diff options
Diffstat (limited to 'src/output')
-rw-r--r-- | src/output/alsa_plugin.c | 52 | ||||
-rw-r--r-- | src/output/recorder_output_plugin.c | 214 |
2 files changed, 264 insertions, 2 deletions
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c index 818c83ca2..f271668b1 100644 --- a/src/output/alsa_plugin.c +++ b/src/output/alsa_plugin.c @@ -183,6 +183,19 @@ get_bitformat(const struct audio_format *af) return SND_PCM_FORMAT_UNKNOWN; } +static snd_pcm_format_t +byteswap_bitformat(snd_pcm_format_t fmt) +{ + switch(fmt) { + case SND_PCM_FORMAT_S16_LE: return SND_PCM_FORMAT_S16_BE; + case SND_PCM_FORMAT_S24_LE: return SND_PCM_FORMAT_S24_BE; + case SND_PCM_FORMAT_S32_LE: return SND_PCM_FORMAT_S32_BE; + case SND_PCM_FORMAT_S16_BE: return SND_PCM_FORMAT_S16_LE; + case SND_PCM_FORMAT_S24_BE: return SND_PCM_FORMAT_S24_LE; + case SND_PCM_FORMAT_S32_BE: return SND_PCM_FORMAT_S32_LE; + default: return SND_PCM_FORMAT_UNKNOWN; + } +} /** * Set up the snd_pcm_t object which was opened by the caller. Set up * the configured settings and the audio format. @@ -208,7 +221,6 @@ alsa_setup(struct alsa_data *ad, struct audio_format *audio_format, configure_hw: /* configure HW params */ snd_pcm_hw_params_alloca(&hwparams); - cmd = "snd_pcm_hw_params_any"; err = snd_pcm_hw_params_any(ad->pcm, hwparams); if (err < 0) @@ -236,13 +248,38 @@ configure_hw: } err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, bitformat); + if (err == -EINVAL && + byteswap_bitformat(bitformat) != SND_PCM_FORMAT_UNKNOWN) { + err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, + byteswap_bitformat(bitformat)); + if (err == 0) { + g_debug("ALSA device \"%s\": converting %u bit to reverse-endian\n", + alsa_device(ad), audio_format->bits); + audio_format->reverse_endian = 1; + } + } if (err == -EINVAL && (audio_format->bits == 24 || audio_format->bits == 16)) { /* fall back to 32 bit, let pcm_convert.c do the conversion */ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, SND_PCM_FORMAT_S32); - if (err == 0) + if (err == 0) { + g_debug("ALSA device \"%s\": converting %u bit to 32 bit\n", + alsa_device(ad), audio_format->bits); + audio_format->bits = 32; + } + } + if (err == -EINVAL && (audio_format->bits == 24 || + audio_format->bits == 16)) { + /* fall back to 32 bit, let pcm_convert.c do the conversion */ + err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, + byteswap_bitformat(SND_PCM_FORMAT_S32)); + if (err == 0) { + g_debug("ALSA device \"%s\": converting %u bit to 32 bit backward-endian\n", + alsa_device(ad), audio_format->bits); audio_format->bits = 32; + audio_format->reverse_endian = 1; + } } if (err == -EINVAL && audio_format->bits != 16) { @@ -255,6 +292,17 @@ configure_hw: audio_format->bits = 16; } } + if (err == -EINVAL && audio_format->bits != 16) { + /* fall back to 16 bit, let pcm_convert.c do the conversion */ + err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, + byteswap_bitformat(SND_PCM_FORMAT_S16)); + if (err == 0) { + g_debug("ALSA device \"%s\": converting %u bit to 16 bit backward-endian\n", + alsa_device(ad), audio_format->bits); + audio_format->bits = 16; + audio_format->reverse_endian = 1; + } + } if (err < 0) { g_set_error(error, alsa_output_quark(), err, diff --git a/src/output/recorder_output_plugin.c b/src/output/recorder_output_plugin.c new file mode 100644 index 000000000..413e5d0d1 --- /dev/null +++ b/src/output/recorder_output_plugin.c @@ -0,0 +1,214 @@ +/* + * Copyright (C) 2003-2009 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program 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. + * + * This program 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 this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "output_api.h" +#include "encoder_plugin.h" +#include "encoder_list.h" + +#include <assert.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <fcntl.h> +#include <unistd.h> +#include <errno.