static BOOL rdpsnd_winmm_open(rdpsndDevicePlugin* device, AUDIO_FORMAT* format, int latency)
{
MMRESULT mmResult;
rdpsndWinmmPlugin* winmm = (rdpsndWinmmPlugin*) device;
if (winmm->hWaveOut)
return TRUE;
rdpsnd_winmm_set_format(device, format, latency);
freerdp_dsp_context_reset_adpcm(winmm->dsp_context);
mmResult = waveOutOpen(&winmm->hWaveOut, WAVE_MAPPER, &winmm->format,
(DWORD_PTR) rdpsnd_winmm_callback_function, (DWORD_PTR) winmm, CALLBACK_FUNCTION);
if (mmResult != MMSYSERR_NOERROR)
{
WLog_ERR(TAG, "waveOutOpen failed: %"PRIu32"", mmResult);
return FALSE;
}
mmResult = waveOutSetVolume(winmm->hWaveOut, winmm->volume);
if (mmResult != MMSYSERR_NOERROR)
{
WLog_ERR(TAG, "waveOutSetVolume failed: %"PRIu32"", mmResult);
return FALSE;
}
return TRUE;
}
static void* audin_alsa_thread_func(void* arg)
{
int error;
uint8* buffer;
int rbytes_per_frame;
int tbytes_per_frame;
snd_pcm_t* capture_handle = NULL;
AudinALSADevice* alsa = (AudinALSADevice*) arg;
DEBUG_DVC("in");
rbytes_per_frame = alsa->actual_channels * alsa->bytes_per_channel;
tbytes_per_frame = alsa->target_channels * alsa->bytes_per_channel;
alsa->buffer = (uint8*) xzalloc(tbytes_per_frame * alsa->frames_per_packet);
alsa->buffer_frames = 0;
buffer = (uint8*) xzalloc(rbytes_per_frame * alsa->frames_per_packet);
freerdp_dsp_context_reset_adpcm(alsa->dsp_context);
do
{
if ((error = snd_pcm_open(&capture_handle, alsa->device_name, SND_PCM_STREAM_CAPTURE, 0)) < 0)
{
DEBUG_WARN("snd_pcm_open (%s)", snd_strerror(error));
break;
}
if (!audin_alsa_set_params(alsa, capture_handle))
{
break;
}
while (!freerdp_thread_is_stopped(alsa->thread))
{
error = snd_pcm_readi(capture_handle, buffer, alsa->frames_per_packet);
if (error == -EPIPE)
{
snd_pcm_recover(capture_handle, error, 0);
continue;
}
else if (error < 0)
{
DEBUG_WARN("snd_pcm_readi (%s)", snd_strerror(error));
break;
}
if (!audin_alsa_thread_receive(alsa, buffer, error * rbytes_per_frame))
break;
}
} while (0);
xfree(buffer);
xfree(alsa->buffer);
alsa->buffer = NULL;
if (capture_handle)
snd_pcm_close(capture_handle);
freerdp_thread_quit(alsa->thread);
DEBUG_DVC("out");
return NULL;
}
static void rdpsnd_server_select_format(rdpsnd_server_context* context, int client_format_index)
{
int bs;
int out_buffer_size;
rdpsndFormat *format;
rdpsnd_server* rdpsnd = (rdpsnd_server*) context;
if (client_format_index < 0 || client_format_index >= context->num_client_formats)
{
printf("rdpsnd_server_select_format: index %d is not correct.\n", client_format_index);
return;
}
rdpsnd->src_bytes_per_sample = context->src_format.wBitsPerSample / 8;
rdpsnd->src_bytes_per_frame = rdpsnd->src_bytes_per_sample * context->src_format.nChannels;
context->selected_client_format = client_format_index;
format = &context->client_formats[client_format_index];
if (format->wFormatTag == 0x11)
{
bs = (format->nBlockAlign - 4 * format->nChannels) * 4;
rdpsnd->out_frames = (format->nBlockAlign * 4 * format->nChannels * 2 / bs + 1) * bs / (format->nChannels * 2);
}
else if (format->wFormatTag == 0x02)
{
bs = (format->nBlockAlign - 7 * format->nChannels) * 2 / format->nChannels + 2;
rdpsnd->out_frames = bs * 4;
}
else
{
rdpsnd->out_frames = 0x4000 / rdpsnd->src_bytes_per_frame;
}
if (format->nSamplesPerSec != context->src_format.nSamplesPerSec)
{
rdpsnd->out_frames = (rdpsnd->out_frames * context->src_format.nSamplesPerSec + format->nSamplesPerSec - 100) / format->nSamplesPerSec;
