mirror of
https://github.com/moonlight-stream/moonlight-embedded.git
synced 2026-04-06 07:56:22 +00:00
126 lines
4.7 KiB
C
126 lines
4.7 KiB
C
/*
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* This file is part of Moonlight Embedded.
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*
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* Copyright (C) 2015, 2016 Iwan Timmer
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*
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* Moonlight 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 3 of the License, or
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* (at your option) any later version.
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*
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* Moonlight 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
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* along with Moonlight; if not, see <http://www.gnu.org/licenses/>.
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*/
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#include "../audio.h"
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#include <stdio.h>
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#include <opus_multistream.h>
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#include <alsa/asoundlib.h>
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#define CHECK_RETURN(f) if ((rc = f) < 0) { printf("Alsa error code %d\n", rc); exit(-1); }
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#define MAX_CHANNEL_COUNT 6
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#define FRAME_SIZE 240
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static snd_pcm_t *handle;
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static OpusMSDecoder* decoder;
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static short pcmBuffer[FRAME_SIZE * MAX_CHANNEL_COUNT];
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static void alsa_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig) {
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int rc;
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unsigned char alsaMapping[6];
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/* The supplied mapping array has order: FL-FR-C-LFE-RL-RR
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* ALSA expects the order: FL-FR-RL-RR-C-LFE
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* We need copy the mapping locally and swap the channels around.
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*/
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alsaMapping[0] = opusConfig->mapping[0];
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alsaMapping[1] = opusConfig->mapping[1];
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if (opusConfig->channelCount == 6) {
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alsaMapping[2] = opusConfig->mapping[4];
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alsaMapping[3] = opusConfig->mapping[5];
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alsaMapping[4] = opusConfig->mapping[2];
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alsaMapping[5] = opusConfig->mapping[3];
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}
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decoder = opus_multistream_decoder_create(opusConfig->sampleRate,
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opusConfig->channelCount,
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opusConfig->streams,
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opusConfig->coupledStreams,
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alsaMapping,
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&rc);
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snd_pcm_hw_params_t *hw_params;
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snd_pcm_sw_params_t *sw_params;
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snd_pcm_uframes_t period_size = FRAME_SIZE * opusConfig->channelCount * 2;
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snd_pcm_uframes_t buffer_size = 12 * period_size;
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unsigned int sampleRate = opusConfig->sampleRate;
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if (audio_device == NULL)
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audio_device = "sysdefault";
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/* Open PCM device for playback. */
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CHECK_RETURN(snd_pcm_open(&handle, audio_device, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK))
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/* Set hardware parameters */
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CHECK_RETURN(snd_pcm_hw_params_malloc(&hw_params));
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CHECK_RETURN(snd_pcm_hw_params_any(handle, hw_params));
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CHECK_RETURN(snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED));
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CHECK_RETURN(snd_pcm_hw_params_set_format(handle, hw_params, SND_PCM_FORMAT_S16_LE));
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CHECK_RETURN(snd_pcm_hw_params_set_rate_near(handle, hw_params, &sampleRate, NULL));
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CHECK_RETURN(snd_pcm_hw_params_set_channels(handle, hw_params, opusConfig->channelCount));
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CHECK_RETURN(snd_pcm_hw_params_set_buffer_size_near(handle, hw_params, &buffer_size));
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CHECK_RETURN(snd_pcm_hw_params_set_period_size_near(handle, hw_params, &period_size, NULL));
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CHECK_RETURN(snd_pcm_hw_params(handle, hw_params));
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snd_pcm_hw_params_free(hw_params);
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/* Set software parameters */
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CHECK_RETURN(snd_pcm_sw_params_malloc(&sw_params));
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CHECK_RETURN(snd_pcm_sw_params_current(handle, sw_params));
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CHECK_RETURN(snd_pcm_sw_params_set_start_threshold(handle, sw_params, buffer_size - period_size));
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CHECK_RETURN(snd_pcm_sw_params_set_avail_min(handle, sw_params, period_size));
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CHECK_RETURN(snd_pcm_sw_params(handle, sw_params));
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snd_pcm_sw_params_free(sw_params);
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CHECK_RETURN(snd_pcm_prepare(handle));
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}
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static void alsa_renderer_cleanup() {
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if (decoder != NULL)
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opus_multistream_decoder_destroy(decoder);
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if (handle != NULL) {
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snd_pcm_drain(handle);
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snd_pcm_close(handle);
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}
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}
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static void alsa_renderer_decode_and_play_sample(char* data, int length) {
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int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
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if (decodeLen > 0) {
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int rc = snd_pcm_writei(handle, pcmBuffer, decodeLen);
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if (rc == -EPIPE)
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snd_pcm_recover(handle, rc, 1);
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if (rc<0)
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printf("Alsa error from writei: %d\n", rc);
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else if (decodeLen != rc)
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printf("Alsa shortm write, write %d frames\n", rc);
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} else {
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printf("Opus error from decode: %d\n", decodeLen);
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}
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}
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AUDIO_RENDERER_CALLBACKS audio_callbacks_alsa = {
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.init = alsa_renderer_init,
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.cleanup = alsa_renderer_cleanup,
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.decodeAndPlaySample = alsa_renderer_decode_and_play_sample,
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.capabilities = CAPABILITY_DIRECT_SUBMIT,
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};
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