Updates pt. 2

audio/drivers/alsa_qsa.c

@@ -52,16 +52,15 @@ static void *alsa_qsa_init(const char *device,
    snd_pcm_channel_params_t params = {0};
    snd_pcm_channel_info_t pi;
    snd_pcm_channel_setup_t setup = {0};
-   alsa_t *alsa;
+   settings_t *settings = config_get_ptr();
+   alsa_t *alsa = (alsa_t*)calloc(1, sizeof(alsa_t));
+   if (!alsa)
+      return NULL;
 
    (void)device;
    (void)rate;
    (void)latency;
 
-   alsa = (alsa_t*)calloc(1, sizeof(alsa_t));
-   if (!alsa)
-      return NULL;
-
    if ((err = snd_pcm_open_preferred(&alsa->pcm, &card, &dev,
                SND_PCM_OPEN_PLAYBACK)) < 0)
    {
@@ -123,8 +122,8 @@ static void *alsa_qsa_init(const char *device,
       goto error;
    }
 
-   if (g_settings.audio.block_frames)
-      alsa->buf_size = g_settings.audio.block_frames * 4;
+   if (settings->audio.block_frames)
+      alsa->buf_size = settings->audio.block_frames * 4;
    else
       alsa->buf_size = next_pow2(32 * latency);
 

audio/drivers/coreaudio.c

@@ -196,6 +196,7 @@ static void *coreaudio_init(const char *device,
    AudioStreamBasicDescription stream_desc = {0};
    static bool session_initialized = false;
    coreaudio_t *dev = NULL;
+   settings_t *settings = config_get_ptr();
 #ifdef OSX_PPC
    ComponentDescription desc = {0};
 #else
@@ -300,7 +301,7 @@ static void *coreaudio_init(const char *device,
 
    RARCH_LOG("[CoreAudio]: Using output sample rate of %.1f Hz\n",
          (float)real_desc.mSampleRate);
-   g_settings.audio.out_rate = real_desc.mSampleRate;
+   settings->audio.out_rate = real_desc.mSampleRate;
 
    /* Set channel layout (fails on iOS). */
 #ifndef IOS
@@ -321,7 +322,7 @@ static void *coreaudio_init(const char *device,
    if (AudioUnitInitialize(dev->dev) != noErr)
       goto error;
 
-   fifo_size = (latency * g_settings.audio.out_rate) / 1000;
+   fifo_size = (latency * settings->audio.out_rate) / 1000;
    fifo_size *= 2 * sizeof(float);
    dev->buffer_size = fifo_size;
 

audio/drivers/gx_audio.c

@@ -83,7 +83,8 @@ static void dma_callback(void)
 static void *gx_audio_init(const char *device,
       unsigned rate, unsigned latency)
 {
-   gx_audio_t *wa = (gx_audio_t*)memalign(32, sizeof(*wa));
+   settings_t *settings = config_get_ptr();
+   gx_audio_t *wa       = (gx_audio_t*)memalign(32, sizeof(*wa));
    if (!wa)
       return NULL;
 
@@ -97,12 +98,12 @@ static void *gx_audio_init(const char *device,
    if (rate < 33000)
    {
       AISetDSPSampleRate(AI_SAMPLERATE_32KHZ);
-      g_settings.audio.out_rate = 32000;
+      settings->audio.out_rate = 32000;
    }
    else
    {
       AISetDSPSampleRate(AI_SAMPLERATE_48KHZ);
-      g_settings.audio.out_rate = 48000;
+      settings->audio.out_rate = 48000;
    }
 
    OSInitThreadQueue(&wa->cond);

audio/drivers/jack.c

@@ -89,9 +89,10 @@ static void shutdown_cb(void *data)
 static int parse_ports(char **dest_ports, const char **jports)
 {
    int i;
-   char *save;
+   char *save = NULL;
    int parsed = 0;
-   const char *con = strtok_r(g_settings.audio.device, ",", &save);
+   settings_t *settings = config_get_ptr();
+   const char *con = strtok_r(settings->audio.device, ",", &save);
 
    if (con)
       dest_ports[parsed++] = strdup(con);
@@ -107,9 +108,10 @@ static int parse_ports(char **dest_ports, const char **jports)
 
 static size_t find_buffersize(jack_t *jd, int latency)
 {
-   int i, buffer_frames, min_buffer_frames, jack_latency = 0;
    jack_latency_range_t range;
-   int frames = latency * g_settings.audio.out_rate / 1000;
+   int i, buffer_frames, min_buffer_frames, jack_latency = 0;
+   settings_t *settings = config_get_ptr();
+   int frames = latency * settings->audio.out_rate / 1000;
 
    for (i = 0; i < 2; i++)
    {
@@ -138,6 +140,7 @@ static void *ja_init(const char *device, unsigned rate, unsigned latency)
    char *dest_ports[2];
    size_t bufsize = 0;
    int parsed = 0;
+   settings_t *settings = config_get_ptr();
    jack_t *jd = (jack_t*)calloc(1, sizeof(jack_t));
 
    if (!jd)
@@ -150,7 +153,7 @@ static void *ja_init(const char *device, unsigned rate, unsigned latency)
    if (jd->client == NULL)
       goto error;
 
