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Diffstat (limited to 'src/lingot-audio-alsa.c')
| -rw-r--r-- | src/lingot-audio-alsa.c | 205 |
1 files changed, 205 insertions, 0 deletions
diff --git a/src/lingot-audio-alsa.c b/src/lingot-audio-alsa.c new file mode 100644 index 0000000..2c134f3 --- /dev/null +++ b/src/lingot-audio-alsa.c @@ -0,0 +1,205 @@ +/* + * lingot, a musical instrument tuner. + * + * Copyright (C) 2004-2011 Ibán Cereijo Graña, Jairo Chapela Martínez. + * + * This file is part of lingot. + * + * lingot 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 2 of the License, or + * (at your option) any later version. + * + * lingot 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 lingot; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <stdlib.h> + +#include "lingot-defs.h" +#include "lingot-audio-alsa.h" +#include "lingot-i18n.h" +#include "lingot-msg.h" + +LingotAudioHandler* lingot_audio_alsa_new(char* device, int sample_rate) { + + LingotAudioHandler* audio = NULL; + +# ifdef ALSA + const char* exception; + snd_pcm_hw_params_t* hw_params = NULL; + int err; + char error_message[1000]; + unsigned int channels = 1; + + audio = malloc(sizeof(LingotAudioHandler)); + audio->read_buffer = NULL; + audio->audio_system = AUDIO_SYSTEM_ALSA; + audio->read_buffer_size = 128; // TODO: size up + + // ALSA allocates some mem to load its config file when we call + // snd_card_next. Now that we're done getting the info, let's tell ALSA + // to unload the info and free up that mem + snd_config_update_free_global(); + + audio->capture_handle = NULL; + + try { + if ((err = snd_pcm_open(&audio->capture_handle, device, + SND_PCM_STREAM_CAPTURE, 0)) < 0) { + sprintf(error_message, "Cannot open audio device %s.\n%s.", device, + snd_strerror(err)); + throw(error_message); + } + + strcpy(audio->device, device); + + if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) { + sprintf(error_message, + "Cannot allocate hardware parameter structure.\n%s.", + snd_strerror(err)); + throw(error_message); + } + + if ((err = snd_pcm_hw_params_any(audio->capture_handle, hw_params)) < 0) { + sprintf(error_message, + "Cannot initialize hardware parameter structure.\n%s.", + snd_strerror(err)); + throw(error_message); + } + + if ((err = snd_pcm_hw_params_set_access(audio->capture_handle, + hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { + sprintf(error_message, "Cannot set access type.\n%s", snd_strerror( + err)); + throw(error_message); + } + + if ((err = snd_pcm_hw_params_set_format(audio->capture_handle, + hw_params, SND_PCM_FORMAT_S16_LE)) < 0) { + sprintf(error_message, "Cannot set sample format.\n%s.", + snd_strerror(err)); + throw(error_message); + } + + unsigned int rate = sample_rate; + + if ((err = snd_pcm_hw_params_set_rate_near(audio->capture_handle, + hw_params, &rate, 0)) < 0) { + sprintf(error_message, "Cannot set sample rate.\n%s.", + snd_strerror(err)); + throw(error_message); + } + + audio->real_sample_rate = rate; + + if ((err = snd_pcm_hw_params_set_channels(audio->capture_handle, + hw_params, channels)) < 0) { + sprintf(error_message, "Cannot set channel count.\n%s.", + snd_strerror(err)); + throw(error_message); + } + + if ((err = snd_pcm_hw_params(audio->capture_handle, hw_params)) < 0) { + sprintf(error_message, "Cannot set parameters.\n%s.", snd_strerror( + err)); + throw(error_message); + } + + if ((err = snd_pcm_prepare(audio->capture_handle)) < 0) { + sprintf(error_message, + "Cannot prepare audio interface for use.\n%s.", + snd_strerror(err)); + throw(error_message); + } + + audio->read_buffer = malloc(channels * audio->read_buffer_size + * sizeof(SAMPLE_TYPE)); + memset(audio->read_buffer, 0, audio->read_buffer_size + * sizeof(SAMPLE_TYPE)); + } catch { + if (audio->capture_handle != NULL) + snd_pcm_close(audio->capture_handle); + free(audio); + audio = NULL; + lingot_msg_add_error(exception); + } + + if (hw_params != NULL) + snd_pcm_hw_params_free(hw_params); + +# else + lingot_msg_add_error( + _("The application has not been built with ALSA support")); +# endif + + return audio; +} + +void lingot_audio_alsa_destroy(LingotAudioHandler* audio) { +# ifdef ALSA + if (audio != NULL) { + snd_pcm_close(audio->capture_handle); + free(audio->read_buffer); + } +# endif +} + +int lingot_audio_alsa_read(LingotAudioHandler* audio) { +# ifdef ALSA + int temp_sret; + int i; + + temp_sret = snd_pcm_readi(audio->capture_handle, audio->read_buffer, + audio->read_buffer_size); + + // if (rand() < 0.001 * RAND_MAX) + // temp_sret = 0; + + if (temp_sret != audio->read_buffer_size) { + char buff[100]; + sprintf(buff, _("Read from audio interface failed.\n%s."), + snd_strerror(temp_sret)); + lingot_msg_add_error(buff); + return -1; + } + + // float point conversion + for (i = 0; i < audio->read_buffer_size; i++) { + audio->flt_read_buffer[i] = audio->read_buffer[i]; + } + +# endif + + return 0; +} + +LingotAudioSystemProperties* lingot_audio_alsa_get_audio_system_properties( + audio_system_t audio_system) { + + LingotAudioSystemProperties* result = + (LingotAudioSystemProperties*) malloc(1 + * sizeof(LingotAudioSystemProperties)); + + // TODO + result->forced_sample_rate = 0; + result->n_devices = 0; + result->devices = NULL; + + result->n_sample_rates = 5; + result->sample_rates = malloc(result->n_sample_rates * sizeof(int)); + result->sample_rates[0] = 8000; + result->sample_rates[1] = 11025; + result->sample_rates[2] = 22050; + result->sample_rates[3] = 44100; + result->sample_rates[4] = 48000; + + return result; +} + |