}
rdpsnd->out_pending_frames = 0;
out_buffer_size = rdpsnd->out_frames * rdpsnd->src_bytes_per_frame;
if (rdpsnd->out_buffer_size < out_buffer_size)
{
rdpsnd->out_buffer = (BYTE*) realloc(rdpsnd->out_buffer, out_buffer_size);
rdpsnd->out_buffer_size = out_buffer_size;
}
freerdp_dsp_context_reset_adpcm(rdpsnd->dsp_context);
}
static void rdpsnd_pulse_open(rdpsndDevicePlugin* device, rdpsndFormat* format, int latency)
{
rdpsndPulsePlugin* pulse = (rdpsndPulsePlugin*)device;
pa_stream_state_t state;
pa_stream_flags_t flags;
pa_buffer_attr buffer_attr = { 0 };
char ss[PA_SAMPLE_SPEC_SNPRINT_MAX];
if (!pulse->context || pulse->stream)
{
DEBUG_WARN("pulse stream has been created.");
return;
}
rdpsnd_pulse_set_format_spec(pulse, format);
pulse->latency = latency;
if (pa_sample_spec_valid(&pulse->sample_spec) == 0)
{
pa_sample_spec_snprint(ss, sizeof(ss), &pulse->sample_spec);
DEBUG_WARN("Invalid sample spec %s", ss);
return;
}
pa_threaded_mainloop_lock(pulse->mainloop);
pulse->stream = pa_stream_new(pulse->context, "freerdp",
&pulse->sample_spec, NULL);
if (!pulse->stream)
{
pa_threaded_mainloop_unlock(pulse->mainloop);
DEBUG_WARN("pa_stream_new failed (%d)",
pa_context_errno(pulse->context));
return;
}
/* install essential callbacks */
pa_stream_set_state_callback(pulse->stream,
rdpsnd_pulse_stream_state_callback, pulse);
pa_stream_set_write_callback(pulse->stream,
rdpsnd_pulse_stream_request_callback, pulse);
flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE;
if (pulse->latency > 0)
{
buffer_attr.maxlength = pa_usec_to_bytes(pulse->latency * 2 * 1000, &pulse->sample_spec);
buffer_attr.tlength = pa_usec_to_bytes(pulse->latency * 1000, &pulse->sample_spec);
buffer_attr.prebuf = (uint32_t) -1;
buffer_attr.minreq = (uint32_t) -1;
buffer_attr.fragsize = (uint32_t) -1;
flags |= PA_STREAM_ADJUST_LATENCY;
}
if (pa_stream_connect_playback(pulse->stream,
pulse->device_name, pulse->latency > 0 ? &buffer_attr : NULL, flags, NULL, NULL) < 0)
{
pa_threaded_mainloop_unlock(pulse->mainloop);
DEBUG_WARN("pa_stream_connect_playback failed (%d)",
pa_context_errno(pulse->context));
return;
}
for (;;)
{
state = pa_stream_get_state(pulse->stream);
if (state == PA_STREAM_READY)
break;
if (!PA_STREAM_IS_GOOD(state))
{
DEBUG_WARN("bad stream state (%d)",
pa_context_errno(pulse->context));
break;
}
pa_threaded_mainloop_wait(pulse->mainloop);
}
pa_threaded_mainloop_unlock(pulse->mainloop);
if (state == PA_STREAM_READY)
{
freerdp_dsp_context_reset_adpcm(pulse->dsp_context);
DEBUG_SVC("connected");
}
else
{
rdpsnd_pulse_close(device);
}
}
static BOOL rdpsnd_server_select_format(RdpsndServerContext* context, int client_format_index)
{
int bs;
int out_buffer_size;
AUDIO_FORMAT *format;
if (client_format_index < 0 || client_format_index >= context->num_client_formats)
{
fprintf(stderr, "%s: index %d is not correct.\n", __FUNCTION__, client_format_index);
return FALSE;
}
context->priv->src_bytes_per_sample = context->src_format.wBitsPerSample / 8;
context->priv->src_bytes_per_frame = context->priv->src_bytes_per_sample * context->src_format.nChannels;
context->selected_client_format = client_format_index;
format = &context->client_formats[client_format_index];
if (format->nSamplesPerSec == 0)
{
fprintf(stderr, "%s: invalid Client Sound Format!!\n", __FUNCTION__);
