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authorMax Kellermann <max@duempel.org>2011-09-17 08:54:50 +0200
committerMax Kellermann <max@duempel.org>2011-09-17 08:54:50 +0200
commit5e22fe488ed7209c6e470e542826da4674e93338 (patch)
tree7fdc279dbf7b5bfde9ff0d23253bf4d7827a7d00 /src/output/mvp_output_plugin.c
parentc666cf1c4494e61286e0e80b6184b4605a4d40d9 (diff)
output: rename plugin source files
Diffstat (limited to 'src/output/mvp_output_plugin.c')
-rw-r--r--src/output/mvp_output_plugin.c334
1 files changed, 334 insertions, 0 deletions
diff --git a/src/output/mvp_output_plugin.c b/src/output/mvp_output_plugin.c
new file mode 100644
index 00000000..be4c8dbc
--- /dev/null
+++ b/src/output/mvp_output_plugin.c
@@ -0,0 +1,334 @@
+/*
+ * Copyright (C) 2003-2011 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 "config.h"
+#include "output_api.h"
+#include "fd_util.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_cloexec("/dev/adec_pcm", O_WRONLY, 0);
+
+ 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->format) {
+ case SAMPLE_FORMAT_S16:
+ mix[1] = 2;
+ break;
+
+ case SAMPLE_FORMAT_S24_P32:
+ mix[1] = 0;
+ break;
+
+ default:
+ g_debug("unsupported sample format %s - falling back to 16 bit",
+ sample_format_to_string(audio_format->format));
+ audio_format->format = SAMPLE_FORMAT_S16;
+ 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;
+
+ md->fd = open_cloexec("/dev/adec_pcm", O_RDWR | O_NONBLOCK, 0);
+ if (md->fd < 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,
+};