Add 5.1 surround sound support

src/audio/alsa.c

@@ -20,30 +20,49 @@
 #include "../audio.h"
 
 #include <stdio.h>
-#include <opus.h>
+#include <opus_multistream.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 MAX_CHANNEL_COUNT 6
-#define CHANNEL_COUNT 2
 #define FRAME_SIZE 240
 
 const char* audio_device = "sysdefault";
 
 static snd_pcm_t *handle;
-static OpusDecoder* decoder;
+static OpusMSDecoder* decoder;
-static short pcmBuffer[FRAME_SIZE * CHANNEL_COUNT];
+static short pcmBuffer[FRAME_SIZE * MAX_CHANNEL_COUNT];
 
-static void alsa_renderer_init() {
+static void alsa_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig) {
   int rc;
-  decoder = opus_decoder_create(SAMPLE_RATE, CHANNEL_COUNT, &rc);
+  unsigned char alsaMapping[6];
+
+  /* The supplied mapping array has order: FL-FR-C-LFE-RL-RR
+   * ALSA expects the order: FL-FR-RL-RR-C-LFE
+   * We need copy the mapping locally and swap the channels around.
+   */
+  alsaMapping[0] = opusConfig->mapping[0];
+  alsaMapping[1] = opusConfig->mapping[1];
+  if (opusConfig->channelCount == 6) {
+    alsaMapping[2] = opusConfig->mapping[4];
+    alsaMapping[3] = opusConfig->mapping[5];
+    alsaMapping[4] = opusConfig->mapping[2];
+    alsaMapping[5] = opusConfig->mapping[3];
+  }
+
+  decoder = opus_multistream_decoder_create(opusConfig->sampleRate,
+                                            opusConfig->channelCount,
+                                            opusConfig->streams,
+                                            opusConfig->coupledStreams,
+                                            alsaMapping,
+                                            &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 period_size = FRAME_SIZE * opusConfig->channelCount * 2;
   snd_pcm_uframes_t buffer_size = 12 * period_size;
-  unsigned int sampleRate = SAMPLE_RATE;
+  unsigned int sampleRate = opusConfig->sampleRate;
 
   /* Open PCM device for playback. */
   CHECK_RETURN(snd_pcm_open(&handle, audio_device, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK))
@@ -54,7 +73,7 @@ static void alsa_renderer_init() {
   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_channels(handle, hw_params, opusConfig->channelCount));
   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));
@@ -73,7 +92,7 @@ static void alsa_renderer_init() {
 
 static void alsa_renderer_cleanup() {
   if (decoder != NULL)
-    opus_decoder_destroy(decoder);
+    opus_multistream_decoder_destroy(decoder);
 
   if (handle != NULL) {
     snd_pcm_drain(handle);
@@ -82,7 +101,7 @@ static void alsa_renderer_cleanup() {
 }
 
 static void alsa_renderer_decode_and_play_sample(char* data, int length) {
-  int decodeLen = opus_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
+  int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
   if (decodeLen > 0) {
     int rc = snd_pcm_writei(handle, pcmBuffer, decodeLen);
     if (rc == -EPIPE)

src/audio/sdl.c

@@ -23,27 +23,34 @@
 #include <SDL_audio.h>
 
 #include <stdio.h>
-#include <opus.h>
+#include <opus_multistream.h>
 
-#define SAMPLE_RATE 48000
+#define MAX_CHANNEL_COUNT 6
-#define CHANNEL_COUNT 2
 #define FRAME_SIZE 240
 
-static OpusDecoder* decoder;
+static OpusMSDecoder* decoder;
-static short pcmBuffer[FRAME_SIZE * CHANNEL_COUNT];
+static short pcmBuffer[FRAME_SIZE * MAX_CHANNEL_COUNT];
 static SDL_AudioDeviceID dev;
+static int channelCount;
 
-static void sdl_renderer_init() {
+static void sdl_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig) {
   int rc;
-  decoder = opus_decoder_create(SAMPLE_RATE, CHANNEL_COUNT, &rc);
+  decoder = opus_multistream_decoder_create(opusConfig->sampleRate,
+                                            opusConfig->channelCount,
+                                            opusConfig->streams,
+                                            opusConfig->coupledStreams,
+                                            opusConfig->mapping,
+                                            &rc);
+
+  channelCount = opusConfig->channelCount;
 
   SDL_InitSubSystem(SDL_INIT_AUDIO);
 
   SDL_AudioSpec want, have;
   SDL_zero(want);
-  want.freq = SAMPLE_RATE;
+  want.freq = opusConfig->sampleRate;
   want.format = AUDIO_S16LSB;
-  want.channels = CHANNEL_COUNT;
+  want.channels = opusConfig->channelCount;
   want.samples = 4096;
 
   dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, SDL_AUDIO_ALLOW_FORMAT_CHANGE);
@@ -58,15 +65,15 @@ static void sdl_renderer_init() {
 
 static void sdl_renderer_cleanup() {
   if (decoder != NULL)
-    opus_decoder_destroy(decoder);
+    opus_multistream_decoder_destroy(decoder);
 
   SDL_CloseAudioDevice(dev);
 }
 
 static void sdl_renderer_decode_and_play_sample(char* data, int length) {
-  int decodeLen = opus_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
+  int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
   if (decodeLen > 0) {
-    SDL_QueueAudio(dev, pcmBuffer, decodeLen * CHANNEL_COUNT * sizeof(short));
+    SDL_QueueAudio(dev, pcmBuffer, decodeLen * channelCount * sizeof(short));
   } else {
     printf("Opus error from decode: %d\n", decodeLen);
   }

src/config.c

@@ -60,6 +60,7 @@ static struct option long_options[] = {
   {"keydir", required_argument, NULL, 'r'},
   {"remote", no_argument, NULL, 's'},
   {"fullscreen", no_argument, NULL, 't'},
+  {"surround", no_argument, NULL, 'u'},
   {0, 0, 0, 0},
 };
 
@@ -187,6 +188,10 @@ static void parse_argument(int c, char* value, PCONFIGURATION config) {
     break;
   case 't':
     config->fullscreen = true;
+    break;
+  case 'u':
+    config->stream.audioConfiguration = AUDIO_CONFIGURATION_51_SURROUND;
+    break;
   case 1:
     if (config->action == NULL)
       config->action = value;
@@ -259,12 +264,15 @@ void config_save(char* filename, PCONFIGURATION config) {
 }
 
 void config_parse(int argc, char* argv[], PCONFIGURATION config) {
+  LiInitializeStreamConfiguration(&config->stream);
+
   config->stream.width = 1280;
   config->stream.height = 720;
   config->stream.fps = 60;
   config->stream.bitrate = -1;
   config->stream.packetSize = 1024;
   config->stream.streamingRemotely = 0;
+  config->stream.audioConfiguration = AUDIO_CONFIGURATION_STEREO;
 
   config->platform = "default";
   config->app = "Steam";

src/main.c

@@ -135,6 +135,7 @@ static void help() {
   printf("\t-app <app>\t\tName of app to stream\n");
   printf("\t-nosops\t\t\tDon't allow GFE to modify game settings\n");
   printf("\t-localaudio\t\tPlay audio locally\n");
+  printf("\t-surround\t\tStream 5.1 surround sound (requires GFE 2.7)\n");
   printf("\t-keydir <directory>\tLoad encryption keys from directory\n");
   #ifdef HAVE_SDL
   printf("\n Video options (SDL Only)\n\n");