948b8f35e6
Remove trailing whitespace found by this command: find -name '*.[ch]' | xargs grep "[[:space:]]$"
335 lines
7.7 KiB
C
335 lines
7.7 KiB
C
/*
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* Copyright (C) 2003-2010 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 "config.h"
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#include "output_api.h"
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#include "fd_util.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_cloexec("/dev/adec_pcm", O_WRONLY, 0);
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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->format) {
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case SAMPLE_FORMAT_S16:
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mix[1] = 2;
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break;
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case SAMPLE_FORMAT_S24_P32:
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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 %s - falling back to 16 bit",
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sample_format_to_string(audio_format->format));
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audio_format->format = SAMPLE_FORMAT_S16;
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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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md->fd = open_cloexec("/dev/adec_pcm", O_RDWR | O_NONBLOCK, 0);
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if (md->fd < 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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