Error AudioDriverWASAPI::init_capture_device(bool reinit) {
Error err = audio_device_init(&audio_input, true, reinit);
if (err != OK)
return err;
// Get the max frames
UINT32 max_frames;
HRESULT hr = audio_input.audio_client->GetBufferSize(&max_frames);
ERR_FAIL_COND_V(hr != S_OK, ERR_CANT_OPEN);
// Set the buffer size
input_buffer.resize(max_frames * CAPTURE_BUFFER_CHANNELS);
input_position = 0;
input_size = 0;
return OK;
}
Error AudioDriverWASAPI::init_render_device(bool reinit) {
Error err = audio_device_init(&audio_output, false, reinit);
if (err != OK)
return err;
switch (audio_output.channels) {
case 2: // Stereo
case 4: // Surround 3.1
case 6: // Surround 5.1
case 8: // Surround 7.1
channels = audio_output.channels;
break;
default:
WARN_PRINTS("WASAPI: Unsupported number of channels: " + itos(audio_output.channels));
channels = 2;
break;
}
UINT32 max_frames;
HRESULT hr = audio_output.audio_client->GetBufferSize(&max_frames);
ERR_FAIL_COND_V(hr != S_OK, ERR_CANT_OPEN);
// Due to WASAPI Shared Mode we have no control of the buffer size
buffer_frames = max_frames;
// Sample rate is independent of channels (ref: https://stackoverflow.com/questions/11048825/audio-sample-frequency-rely-on-channels)
samples_in.resize(buffer_frames * channels);
input_position = 0;
input_size = 0;
print_verbose("WASAPI: detected " + itos(channels) + " channels");
print_verbose("WASAPI: audio buffer frames: " + itos(buffer_frames) + " calculated latency: " + itos(buffer_frames * 1000 / mix_rate) + "ms");
return OK;
}
void wavplay_thread_entry(void *parameter)
{
FILE *fp = NULL;
uint16_t *buffer = NULL;
struct wav_info *info = NULL;
fp = fopen(file_name, "rb");
if (!fp)
{
printf("open file failed!\n");
goto __exit;
}
info = (struct wav_info *) malloc(sizeof(*info));
if (!info) goto __exit;
if (fread(&(info->header), sizeof(struct RIFF_HEADER_DEF), 1, fp) != 1) goto __exit;
if (fread(&(info->fmt_block), sizeof(struct FMT_BLOCK_DEF), 1, fp) != 1) goto __exit;
if (fread(&(info->data_block), sizeof(struct DATA_BLOCK_DEF), 1, fp) != 1) goto __exit;
printf("wav information:\n");
printf("samplerate %u\n", info->fmt_block.wav_format.SamplesPerSec);
printf("channel %u\n", info->fmt_block.wav_format.Channels);
audio_device_init();
audio_device_open();
audio_device_set_rate(info->fmt_block.wav_format.SamplesPerSec);
while (!feof(fp))
{
int length;
buffer = (uint16_t *)audio_device_get_buffer(RT_NULL);
length = fread(buffer, 1, BUFSZ, fp);
if (length)
{
if (info->fmt_block.wav_format.Channels == 1)
{
/* extend to stereo channels */
int index;
uint16_t *ptr;
ptr = (uint16_t *)((uint8_t *)buffer + BUFSZ * 2);
for (index = 1; index < BUFSZ / 2; index ++)
{
*ptr = *(ptr - 1) = buffer[BUFSZ / 2 - index];
ptr -= 2;
}
length = length * 2;
}
audio_device_write((uint8_t *)buffer, length);
}
else
{
audio_device_put_buffer((uint8_t *)buffer);
break;
}
}
audio_device_close();
__exit:
if (fp) fclose(fp);
if (info) free(info);
}