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "recorder" + +struct recorder_output { + /** + * The configured encoder plugin. + */ + struct encoder *encoder; + + /** + * The destination file name. + */ + const char *path; + + /** + * The destination file descriptor. + */ + int fd; + + /** + * The buffer for encoder_read(). + */ + char buffer[32768]; +}; + +/** + * The quark used for GError.domain. + */ +static inline GQuark +recorder_output_quark(void) +{ + return g_quark_from_static_string("recorder_output"); +} + +static void * +recorder_output_init(G_GNUC_UNUSED const struct audio_format *audio_format, + const struct config_param *param, GError **error_r) +{ + struct recorder_output *recorder = g_new(struct recorder_output, 1); + const char *encoder_name; + const struct encoder_plugin *encoder_plugin; + + /* read configuration */ + + encoder_name = config_get_block_string(param, "encoder", "vorbis"); + encoder_plugin = encoder_plugin_get(encoder_name); + if (encoder_plugin == NULL) { + g_set_error(error_r, recorder_output_quark(), 0, + "No such encoder: %s", encoder_name); + return NULL; + } + + recorder->path = config_get_block_string(param, "path", NULL); + if (recorder->path == NULL) { + g_set_error(error_r, recorder_output_quark(), 0, + "'path' not configured"); + return NULL; + } + + /* initialize encoder */ + + recorder->encoder = encoder_init(encoder_plugin, param, error_r); + if (recorder->encoder == NULL) + return NULL; + + return recorder; +} + +static void +recorder_output_finish(void *data) +{ + struct recorder_output *recorder = data; + + encoder_finish(recorder->encoder); + g_free(recorder); +} + +/** + * Writes pending data from the encoder to the output file. + */ +static bool +recorder_output_encoder_to_file(struct recorder_output *recorder, + GError **error_r) +{ + size_t size = 0, position, nbytes; + + assert(recorder->fd >= 0); + + /* read from the encoder */ + + size = encoder_read(recorder->encoder, recorder->buffer, + sizeof(recorder->buffer)); + if (size == 0) + return true; + + /* write everything into the file */ + + position = 0; + while (true) { + nbytes = write(recorder->fd, recorder->buffer + position, + size - position); + if (nbytes > 0) { + position += (size_t)nbytes; + if (position >= size) + return true; + } else if (nbytes == 0) { + /* shouldn't happen for files */ + g_set_error(error_r, recorder_output_quark(), 0, + "write() returned 0"); + return false; + } else if (errno != EINTR) { + g_set_error(error_r, recorder_output_quark(), 0, + "Failed to write to '%s': %s", + recorder->path, g_strerror(errno)); + return false; + } + } +} + +static bool +recorder_output_open(void *data, struct audio_format *audio_format, + GError **error_r) +{ + struct recorder_output *recorder = data; + bool success; + + /* create the output file */ + + recorder->fd = creat(recorder->path, 0666); + if (recorder->fd < 0) { + g_set_error(error_r, recorder_output_quark(), 0, + "Failed to create '%s': %s", + recorder->path, g_strerror(errno)); + return false; + } + + /* open the encoder */ + + success = encoder_open(recorder->encoder, audio_format, error_r); + if (!success) { + close(recorder->fd); + unlink(recorder->path); + return false; + } + + return true; +} + +static void +recorder_output_close(void *data) +{ + struct recorder_output *recorder = data; + + /* flush the encoder and write the rest to the file */ + + if (encoder_flush(recorder->encoder, NULL)) + recorder_output_encoder_to_file(recorder, NULL); + + /* now really close everything */ + + encoder_close(recorder->encoder); + + close(recorder->fd); +} + +static size_t +recorder_output_play(void *data, const void *chunk, size_t size, + GError **error_r) +{ + struct recorder_output *recorder = data; + + return encoder_write(recorder->encoder, chunk, size, error_r) && + recorder_output_encoder_to_file(recorder, error_r) + ? size : 0; +} + +const struct audio_output_plugin recorder_output_plugin = { + .name = "recorder", + .init = recorder_output_init, + .finish = recorder_output_finish, + .open = recorder_output_open, + .close = recorder_output_close, + .play = recorder_output_play, +}; |