/* * 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. */ /* * Media MVP audio output based on code from MVPMC project: * http://mvpmc.sourceforge.net/ */ #include "../output_api.h" #include <glib.h> #include <sys/types.h> #include <sys/stat.h> #include <sys/ioctl.h> #include <fcntl.h> #include <errno.h> #include <unistd.h> #include <stdlib.h> #undef G_LOG_DOMAIN #define G_LOG_DOMAIN "mvp" typedef struct { unsigned long dsp_status; unsigned long stream_decode_type; unsigned long sample_rate; unsigned long bit_rate; unsigned long raw[64 / sizeof(unsigned long)]; } aud_status_t; #define MVP_SET_AUD_STOP _IOW('a',1,int) #define MVP_SET_AUD_PLAY _IOW('a',2,int) #define MVP_SET_AUD_PAUSE _IOW('a',3,int) #define MVP_SET_AUD_UNPAUSE _IOW('a',4,int) #define MVP_SET_AUD_SRC _IOW('a',5,int) #define MVP_SET_AUD_MUTE _IOW('a',6,int) #define MVP_SET_AUD_BYPASS _IOW('a',8,int) #define MVP_SET_AUD_CHANNEL _IOW('a',9,int) #define MVP_GET_AUD_STATUS _IOR('a',10,aud_status_t) #define MVP_SET_AUD_VOLUME _IOW('a',13,int) #define MVP_GET_AUD_VOLUME _IOR('a',14,int) #define MVP_SET_AUD_STREAMTYPE _IOW('a',15,int) #define MVP_SET_AUD_FORMAT _IOW('a',16,int) #define MVP_GET_AUD_SYNC _IOR('a',21,pts_sync_data_t*) #define MVP_SET_AUD_STC _IOW('a',22,long long int *) #define MVP_SET_AUD_SYNC _IOW('a',23,int) #define MVP_SET_AUD_END_STREAM _IOW('a',25,int) #define MVP_SET_AUD_RESET _IOW('a',26,int) #define MVP_SET_AUD_DAC_CLK _IOW('a',27,int) #define MVP_GET_AUD_REGS _IOW('a',28,aud_ctl_regs_t*) struct mvp_data { struct audio_format audio_format; int fd; }; static const unsigned mvp_sample_rates[][3] = { {9, 8000, 32000}, {10, 11025, 44100}, {11, 12000, 48000}, {1, 16000, 32000}, {2, 22050, 44100}, {3, 24000, 48000}, {5, 32000, 32000}, {0, 44100, 44100}, {7, 48000, 48000}, {13, 64000, 32000}, {14, 88200, 44100}, {15, 96000, 48000} }; /** * The quark used for GError.domain. */ static inline GQuark mvp_output_quark(void) { return g_quark_from_static_string("mvp_output"); } /** * Translate a sample rate to a MVP sample rate. * * @param sample_rate the sample rate in Hz */ static unsigned mvp_find_sample_rate(unsigned sample_rate) { for (unsigned i = 0; i < G_N_ELEMENTS(mvp_sample_rates); ++i) if (mvp_sample_rates[i][1] == sample_rate) return mvp_sample_rates[i][0]; return (unsigned)-1; } static bool mvp_output_test_default_device(void) { int fd; fd = open("/dev/adec_pcm", O_WRONLY); if (fd >= 0) { close(fd); return true; } g_warning("Error opening PCM device \"/dev/adec_pcm\": %s\n", strerror(errno)); return false; } static void * mvp_output_init(G_GNUC_UNUSED const struct audio_format *audio_format, G_GNUC_UNUSED const struct config_param *param, G_GNUC_UNUSED GError **error) { struct mvp_data *md = g_new(struct mvp_data, 1); md->fd = -1; return md; } static void mvp_output_finish(void *data) { struct mvp_data *md = data; g_free(md); } static bool mvp_set_pcm_params(struct mvp_data *md, struct audio_format *audio_format, GError **error) { unsigned mix[5]; switch (audio_format->channels) { case 1: mix[0] = 1; break; case 2: mix[0] = 0; break; default: g_debug("unsupported channel count %u - falling back to stereo", audio_format->channels); audio_format->channels = 2; mix[0] = 0; break; } /* 0,1=24bit(24) , 2,3=16bit */ switch (audio_format->bits) { case 16: mix[1] = 