moonlight-stream/moonlight-embedded
1
0
Fork 0
mirror of https://github.com/moonlight-stream/moonlight-embedded.git synced 2026-04-24 00:56:42 +00:00
Code Issues Packages Projects Releases Wiki Activity

Move audio code to seperate directory

Browse Source
This commit is contained in:
Iwan Timmer
2015-07-12 21:31:53 +02:00
parent f167c6febb
commit 2ba4840569
5 changed files with 20 additions and 10 deletions
Download Patch File Download Diff File Expand all files Collapse all files

104
src/audio/alsa.c Normal file
View File

@@ -0,0 +1,104 @@
/*
* This file is part of Moonlight Embedded.
*
* Copyright (C) 2015 Iwan Timmer
*
* Moonlight 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 3 of the License, or
* (at your option) any later version.
*
* Moonlight 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 Moonlight; if not, see <http://www.gnu.org/licenses/>.
*/
#include "../audio.h"
#include <stdio.h>
#include <opus.h>
#include <alsa/asoundlib.h>
#define CHECK_RETURN(f) if ((rc = f) < 0) { printf("Alsa error code %d\n", rc); exit(-1); }
#define SAMPLE_RATE 48000
#define CHANNEL_COUNT 2
#define FRAME_SIZE 240
const char* audio_device = "sysdefault";
static snd_pcm_t *handle;
static OpusDecoder* decoder;
static short pcmBuffer[FRAME_SIZE * CHANNEL_COUNT];
static void alsa_renderer_init() {
int rc;
decoder = opus_decoder_create(SAMPLE_RATE, CHANNEL_COUNT, &rc);
snd_pcm_hw_params_t *hw_params;
snd_pcm_sw_params_t *sw_params;
snd_pcm_uframes_t period_size = FRAME_SIZE * CHANNEL_COUNT * 2;
snd_pcm_uframes_t buffer_size = 12 * period_size;
unsigned int sampleRate = SAMPLE_RATE;
/* Open PCM device for playback. */
CHECK_RETURN(snd_pcm_open(&handle, audio_device, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK))
/* Set hardware parameters */
CHECK_RETURN(snd_pcm_hw_params_malloc(&hw_params));
CHECK_RETURN(snd_pcm_hw_params_any(handle, hw_params));
CHECK_RETURN(snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED));
CHECK_RETURN(snd_pcm_hw_params_set_format(handle, hw_params, SND_PCM_FORMAT_S16_LE));
CHECK_RETURN(snd_pcm_hw_params_set_rate_near(handle, hw_params, &sampleRate, NULL));
CHECK_RETURN(snd_pcm_hw_params_set_channels(handle, hw_params, CHANNEL_COUNT));
CHECK_RETURN(snd_pcm_hw_params_set_buffer_size_near(handle, hw_params, &buffer_size));
CHECK_RETURN(snd_pcm_hw_params_set_period_size_near(handle, hw_params, &period_size, NULL));
CHECK_RETURN(snd_pcm_hw_params(handle, hw_params));
snd_pcm_hw_params_free(hw_params);
/* Set software parameters */
CHECK_RETURN(snd_pcm_sw_params_malloc(&sw_params));
CHECK_RETURN(snd_pcm_sw_params_current(handle, sw_params));
CHECK_RETURN(snd_pcm_sw_params_set_start_threshold(handle, sw_params, buffer_size - period_size));
CHECK_RETURN(snd_pcm_sw_params_set_avail_min(handle, sw_params, period_size));
CHECK_RETURN(snd_pcm_sw_params(handle, sw_params));
snd_pcm_sw_params_free(sw_params);
CHECK_RETURN(snd_pcm_prepare(handle));
}
static void alsa_renderer_cleanup() {
if (decoder != NULL)
opus_decoder_destroy(decoder);
if (handle != NULL) {
snd_pcm_drain(handle);
snd_pcm_close(handle);
}
}
static void alsa_renderer_decode_and_play_sample(char* data, int length) {
int decodeLen = opus_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
if (decodeLen > 0) {
int rc = snd_pcm_writei(handle, pcmBuffer, decodeLen);
if (rc == -EPIPE)
snd_pcm_recover(handle, rc, 1);
if (rc<0)
printf("Alsa error from writei: %d\n", rc);
else if (decodeLen != rc)
printf("Alsa shortm write, write %d frames\n", rc);
} else {
printf("Opus error from decode: %d\n", decodeLen);
}
}
AUDIO_RENDERER_CALLBACKS audio_callbacks_alsa = {
.init = alsa_renderer_init,
.cleanup = alsa_renderer_cleanup,
.decodeAndPlaySample = alsa_renderer_decode_and_play_sample,
};