feat: add oss audio callback for FreeBSD

OSS is the native sound system of the FreeBSD.It will be used on the x11_vaapi platform.It was added to reduce moonlight's dependency on FreeBSD.

CMakeLists.txt

@@ -87,6 +87,13 @@ add_executable(moonlight ${SRC_LIST})
 target_link_libraries(moonlight m)
 target_link_libraries(moonlight gamestream)
 
+if (CMAKE_SYSTEM_NAME MATCHES "FreeBSD")
+  set(ALSA_FOUND FALSE)
+  set(PULSE_FOUND FALSE)
+  set(CEC_FOUND FALSE)
+  target_sources(moonlight PRIVATE ./src/audio/oss.c)
+endif()
+
 if (CEC_FOUND)
   list(APPEND MOONLIGHT_DEFINITIONS HAVE_LIBCEC)
   list(APPEND MOONLIGHT_OPTIONS CEC)

src/audio/audio.h

@@ -31,3 +31,6 @@ extern AUDIO_RENDERER_CALLBACKS audio_callbacks_sdl;
 extern AUDIO_RENDERER_CALLBACKS audio_callbacks_pulse;
 bool audio_pulse_init(char* audio_device);
 #endif
+#ifdef __FreeBSD__
+extern AUDIO_RENDERER_CALLBACKS audio_callbacks_oss;
+#endif

src/audio/oss.c

@@ -0,0 +1,105 @@
+/*
+ * This file is part of Moonlight Embedded.
+ *
+ * Copyright (C) 2015-2017 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/>.
+ */
+
+#ifdef __FreeBSD__
+#include <sys/soundcard.h>
+#include "audio.h"
+
+#include <opus_multistream.h>
+#include <stdio.h>
+#include <sys/ioctl.h>
+
+#include <fcntl.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+static OpusMSDecoder* decoder;
+static short* pcmBuffer;
+static int samplesPerFrame;
+static int channelCount;
+static int fd;
+
+static int oss_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig, void* context, int arFlags) {
+  int rc;
+  decoder = opus_multistream_decoder_create(opusConfig->sampleRate, opusConfig->channelCount, opusConfig->streams, opusConfig->coupledStreams, opusConfig->mapping, &rc);
+
+  channelCount = opusConfig->channelCount;
+  samplesPerFrame = opusConfig->samplesPerFrame;
+  pcmBuffer = malloc(sizeof(short) * channelCount * samplesPerFrame);
+  if (pcmBuffer == NULL)
+    return -1;
+
+  const char* oss_name = "/dev/dsp";
+  fd = open(oss_name, O_WRONLY);
+  // buffer size for fragment ,selector 12 is 4096;11 is 2048;10 is 1024; 13is 8192
+  if (fd == -1) {
+    close(fd);
+    printf("Open audio device /dev/dsp faild!!!");
+    return -1;
+  }
+  int frag = 12;
+  if (ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &frag) == -1)
+    printf("Set framgment for /dev/dsp faild.");
+
+  int format = AFMT_S16_LE;
+  int channels = opusConfig->channelCount;
+  int rate = opusConfig->sampleRate;
+  if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1)
+    printf("Set framgment for /dev/dsp faild.");
+  if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1)
+    printf("Set channels for /dev/dsp faild.");
+  if (ioctl(fd, SNDCTL_DSP_SPEED, &rate) == -1)
+    printf("Set sameple rate for /dev/dsp faild.");
+
+  return 0;
+}
+
+static void oss_renderer_cleanup() {
+  if (decoder != NULL) {
+    opus_multistream_decoder_destroy(decoder);
+    decoder = NULL;
+  }
+
+  if (pcmBuffer != NULL) {
+    free(pcmBuffer);
+    pcmBuffer = NULL;
+  }
+
+  if (fd != 0) {
+    close(fd);
+    fd = 0;
+  }
+}
+
+static void oss_renderer_decode_and_play_sample(char* data, int length) {
+  int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, samplesPerFrame, 0);
+  if (decodeLen > 0) {
+    write(fd, pcmBuffer, decodeLen * channelCount * sizeof(short));
+  } else {
+    printf("Opus error from decode: %d\n", decodeLen);
+  }
+}
+
+AUDIO_RENDERER_CALLBACKS audio_callbacks_oss = {
+  .init = oss_renderer_init,
+  .cleanup = oss_renderer_cleanup,
+  .decodeAndPlaySample = oss_renderer_decode_and_play_sample,
+  .capabilities = CAPABILITY_DIRECT_SUBMIT | CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION,
+};
+#endif

src/platform.c

@@ -202,6 +202,9 @@ AUDIO_RENDERER_CALLBACKS* platform_get_audio(enum platform system, char* audio_d
     #ifdef HAVE_ALSA
     return &audio_callbacks_alsa;
     #endif
+    #ifdef __FreeBSD__
+    return &audio_callbacks_oss;
+    #endif
   }
   return NULL;
 }