static pa_stream *connect_playback_stream(const char *device_name,
pa_threaded_mainloop *loop, pa_context *context,
pa_stream_flags_t flags, pa_buffer_attr *attr, pa_sample_spec *spec,
pa_channel_map *chanmap)
{
pa_stream_state_t state;
pa_stream *stream;
stream = pa_stream_new_with_proplist(context, "Playback Stream", spec, chanmap, prop_filter);
if(!stream)
{
ERR("pa_stream_new_with_proplist() failed: %s\n", pa_strerror(pa_context_errno(context)));
return NULL;
}
pa_stream_set_state_callback(stream, stream_state_callback, loop);
if(pa_stream_connect_playback(stream, device_name, attr, flags, NULL, NULL) < 0)
{
ERR("Stream did not connect: %s\n", pa_strerror(pa_context_errno(context)));
pa_stream_unref(stream);
return NULL;
}
while((state=pa_stream_get_state(stream)) != PA_STREAM_READY)
{
if(!PA_STREAM_IS_GOOD(state))
{
ERR("Stream did not get ready: %s\n", pa_strerror(pa_context_errno(context)));
pa_stream_unref(stream);
return NULL;
}
pa_threaded_mainloop_wait(loop);
}
pa_stream_set_state_callback(stream, NULL, NULL);
return stream;
}
static ALCboolean pulse_reset_playback(ALCdevice *device) //{{{
{
pulse_data *data = device->ExtraData;
pa_stream_state_t state;
ppa_threaded_mainloop_lock(data->loop);
data->frame_size = aluBytesFromFormat(device->Format) *
aluChannelsFromFormat(device->Format);
data->attr.minreq = data->frame_size * device->UpdateSize;
data->attr.prebuf = -1;
data->attr.maxlength = -1;
data->attr.fragsize = -1;
data->attr.tlength = data->attr.minreq * device->NumUpdates;
data->stream_name = "Playback Stream";
switch(aluBytesFromFormat(device->Format))
{
case 1:
data->spec.format = PA_SAMPLE_U8;
break;
case 2:
data->spec.format = PA_SAMPLE_S16NE;
break;
case 4:
data->spec.format = PA_SAMPLE_FLOAT32NE;
break;
default:
AL_PRINT("Unknown format: 0x%x\n", device->Format);
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
data->spec.rate = device->Frequency;
data->spec.channels = aluChannelsFromFormat(device->Format);
if(ppa_sample_spec_valid(&data->spec) == 0)
{
AL_PRINT("Invalid sample format\n");
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
data->stream = ppa_stream_new(data->context, data->stream_name, &data->spec, NULL);
if(!data->stream)
{
AL_PRINT("pa_stream_new() failed: %s\n",
ppa_strerror(ppa_context_errno(data->context)));
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
if(ppa_stream_connect_playback(data->stream, NULL, &data->attr, PA_STREAM_ADJUST_LATENCY, NULL, NULL) < 0)
{
AL_PRINT("Stream did not connect: %s\n",
ppa_strerror(ppa_context_errno(data->context)));
ppa_stream_unref(data->stream);
data->stream = NULL;
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
while((state=ppa_stream_get_state(data->stream)) != PA_STREAM_READY)
{
if(!PA_STREAM_IS_GOOD(state))
{
AL_PRINT("Stream did not get ready: %s\n",
ppa_strerror(ppa_context_errno(data->context)));
ppa_stream_unref(data->stream);
data->stream = NULL;
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
ppa_threaded_mainloop_unlock(data->loop);
Sleep(1);
ppa_threaded_mainloop_lock(data->loop);
}
data->attr = *(ppa_stream_get_buffer_attr(data->stream));
if((data->attr.tlength%data->attr.minreq) != 0)
AL_PRINT("tlength (%d) is not a multiple of minreq (%d)!\n",
data->attr.tlength, data->attr.minreq);
device->UpdateSize = data->attr.minreq;
device->NumUpdates = data->attr.tlength/data->attr.minreq;
ppa_stream_set_write_callback(data->stream, stream_write_callback, device);
ppa_threaded_mainloop_unlock(data->loop);
return ALC_TRUE;
} //}}}
void SetupSound (void)
{
int error_number;
// Acquire mainloop ///////////////////////////////////////////////////////
device.mainloop = pa_threaded_mainloop_new ();
if (device.mainloop == NULL)
{
