noutputs) 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){ @@ -249,7 +249,7 @@ audio_alsa_sink::check_topology (int ninputs, int noutputs) 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. @@ -327,7 +327,7 @@ audio_alsa_sink::work_s16 (int noutput_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; @@ -351,7 +351,7 @@ audio_alsa_sink::work_s16 (int noutput_items, for (unsigned int chan = 0; chan < nchan; chan++) in[chan] += d_period_size; - if (!write_buffer (buf, d_period_size, sizeof_frame)) + if (!write_buffer (buf, d_period_size, sizeof_frame)) return -1; // No fixing this problem. Say we're done. } @@ -369,7 +369,7 @@ audio_alsa_sink::work_s32 (int noutput_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; @@ -393,7 +393,7 @@ audio_alsa_sink::work_s32 (int noutput_items, for (unsigned int chan = 0; chan < nchan; chan++) in[chan] += d_period_size; - if (!write_buffer (buf, d_period_size, sizeof_frame)) + if (!write_buffer (buf, d_period_size, sizeof_frame)) return -1; // No fixing this problem. Say we're done. } @@ -411,7 +411,7 @@ audio_alsa_sink::work_s16_1x2 (int noutput_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]; @@ -435,7 +435,7 @@ audio_alsa_sink::work_s16_1x2 (int noutput_items, // update src pointers in[0] += d_period_size; - if (!write_buffer (buf, d_period_size, sizeof_frame)) + if (!write_buffer (buf, d_period_size, sizeof_frame)) return -1; // No fixing this problem. Say we're done. } @@ -453,7 +453,7 @@ audio_alsa_sink::work_s32_1x2 (int noutput_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]; @@ -477,7 +477,7 @@ audio_alsa_sink::work_s32_1x2 (int noutput_items, // update src pointers in[0] += d_period_size; - if (!write_buffer (buf, d_period_size, sizeof_frame)) + if (!write_buffer (buf, d_period_size, sizeof_frame)) return -1; // No fixing this problem. Say we're done. } @@ -496,7 +496,7 @@ audio_alsa_sink::write_buffer (const void *vbuffer, { if (d_ok_to_block == true) continue; // try again - + break; } |