moonlight-stream/moonlight-embedded
1
0
Fork 0
mirror of https://github.com/moonlight-stream/moonlight-embedded.git synced 2025-07-01 23:35:47 +00:00
Code Issues Packages Projects Releases Wiki Activity

Add arbitrary audio duration support to all audio backends

Browse Source
This commit is contained in:
Cameron Gutman 2021-07-25 14:01:19 -05:00
parent 4e09dccfc0
commit fb57f3cb4d
5 changed files with 77 additions and 22 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

24
src/audio/alsa.c
View File

@ -29,7 +29,8 @@
static snd_pcm_t *handle;
static OpusMSDecoder* decoder;
static short pcmBuffer[FRAME_SIZE * AUDIO_CONFIGURATION_MAX_CHANNEL_COUNT];
static short* pcmBuffer;
static int samplesPerFrame;
static int alsa_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig, void* context, int arFlags) {
int rc;
@ -47,11 +48,16 @@ static int alsa_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGUR
alsaMapping[5] = opusConfig->mapping[3];
}
samplesPerFrame = opusConfig->samplesPerFrame;
pcmBuffer = malloc(sizeof(short) * opusConfig->channelCount * samplesPerFrame);
if (pcmBuffer == NULL)
return -1;
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 * FRAME_BUFFER;
snd_pcm_uframes_t period_size = samplesPerFrame * FRAME_BUFFER;
snd_pcm_uframes_t buffer_size = 2 * period_size;
unsigned int sampleRate = opusConfig->sampleRate;
@ -88,17 +94,25 @@ static int alsa_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGUR
}
static void alsa_renderer_cleanup() {
if (decoder != NULL)
if (decoder != NULL) {
opus_multistream_decoder_destroy(decoder);
decoder = NULL;
}
if (handle != NULL) {
snd_pcm_drain(handle);
snd_pcm_close(handle);
handle = NULL;
}
if (pcmBuffer != NULL) {
free(pcmBuffer);
pcmBuffer = NULL;
}
}
static void alsa_renderer_decode_and_play_sample(char* data, int length) {
int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, samplesPerFrame, 0);
if (decodeLen > 0) {
int rc = snd_pcm_writei(handle, pcmBuffer, decodeLen);
if (rc == -EPIPE)
@ -117,5 +131,5 @@ AUDIO_RENDERER_CALLBACKS audio_callbacks_alsa = {
.init = alsa_renderer_init,
.cleanup = alsa_renderer_cleanup,
.decodeAndPlaySample = alsa_renderer_decode_and_play_sample,
.capabilities = CAPABILITY_DIRECT_SUBMIT,
.capabilities = CAPABILITY_DIRECT_SUBMIT | CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION,
};

1
src/audio/audio.h
View File

@ -21,7 +21,6 @@
#include <Limelight.h>
#define FRAME_SIZE 240
#define FRAME_BUFFER 12
#ifdef HAVE_ALSA

21
src/audio/omx.c
View File

@ -29,8 +29,9 @@ static OpusMSDecoder* decoder;
ILCLIENT_T* handle;
COMPONENT_T* component;
static OMX_BUFFERHEADERTYPE *buf;
static short pcmBuffer[FRAME_SIZE * AUDIO_CONFIGURATION_MAX_CHANNEL_COUNT];
static short* pcmBuffer;
static int channelCount;
static int samplesPerFrame;
static int omx_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig, void* context, int arFlags) {
int rc, error;
@ -39,6 +40,11 @@ static int omx_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURA
char* componentName = "audio_render";
channelCount = opusConfig->channelCount;
samplesPerFrame = opusConfig->samplesPerFrame;
pcmBuffer = malloc(sizeof(short) * channelCount * samplesPerFrame);
if (pcmBuffer == NULL)
return -1;
/* The supplied mapping array has order: FL-FR-C-LFE-RL-RR-SL-SR
* OMX expects the order: FL-FR-LFE-C-RL-RR-SL-SR
* We need copy the mapping locally and swap the channels around.
@ -155,8 +161,10 @@ static int omx_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURA
}
static void omx_renderer_cleanup() {
if (decoder != NULL)
if (decoder != NULL) {
opus_multistream_decoder_destroy(decoder);
decoder = NULL;
}
if (handle != NULL) {
if((buf = ilclient_get_input_buffer(component, 100, 1)) == NULL){
fprintf(stderr, "Can't get audio buffer\n");
@ -174,11 +182,16 @@ static void omx_renderer_cleanup() {
ilclient_disable_port_buffers(component, 100, NULL, NULL, NULL);
ilclient_change_component_state(component, OMX_StateIdle);
ilclient_change_component_state(component, OMX_StateLoaded);
handle = NULL;
}
if (pcmBuffer != NULL) {
free(pcmBuffer);
pcmBuffer = NULL;
}
}
static void omx_renderer_decode_and_play_sample(char* data, int length) {
int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, samplesPerFrame, 0);
if (decodeLen > 0) {
buf = ilclient_get_input_buffer(component, 100, 1);
buf->nOffset = 0;
@ -199,5 +212,5 @@ AUDIO_RENDERER_CALLBACKS audio_callbacks_omx = {
.init = omx_renderer_init,
.cleanup = omx_renderer_cleanup,
.decodeAndPlaySample = omx_renderer_decode_and_play_sample,
.capabilities = CAPABILITY_DIRECT_SUBMIT,
.capabilities = CAPABILITY_DIRECT_SUBMIT | CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION,
};

