0aee49bdf8
This updates the copyright header to all be the same, which is pretty much an update of where to mail request for a copy of the GPL and the years of the MPD project. This also puts all committers under 'The Music Player Project' umbrella. These entries should go individually in the AUTHORS file, for consistancy.
332 lines
7.6 KiB
C
332 lines
7.6 KiB
C
/*
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* Copyright (C) 2003-2009 The Music Player Daemon Project
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* http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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/*
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* Media MVP audio output based on code from MVPMC project:
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* http://mvpmc.sourceforge.net/
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*/
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#include "../output_api.h"
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#include <glib.h>
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#include <sys/types.h>
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#include <sys/stat.h>
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#include <sys/ioctl.h>
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#include <fcntl.h>
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#include <errno.h>
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#include <unistd.h>
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#include <stdlib.h>
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#undef G_LOG_DOMAIN
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#define G_LOG_DOMAIN "mvp"
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typedef struct {
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unsigned long dsp_status;
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unsigned long stream_decode_type;
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unsigned long sample_rate;
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unsigned long bit_rate;
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unsigned long raw[64 / sizeof(unsigned long)];
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} aud_status_t;
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#define MVP_SET_AUD_STOP _IOW('a',1,int)
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#define MVP_SET_AUD_PLAY _IOW('a',2,int)
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#define MVP_SET_AUD_PAUSE _IOW('a',3,int)
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#define MVP_SET_AUD_UNPAUSE _IOW('a',4,int)
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#define MVP_SET_AUD_SRC _IOW('a',5,int)
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#define MVP_SET_AUD_MUTE _IOW('a',6,int)
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#define MVP_SET_AUD_BYPASS _IOW('a',8,int)
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#define MVP_SET_AUD_CHANNEL _IOW('a',9,int)
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#define MVP_GET_AUD_STATUS _IOR('a',10,aud_status_t)
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#define MVP_SET_AUD_VOLUME _IOW('a',13,int)
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#define MVP_GET_AUD_VOLUME _IOR('a',14,int)
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#define MVP_SET_AUD_STREAMTYPE _IOW('a',15,int)
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#define MVP_SET_AUD_FORMAT _IOW('a',16,int)
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#define MVP_GET_AUD_SYNC _IOR('a',21,pts_sync_data_t*)
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#define MVP_SET_AUD_STC _IOW('a',22,long long int *)
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#define MVP_SET_AUD_SYNC _IOW('a',23,int)
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#define MVP_SET_AUD_END_STREAM _IOW('a',25,int)
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#define MVP_SET_AUD_RESET _IOW('a',26,int)
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#define MVP_SET_AUD_DAC_CLK _IOW('a',27,int)
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#define MVP_GET_AUD_REGS _IOW('a',28,aud_ctl_regs_t*)
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struct mvp_data {
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struct audio_format audio_format;
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int fd;
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};
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static const unsigned mvp_sample_rates[][3] = {
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{9, 8000, 32000},
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{10, 11025, 44100},
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{11, 12000, 48000},
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{1, 16000, 32000},
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{2, 22050, 44100},
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{3, 24000, 48000},
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{5, 32000, 32000},
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{0, 44100, 44100},
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{7, 48000, 48000},
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{13, 64000, 32000},
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{14, 88200, 44100},
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{15, 96000, 48000}
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};
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/**
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* The quark used for GError.domain.
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*/
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static inline GQuark
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mvp_output_quark(void)
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{
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return g_quark_from_static_string("mvp_output");
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}
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/**
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* Translate a sample rate to a MVP sample rate.
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*
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* @param sample_rate the sample rate in Hz
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*/
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static unsigned
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mvp_find_sample_rate(unsigned sample_rate)
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{
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for (unsigned i = 0; i < G_N_ELEMENTS(mvp_sample_rates); ++i)
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if (mvp_sample_rates[i][1] == sample_rate)
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return mvp_sample_rates[i][0];
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return (unsigned)-1;
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}
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static bool
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mvp_output_test_default_device(void)
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{
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int fd;
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fd = open("/dev/adec_pcm", O_WRONLY);
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if (fd >= 0) {
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close(fd);
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return true;
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}
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g_warning("Error opening PCM device \"/dev/adec_pcm\": %s\n",
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strerror(errno));
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return false;
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}
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static void *
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mvp_output_init(G_GNUC_UNUSED const struct audio_format *audio_format,
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G_GNUC_UNUSED const struct config_param *param,
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G_GNUC_UNUSED GError **error)
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{
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struct mvp_data *md = g_new(struct mvp_data, 1);
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md->fd = -1;
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return md;
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}
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static void
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mvp_output_finish(void *data)
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{
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struct mvp_data *md = data;
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g_free(md);
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}
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static bool
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mvp_set_pcm_params(struct mvp_data *md, struct audio_format *audio_format,
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GError **error)
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{
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unsigned mix[5];
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switch (audio_format->channels) {
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case 1:
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mix[0] = 1;
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break;
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case 2:
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mix[0] = 0;
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break;
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default:
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g_debug("unsupported channel count %u - falling back to stereo",
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audio_format->channels);
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audio_format->channels = 2;
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mix[0] = 0;
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break;
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}
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/* 0,1=24bit(24) , 2,3=16bit */
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switch (audio_format->bits) {
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case 16:
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mix[1] = 2;
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break;
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case 24:
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mix[1] = 0;
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break;
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default:
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g_debug("unsupported sample format %u - falling back to stereo",
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audio_format->bits);
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audio_format->bits = 16;
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mix[1] = 2;
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break;
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}
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mix[3] = 0; /* stream type? */
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mix[4] = G_BYTE_ORDER == G_LITTLE_ENDIAN;
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/*
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* if there is an exact match for the frequency, use it.