return FALSE;
}
switch(format->wFormatTag)
{
case WAVE_FORMAT_DVI_ADPCM:
bs = (format->nBlockAlign - 4 * format->nChannels) * 4;
context->priv->out_frames = (format->nBlockAlign * 4 * format->nChannels * 2 / bs + 1) * bs / (format->nChannels * 2);
break;
case WAVE_FORMAT_ADPCM:
bs = (format->nBlockAlign - 7 * format->nChannels) * 2 / format->nChannels + 2;
context->priv->out_frames = bs * 4;
break;
default:
context->priv->out_frames = 0x4000 / context->priv->src_bytes_per_frame;
break;
}
if (format->nSamplesPerSec != context->src_format.nSamplesPerSec)
{
context->priv->out_frames = (context->priv->out_frames * context->src_format.nSamplesPerSec + format->nSamplesPerSec - 100) / format->nSamplesPerSec;
}
context->priv->out_pending_frames = 0;
out_buffer_size = context->priv->out_frames * context->priv->src_bytes_per_frame;
if (context->priv->out_buffer_size < out_buffer_size)
{
BYTE *newBuffer;
newBuffer = (BYTE *)realloc(context->priv->out_buffer, out_buffer_size);
if (!newBuffer)
return FALSE;
context->priv->out_buffer = newBuffer;
context->priv->out_buffer_size = out_buffer_size;
}
freerdp_dsp_context_reset_adpcm(context->priv->dsp_context);
return TRUE;
}
/**
* Function description
*
* @return 0 on success, otherwise a Win32 error code
*/
static UINT rdpsnd_server_select_format(RdpsndServerContext* context, int client_format_index)
{
int bs;
int out_buffer_size;
AUDIO_FORMAT *format;
UINT error = CHANNEL_RC_OK;
if (client_format_index < 0 || client_format_index >= context->num_client_formats)
{
WLog_ERR(TAG, "index %d is not correct.", client_format_index);
return ERROR_INVALID_DATA;
}
EnterCriticalSection(&context->priv->lock);
context->priv->src_bytes_per_sample = context->src_format.wBitsPerSample / 8;
context->priv->src_bytes_per_frame = context->priv->src_bytes_per_sample * context->src_format.nChannels;
context->selected_client_format = client_format_index;
format = &context->client_formats[client_format_index];
if (format->nSamplesPerSec == 0)
{
WLog_ERR(TAG, "invalid Client Sound Format!!");
error = ERROR_INVALID_DATA;
goto out;
}
if (context->latency <= 0)
context->latency = 50;
context->priv->out_frames = context->src_format.nSamplesPerSec * context->latency / 1000;
if (context->priv->out_frames < 1)
context->priv->out_frames = 1;
switch(format->wFormatTag)
{
case WAVE_FORMAT_DVI_ADPCM:
bs = (format->nBlockAlign - 4 * format->nChannels) * 4;
context->priv->out_frames -= context->priv->out_frames % bs;
if (context->priv->out_frames < bs)
context->priv->out_frames = bs;
break;
case WAVE_FORMAT_ADPCM:
bs = (format->nBlockAlign - 7 * format->nChannels) * 2 / format->nChannels + 2;
context->priv->out_frames -= context->priv->out_frames % bs;
if (context->priv->out_frames < bs)
context->priv->out_frames = bs;
break;
}
context->priv->out_pending_frames = 0;
out_buffer_size = context->priv->out_frames * context->priv->src_bytes_per_frame;
if (context->priv->out_buffer_size < out_buffer_size)
{
BYTE *newBuffer;
newBuffer = (BYTE *)realloc(context->priv->out_buffer, out_buffer_size);
if (!newBuffer)
{
WLog_ERR(TAG, "realloc failed!");
error = CHANNEL_RC_NO_MEMORY;
goto out;
}
context->priv->out_buffer = newBuffer;
context->priv->out_buffer_size = out_buffer_size;
}
freerdp_dsp_context_reset_adpcm(context->priv->dsp_context);
out:
LeaveCriticalSection(&context->priv->lock);
return error;
}