fprintf (stderr, "Could not acquire PulseAudio main loop\n");
return;
}
// Acquire context ////////////////////////////////////////////////////////
device.api = pa_threaded_mainloop_get_api (device.mainloop);
device.context = pa_context_new (device.api, "PCSXR");
pa_context_set_state_callback (device.context, context_state_cb, &device);
if (device.context == NULL)
{
fprintf (stderr, "Could not acquire PulseAudio device context\n");
return;
}
// Connect to PulseAudio server ///////////////////////////////////////////
if (pa_context_connect (device.context, NULL, 0, NULL) < 0)
{
error_number = pa_context_errno (device.context);
fprintf (stderr, "Could not connect to PulseAudio server: %s\n", pa_strerror(error_number));
return;
}
// Run mainloop until sever context is ready //////////////////////////////
pa_threaded_mainloop_lock (device.mainloop);
if (pa_threaded_mainloop_start (device.mainloop) < 0)
{
fprintf (stderr, "Could not start mainloop\n");
return;
}
pa_context_state_t context_state;
context_state = pa_context_get_state (device.context);
while (context_state != PA_CONTEXT_READY)
{
context_state = pa_context_get_state (device.context);
if (! PA_CONTEXT_IS_GOOD (context_state))
{
error_number = pa_context_errno (device.context);
fprintf (stderr, "Context state is not good: %s\n", pa_strerror (error_number));
return;
}
else if (context_state == PA_CONTEXT_READY)
break;
else
fprintf (stderr, "PulseAudio context state is %d\n", context_state);
pa_threaded_mainloop_wait (device.mainloop);
}
// Set sample spec ////////////////////////////////////////////////////////
device.spec.format = PA_SAMPLE_S16NE;
if (iDisStereo)
device.spec.channels = 1;
else
device.spec.channels = 2;
device.spec.rate = settings.frequency;
pa_buffer_attr buffer_attributes;
buffer_attributes.tlength = pa_bytes_per_second (& device.spec) / 5;
buffer_attributes.maxlength = buffer_attributes.tlength * 3;
buffer_attributes.minreq = buffer_attributes.tlength / 3;
buffer_attributes.prebuf = buffer_attributes.tlength;
//maxlength = buffer_attributes.maxlength;
//fprintf (stderr, "Total space: %u\n", buffer_attributes.maxlength);
//fprintf (stderr, "Minimum request size: %u\n", buffer_attributes.minreq);
//fprintf (stderr, "Bytes needed before playback: %u\n", buffer_attributes.prebuf);
//fprintf (stderr, "Target buffer size: %lu\n", buffer_attributes.tlength);
// Acquire new stream using spec //////////////////////////////////////////
device.stream = pa_stream_new (device.context, "PCSXR", &device.spec, NULL);
if (device.stream == NULL)
{
error_number = pa_context_errno (device.context);
fprintf (stderr, "Could not acquire new PulseAudio stream: %s\n", pa_strerror (error_number));
return;
}
// Set callbacks for server events ////////////////////////////////////////
pa_stream_set_state_callback (device.stream, stream_state_cb, &device);
pa_stream_set_write_callback (device.stream, stream_request_cb, &device);
pa_stream_set_latency_update_callback (device.stream, stream_latency_update_cb, &device);
// Ready stream for playback //////////////////////////////////////////////
pa_stream_flags_t flags = (pa_stream_flags_t) (PA_STREAM_ADJUST_LATENCY | PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE);
//pa_stream_flags_t flags = (pa_stream_flags_t) (PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_EARLY_REQUESTS);
if (pa_stream_connect_playback (device.stream, NULL, &buffer_attributes, flags, NULL, NULL) < 0)
{
pa_context_errno (device.context);
fprintf (stderr, "Could not connect for playback: %s\n", pa_strerror (error_number));
return;
}
// Run mainloop until stream is ready /////////////////////////////////////
pa_stream_state_t stream_state;
stream_state = pa_stream_get_state (device.stream);
while (stream_state != PA_STREAM_READY)