-   g_settings.audio.out_rate = jack_get_sample_rate(jd->client);
+   settings->audio.out_rate = jack_get_sample_rate(jd->client);
 
    jack_set_process_callback(jd->client, process_cb, jd);
    jack_on_shutdown(jd->client, shutdown_cb, jd);

audio/drivers/opensl.c

@@ -109,6 +109,7 @@ static void *sl_init(const char *device, unsigned rate, unsigned latency)
    SLDataSink audio_sink    = {0};
    SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {0};
    SLDataLocator_OutputMix loc_outmix              = {0};
+   settings_t *settings = config_get_ptr();
 
    (void)device;
 
@@ -129,8 +130,8 @@ static void *sl_init(const char *device, unsigned rate, unsigned latency)
    GOTO_IF_FAIL(SLEngineItf_CreateOutputMix(sl->engine, &sl->output_mix, 0, NULL, NULL));
    GOTO_IF_FAIL(SLObjectItf_Realize(sl->output_mix, SL_BOOLEAN_FALSE));
 
-   if (g_settings.audio.block_frames)
-      sl->buf_size = g_settings.audio.block_frames * 4;
+   if (settings->audio.block_frames)
+      sl->buf_size = settings->audio.block_frames * 4;
    else
       sl->buf_size = next_pow2(32 * latency);
 

audio/drivers/oss.c

@@ -45,6 +45,7 @@ static void *oss_init(const char *device, unsigned rate, unsigned latency)
 {
    int frags, frag, channels, format, new_rate;
    int *fd = (int*)calloc(1, sizeof(int));
+   settings_t *settings = config_get_ptr();
 
    if (fd == NULL)
       return NULL;
@@ -99,7 +100,7 @@ static void *oss_init(const char *device, unsigned rate, unsigned latency)
    if (new_rate != (int)rate)
    {
       RARCH_WARN("Requested sample rate not supported. Adjusting output rate to %d Hz.\n", new_rate);
-      g_settings.audio.out_rate = new_rate;
+      settings->audio.out_rate = new_rate;
    }
 
    return fd;

audio/drivers/rwebaudio.c

@@ -27,13 +27,14 @@ static void rwebaudio_free(void *data)
 
 static void *rwebaudio_init(const char *device, unsigned rate, unsigned latency)
 {
-   void *data;
+   settings_t *settings = config_get_ptr();
+   void *data           = RWebAudioInit(latency);
+
    (void)device;
    (void)rate;
-   data = RWebAudioInit(latency);
 
    if (data)
-      g_settings.audio.out_rate = RWebAudioSampleRate();
+      settings->audio.out_rate = RWebAudioSampleRate();
    return data;
 }
 

audio/drivers/sdl_audio.c

@@ -69,7 +69,9 @@ static void *sdl_audio_init(const char *device,
    void *tmp;
    SDL_AudioSpec out;
    SDL_AudioSpec spec = {0};
-   sdl_audio_t *sdl = NULL;
+   settings_t *settings = config_get_ptr();
+   sdl_audio_t *sdl     = NULL;
+
    (void)device;
 
    if (SDL_WasInit(0) == 0)
@@ -103,13 +105,14 @@ static void *sdl_audio_init(const char *device,
       free(sdl);
       return 0;
    }
-   g_settings.audio.out_rate = out.freq;
+
+   settings->audio.out_rate = out.freq;
 
    sdl->lock = slock_new();
    sdl->cond = scond_new();
 
    RARCH_LOG("SDL audio: Requested %u ms latency, got %d ms\n", 
-         latency, (int)(out.samples * 4 * 1000 / g_settings.audio.out_rate));
+         latency, (int)(out.samples * 4 * 1000 / settings->audio.out_rate));
 
    /* Create a buffer twice as big as needed and prefill the buffer. */
    bufsize = out.samples * 4 * sizeof(int16_t);

audio/drivers/xenon360_audio.c

@@ -36,6 +36,7 @@ static void *xenon360_audio_init(const char *device,
       unsigned rate, unsigned latency)
 {
    static bool inited = false;
+   settings_t *settings = config_get_ptr();
 
    if (!inited)
    {
@@ -43,7 +44,7 @@ static void *xenon360_audio_init(const char *device,
       inited = true;
    }
 
-   g_settings.audio.out_rate = SOUND_FREQUENCY;
+   settings->audio.out_rate = SOUND_FREQUENCY;
    return calloc(1, sizeof(xenon_audio_t));
 }