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*/
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mix[2] = mvp_find_sample_rate(audio_format->sample_rate);
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if (mix[2] == (unsigned)-1) {
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g_set_error(error, mvp_output_quark(), 0,
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"Can not find suitable output frequency for %u",
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audio_format->sample_rate);
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return false;
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}
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if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) {
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g_set_error(error, mvp_output_quark(), errno,
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"Can not set audio format");
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return false;
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}
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if (ioctl(md->fd, MVP_SET_AUD_SYNC, 2) != 0) {
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g_set_error(error, mvp_output_quark(), errno,
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"Can not set audio sync");
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return false;
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}
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if (ioctl(md->fd, MVP_SET_AUD_PLAY, 0) < 0) {
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g_set_error(error, mvp_output_quark(), errno,
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"Can not set audio play mode");
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return false;
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}
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return true;
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}
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static bool
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mvp_output_open(void *data, struct audio_format *audio_format, GError **error)
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{
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struct mvp_data *md = data;
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long long int stc = 0;
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int mix[5] = { 0, 2, 7, 1, 0 };
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bool success;
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if ((md->fd = open("/dev/adec_pcm", O_RDWR | O_NONBLOCK)) < 0) {
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g_set_error(error, mvp_output_quark(), errno,
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"Error opening /dev/adec_pcm: %s",
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strerror(errno));
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return false;
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}
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if (ioctl(md->fd, MVP_SET_AUD_SRC, 1) < 0) {
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g_set_error(error, mvp_output_quark(), errno,
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"Error setting audio source: %s",
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strerror(errno));
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return false;
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}
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if (ioctl(md->fd, MVP_SET_AUD_STREAMTYPE, 0) < 0) {
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g_set_error(error, mvp_output_quark(), errno,
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"Error setting audio streamtype: %s",
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strerror(errno));
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return false;
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}
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if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) {
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g_set_error(error, mvp_output_quark(), errno,
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"Error setting audio format: %s",
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strerror(errno));
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return false;
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}
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ioctl(md->fd, MVP_SET_AUD_STC, &stc);
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if (ioctl(md->fd, MVP_SET_AUD_BYPASS, 1) < 0) {
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g_set_error(error, mvp_output_quark(), errno,
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"Error setting audio streamtype: %s",
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strerror(errno));
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return false;
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}
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success = mvp_set_pcm_params(md, audio_format, error);
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if (!success)
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return false;
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md->audio_format = *audio_format;
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return true;
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}
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static void mvp_output_close(void *data)
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{
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struct mvp_data *md = data;
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if (md->fd >= 0)
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close(md->fd);
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md->fd = -1;
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}
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static void mvp_output_cancel(void *data)
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{
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struct mvp_data *md = data;
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if (md->fd >= 0) {
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ioctl(md->fd, MVP_SET_AUD_RESET, 0x11);
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close(md->fd);
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md->fd = -1;
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}
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}
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static size_t
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mvp_output_play(void *data, const void *chunk, size_t size, GError **error)
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{
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struct mvp_data *md = data;
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ssize_t ret;
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/* reopen the device since it was closed by dropBufferedAudio */
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if (md->fd < 0) {
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bool success;
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success = mvp_output_open(md, &md->audio_format, error);
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if (!success)
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return 0;
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}
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while (true) {
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ret = write(md->fd, chunk, size);
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if (ret > 0)
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return (size_t)ret;
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if (ret < 0) {
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if (errno == EINTR)
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continue;
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g_set_error(error, mvp_output_quark(), errno,
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"Failed to write: %s", strerror(errno));
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return 0;
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}
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}
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}
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const struct audio_output_plugin mvp_output_plugin = {
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.name = "mvp",
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.test_default_device = mvp_output_test_default_device,
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.init = mvp_output_init,
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.finish = mvp_output_finish,
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.open = mvp_output_open,
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.close = mvp_output_close,
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.play = mvp_output_play,
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.cancel = mvp_output_cancel,
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};
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