{
stream_state = pa_stream_get_state (device.stream);
if (stream_state == PA_STREAM_READY)
break;
else if (! PA_STREAM_IS_GOOD (stream_state))
{
error_number = pa_context_errno (device.context);
fprintf (stderr, "Stream state is not good: %s\n", pa_strerror (error_number));
return;
}
else
fprintf (stderr, "PulseAudio stream state is %d\n", stream_state);
pa_threaded_mainloop_wait (device.mainloop);
}
pa_threaded_mainloop_unlock (device.mainloop);
fprintf (stderr, "PulseAudio should be connected\n");
return;
}
static void audin_pulse_open(IAudinDevice* device, AudinReceive receive, void* user_data)
{
pa_stream_state_t state;
pa_buffer_attr buffer_attr = { 0 };
AudinPulseDevice* pulse = (AudinPulseDevice*) device;
if (!pulse->context)
return;
if (!pulse->sample_spec.rate || pulse->stream)
return;
DEBUG_DVC("");
pulse->receive = receive;
pulse->user_data = user_data;
pa_threaded_mainloop_lock(pulse->mainloop);
pulse->stream = pa_stream_new(pulse->context, "freerdp_audin",
&pulse->sample_spec, NULL);
if (!pulse->stream)
{
pa_threaded_mainloop_unlock(pulse->mainloop);
DEBUG_DVC("pa_stream_new failed (%d)",
pa_context_errno(pulse->context));
return;
}
pulse->bytes_per_frame = pa_frame_size(&pulse->sample_spec);
pa_stream_set_state_callback(pulse->stream,
audin_pulse_stream_state_callback, pulse);
pa_stream_set_read_callback(pulse->stream,
audin_pulse_stream_request_callback, pulse);
buffer_attr.maxlength = (uint32_t) -1;
buffer_attr.tlength = (uint32_t) -1;
buffer_attr.prebuf = (uint32_t) -1;
buffer_attr.minreq = (uint32_t) -1;
/* 500ms latency */
buffer_attr.fragsize = pa_usec_to_bytes(500000, &pulse->sample_spec);
if (pa_stream_connect_record(pulse->stream,
pulse->device_name[0] ? pulse->device_name : NULL,
&buffer_attr, PA_STREAM_ADJUST_LATENCY) < 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)
{
memset(&pulse->adpcm, 0, sizeof(ADPCM));
pulse->buffer = xzalloc(pulse->bytes_per_frame * pulse->frames_per_packet);
pulse->buffer_frames = 0;
DEBUG_DVC("connected");
}
else
{
audin_pulse_close(device);
}
}
static gboolean
gst_pulsesrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
{
pa_buffer_attr wanted;
const pa_buffer_attr *actual;
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_threaded_mainloop_lock (pulsesrc->mainloop);
wanted.maxlength = -1;
wanted.tlength = -1;
wanted.prebuf = 0;
wanted.minreq = -1;
wanted.fragsize = spec->segsize;
GST_INFO_OBJECT (pulsesrc, "maxlength: %d", wanted.maxlength);
GST_INFO_OBJECT (pulsesrc, "tlength: %d", wanted.tlength);
GST_INFO_OBJECT (pulsesrc, "prebuf: %d", wanted.prebuf);
GST_INFO_OBJECT (pulsesrc, "minreq: %d", wanted.minreq);
GST_INFO_OBJECT (pulsesrc, "fragsize: %d", wanted.fragsize);
if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &wanted,
PA_STREAM_INTERPOLATE_TIMING |
PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_NOT_MONOTONOUS |
#ifdef HAVE_PULSE_0_9_11
PA_STREAM_ADJUST_LATENCY |
#endif
PA_STREAM_START_CORKED) < 0) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
pulsesrc->corked = TRUE;
for (;;) {
pa_stream_state_t state;
state = pa_stream_get_state (pulsesrc->stream);
if (!PA_STREAM_IS_GOOD (state)) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
if (state == PA_STREAM_READY)
break;
/* Wait until the stream is ready */
pa_threaded_mainloop_wait (pulsesrc->mainloop);
}
/* get the actual buffering properties now */
actual = pa_stream_get_buffer_attr (pulsesrc->stream);
GST_INFO_OBJECT (pulsesrc, "maxlength: %d", actual->maxlength);
GST_INFO_OBJECT (pulsesrc, "tlength: %d (wanted: %d)",
actual->tlength, wanted.tlength);
GST_INFO_OBJECT (pulsesrc, "prebuf: %d", actual->prebuf);
GST_INFO_OBJECT (pulsesrc, "minreq: %d (wanted %d)", actual->minreq,
wanted.minreq);