24
src/audio/pulse.c
View File

@ -29,7 +29,8 @@
static OpusMSDecoder* decoder;
static pa_simple *dev = NULL;
static short pcmBuffer[FRAME_SIZE * AUDIO_CONFIGURATION_MAX_CHANNEL_COUNT];
static short* pcmBuffer;
static int samplesPerFrame;
static int channelCount;
bool audio_pulse_init(char* audio_device) {
@ -53,6 +54,10 @@ static int pulse_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGU
unsigned char alsaMapping[AUDIO_CONFIGURATION_MAX_CHANNEL_COUNT];
channelCount = opusConfig->channelCount;
samplesPerFrame = opusConfig->samplesPerFrame;
pcmBuffer = malloc(sizeof(short) * channelCount * samplesPerFrame);
if (pcmBuffer == NULL)
return -1;
/* The supplied mapping array has order: FL-FR-C-LFE-RL-RR-SL-SR
* ALSA expects the order: FL-FR-RL-RR-C-LFE-SL-SR
@ -86,7 +91,7 @@ static int pulse_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGU
}
static void pulse_renderer_decode_and_play_sample(char* data, int length) {
int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, samplesPerFrame, 0);
if (decodeLen > 0) {
int error;
int rc = pa_simple_write(dev, pcmBuffer, decodeLen * sizeof(short) * channelCount, &error);
@ -99,12 +104,23 @@ static void pulse_renderer_decode_and_play_sample(char* data, int length) {
}
static void pulse_renderer_cleanup() {
pa_simple_free(dev);
if (decoder != NULL) {
opus_multistream_decoder_destroy(decoder);
decoder = NULL;
}
if (dev != NULL) {
pa_simple_free(dev);
dev = NULL;
}
if (pcmBuffer != NULL) {
free(pcmBuffer);
pcmBuffer = NULL;
}
}
AUDIO_RENDERER_CALLBACKS audio_callbacks_pulse = {
.init = pulse_renderer_init,
.cleanup = pulse_renderer_cleanup,
.decodeAndPlaySample = pulse_renderer_decode_and_play_sample,
.capabilities = CAPABILITY_DIRECT_SUBMIT,
.capabilities = CAPABILITY_DIRECT_SUBMIT | CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION,
};

29
src/audio/sdl.c
View File

@ -26,7 +26,8 @@
#include <opus_multistream.h>
static OpusMSDecoder* decoder;
static short pcmBuffer[FRAME_SIZE * AUDIO_CONFIGURATION_MAX_CHANNEL_COUNT];
static short* pcmBuffer;
static int samplesPerFrame;
static SDL_AudioDeviceID dev;
static int channelCount;
@ -35,6 +36,10 @@ static int sdl_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURA
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;
SDL_InitSubSystem(SDL_INIT_AUDIO);
@ -45,13 +50,11 @@ static int sdl_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURA
want.channels = opusConfig->channelCount;
want.samples = 4096;
dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, SDL_AUDIO_ALLOW_FORMAT_CHANGE);
dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0);
if (dev == 0) {
printf("Failed to open audio: %s\n", SDL_GetError());
return -1;
} else {
if (have.format != want.format) // we let this one thing change.
printf("We didn't get requested audio format.\n");
SDL_PauseAudioDevice(dev, 0); // start audio playing.
}
@ -59,14 +62,24 @@ static int sdl_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURA
}
static void sdl_renderer_cleanup() {
if (decoder != NULL)
if (decoder != NULL) {
opus_multistream_decoder_destroy(decoder);
decoder = NULL;
}
SDL_CloseAudioDevice(dev);
if (pcmBuffer != NULL) {
free(pcmBuffer);
pcmBuffer = NULL;
}
if (dev != 0) {
SDL_CloseAudioDevice(dev);
dev = 0;
}
}
static void sdl_renderer_decode_and_play_sample(char* data, int length) {
int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, samplesPerFrame, 0);
if (decodeLen > 0) {
SDL_QueueAudio(dev, pcmBuffer, decodeLen * channelCount * sizeof(short));
} else {
@ -78,5 +91,5 @@ AUDIO_RENDERER_CALLBACKS audio_callbacks_sdl = {
.init = sdl_renderer_init,
.cleanup = sdl_renderer_cleanup,
.decodeAndPlaySample = sdl_renderer_decode_and_play_sample,
.capabilities = CAPABILITY_DIRECT_SUBMIT,
.capabilities = CAPABILITY_DIRECT_SUBMIT | CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION,
};