2; break; case 24: mix[1] = 0; break; default: g_debug("unsupported sample format %u - falling back to stereo", audio_format->bits); audio_format->bits = 16; mix[1] = 2; break; } mix[3] = 0; /* stream type? */ mix[4] = G_BYTE_ORDER == G_LITTLE_ENDIAN; /* * if there is an exact match for the frequency, use it. */ mix[2] = mvp_find_sample_rate(audio_format->sample_rate); if (mix[2] == (unsigned)-1) { g_set_error(error, mvp_output_quark(), 0, "Can not find suitable output frequency for %u", audio_format->sample_rate); return false; } if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) { g_set_error(error, mvp_output_quark(), errno, "Can not set audio format"); return false; } if (ioctl(md->fd, MVP_SET_AUD_SYNC, 2) != 0) { g_set_error(error, mvp_output_quark(), errno, "Can not set audio sync"); return false; } if (ioctl(md->fd, MVP_SET_AUD_PLAY, 0) < 0) { g_set_error(error, mvp_output_quark(), errno, "Can not set audio play mode"); return false; } return true; } static bool mvp_output_open(void *data, struct audio_format *audio_format, GError **error) { struct mvp_data *md = data; long long int stc = 0; int mix[5] = { 0, 2, 7, 1, 0 }; bool success; if ((md->fd = open("/dev/adec_pcm", O_RDWR | O_NONBLOCK)) < 0) { g_set_error(error, mvp_output_quark(), errno, "Error opening /dev/adec_pcm: %s", strerror(errno)); return false; } if (ioctl(md->fd, MVP_SET_AUD_SRC, 1) < 0) { g_set_error(error, mvp_output_quark(), errno, "Error setting audio source: %s", strerror(errno)); return false; } if (ioctl(md->fd, MVP_SET_AUD_STREAMTYPE, 0) < 0) { g_set_error(error, mvp_output_quark(), errno, "Error setting audio streamtype: %s", strerror(errno)); return false; } if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) { g_set_error(error, mvp_output_quark(), errno, "Error setting audio format: %s", strerror(errno)); return false; } ioctl(md->fd, MVP_SET_AUD_STC, &stc); if (ioctl(md->fd, MVP_SET_AUD_BYPASS, 1) < 0) { g_set_error(error, mvp_output_quark(), errno, "Error setting audio streamtype: %s", strerror(errno)); return false; } success = mvp_set_pcm_params(md, audio_format, error); if (!success) return false; md->audio_format = *audio_format; return true; } static void mvp_output_close(void *data) { struct mvp_data *md = data; if (md->fd >= 0) close(md->fd); md->fd = -1; } static void mvp_output_cancel(void *data) { struct mvp_data *md = data; if (md->fd >= 0) { ioctl(md->fd, MVP_SET_AUD_RESET, 0x11); close(md->fd); md->fd = -1; } } static size_t mvp_output_play(void *data, const void *chunk, size_t size, GError **error) { struct mvp_data *md = data; ssize_t ret; /* reopen the device since it was closed by dropBufferedAudio */ if (md->fd < 0) { bool success; success = mvp_output_open(md, &md->audio_format, error); if (!success) return 0; } while (true) { ret = write(md->fd, chunk, size); if (ret > 0) return (size_t)ret; if (ret < 0) { if (errno == EINTR) continue; g_set_error(error, mvp_output_quark(), errno, "Failed to write: %s", strerror(errno)); return 0; } } } const struct audio_output_plugin mvp_output_plugin = { .name = "mvp", .test_default_device = mvp_output_test_default_device, .init = mvp_output_init, .finish = mvp_output_finish, .open = mvp_output_open, .close = mvp_output_close, .play = mvp_output_play, .cancel = mvp_output_cancel, };