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Provide selected audio device as context

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This commit is contained in:
Iwan Timmer
2017-05-28 16:06:37 +02:00
parent c2036acca1
commit f5cb2d1880
11 changed files with 17 additions and 16 deletions
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3
src/audio/alsa.c
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@@ -29,7 +29,7 @@ static snd_pcm_t *handle;
static OpusMSDecoder* decoder;
static short pcmBuffer[FRAME_SIZE * MAX_CHANNEL_COUNT];
static int alsa_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig) {
static int alsa_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig, void* context, int arFlags) {
int rc;
unsigned char alsaMapping[MAX_CHANNEL_COUNT];
@@ -54,6 +54,7 @@ static int alsa_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGUR
snd_pcm_uframes_t buffer_size = 2 * period_size;
unsigned int sampleRate = opusConfig->sampleRate;
char* audio_device = (char*) context;
if (audio_device == NULL)
audio_device = "sysdefault";

4
src/audio/audio.h
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@@ -25,8 +25,6 @@
#define FRAME_SIZE 240
#define FRAME_BUFFER 12
extern const char* audio_device;
#ifdef HAVE_ALSA
extern AUDIO_RENDERER_CALLBACKS audio_callbacks_alsa;
#endif
@@ -35,5 +33,5 @@ extern AUDIO_RENDERER_CALLBACKS audio_callbacks_sdl;
#endif
#ifdef HAVE_PULSE
extern AUDIO_RENDERER_CALLBACKS audio_callbacks_pulse;
bool audio_pulse_init();
bool audio_pulse_init(char* audio_device);
#endif

2
src/audio/omx.c
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@@ -32,7 +32,7 @@ static OMX_BUFFERHEADERTYPE *buf;
static short pcmBuffer[FRAME_SIZE * MAX_CHANNEL_COUNT];
static int channelCount;
static int omx_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig) {
static int omx_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig, void* context, int arFlags) {
int rc, error;
OMX_ERRORTYPE err;
unsigned char omxMapping[MAX_CHANNEL_COUNT];

5
src/audio/pulse.c
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@@ -31,7 +31,7 @@ static pa_simple *dev = NULL;
static short pcmBuffer[FRAME_SIZE * MAX_CHANNEL_COUNT];
static int channelCount;
bool audio_pulse_init() {
bool audio_pulse_init(char* audio_device) {
pa_sample_spec spec = {
.format = PA_SAMPLE_S16LE,
.rate = 44000,
@@ -47,7 +47,7 @@ bool audio_pulse_init() {
return (bool) dev;
}
static int pulse_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig) {
static int pulse_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig, void* context, int arFlags) {
int rc, error;
unsigned char alsaMapping[MAX_CHANNEL_COUNT];
@@ -74,6 +74,7 @@ static int pulse_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGU
.channels = opusConfig->channelCount
};
char* audio_device = (char*) context;
dev = pa_simple_new(audio_device, "Moonlight Embedded", PA_STREAM_PLAYBACK, NULL, "Streaming", &spec, NULL, NULL, &error);
if (!dev) {

2
src/audio/sdl.c
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@@ -30,7 +30,7 @@ static short pcmBuffer[FRAME_SIZE * MAX_CHANNEL_COUNT];
static SDL_AudioDeviceID dev;
static int channelCount;
static int sdl_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig) {
static int sdl_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);