GST_INFO_OBJECT (pulsesrc, "fragsize: %d (wanted %d)",
actual->fragsize, wanted.fragsize);
if (actual->fragsize >= wanted.fragsize) {
spec->segsize = actual->fragsize;
} else {
spec->segsize = actual->fragsize * (wanted.fragsize / actual->fragsize);
}
spec->segtotal = actual->maxlength / spec->segsize;
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return TRUE;
unlock_and_fail:
{
gst_pulsesrc_destroy_stream (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return FALSE;
}
}
INT16 m1sdr_Init(int sample_rate)
{
int format, stereo, rate, fsize, err, state;
unsigned int nfreq, periodtime;
snd_pcm_hw_params_t *hwparams;
#ifdef USE_SDL
SDL_AudioSpec aspec;
#endif
pa_channel_map chanmap;
pa_buffer_attr my_pa_attr;
hw_present = 0;
m1sdr_Callback = NULL;
nDSoundSegLen = sample_rate / 60;
switch (lnxdrv_apimode)
{
case 0: // SDL
#ifdef USE_SDL
SDL_InitSubSystem(SDL_INIT_AUDIO);
m1sdr_SetSamplesPerTick(sample_rate/60);
playbuf = 0;
writebuf = 1;
aspec.freq = sample_rate;
aspec.format = AUDIO_S16SYS; // keep endian independant
aspec.channels = 2;
aspec.samples = 512; // has to be a power of 2, and we want it smaller than our buffer size
aspec.callback = sdl_callback;
aspec.userdata = 0;
if (SDL_OpenAudio(&aspec, NULL) < 0)
{
printf("ERROR: can't open SDL audio\n");
return 0;
}
// make sure we don't start yet
SDL_PauseAudio(1);
#endif
break;
case 1: // ALSA
// Try to open audio device
if ((err = snd_pcm_open(&pHandle, "default", SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
fprintf(stderr, "ALSA: Could not open soundcard (%s)\n", snd_strerror(err));
hw_present = 0;
return 0;
}
if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0) {
fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n",
snd_strerror(err));
return 0;
}
// Init hwparams with full configuration space
if ((err = snd_pcm_hw_params_any(pHandle, hwparams)) < 0) {
fprintf(stderr, "ALSA: couldn't set hw params (%s)\n", snd_strerror(err));
hw_present = 0;
return 0;
}
// Set access type
if ((err = snd_pcm_hw_params_set_access(pHandle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
fprintf(stderr, "ALSA: can't set access (%s)\n", snd_strerror(err));
return 0;
}
// Set sample format
if ((err = snd_pcm_hw_params_set_format(pHandle, hwparams, SND_PCM_FORMAT_S16)) < 0) {
fprintf(stderr, "ALSA: can't set format (%s)\n", snd_strerror(err));
return 0;
}
// Set sample rate (nearest possible)
nfreq = sample_rate;
if ((err = snd_pcm_hw_params_set_rate_near(pHandle, hwparams, &nfreq, 0)) < 0) {
fprintf(stderr, "ALSA: can't set sample rate (%s)\n", snd_strerror(err));
return 0;
}
// Set number of channels
if ((err = snd_pcm_hw_params_set_channels(pHandle, hwparams, 2)) < 0) {
fprintf(stderr, "ALSA: can't set stereo (%s)\n", snd_strerror(err));
return 0;
}
// Set period time (nearest possible)
periodtime = 16;
if ((err = snd_pcm_hw_params_set_period_time_near(pHandle, hwparams, &periodtime, 0)) < 0) {
fprintf(stderr, "ALSA: can't set period time (%s)\n", snd_strerror(err));
return 0;
}
// Apply HW parameter settings to PCM device and prepare device
if ((err = snd_pcm_hw_params(pHandle, hwparams)) < 0) {
fprintf(stderr, "ALSA: unable to install hw_params (%s)\n", snd_strerror(err));
snd_pcm_hw_params_free(hwparams);
return 0;
}
snd_pcm_hw_params_free(hwparams);
if ((err = snd_pcm_prepare(pHandle)) < 0) {
fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror(err));
return 0;
}
break;
case 2: // OSS
audiofd = open("/dev/dsp", O_WRONLY, 0);
if (audiofd == -1)
{
audiofd = open("/dev/dsp1", O_WRONLY, 0);
if (audiofd == -1)
{
perror("/dev/dsp1");
return(0);
}
}
// reset things
ioctl(audiofd, SNDCTL_DSP_RESET, 0);
is_broken_driver = 0;
num_frags = NUM_FRAGS_NORMAL;
// set the buffer size we want
fsize = OSS_FRAGMENT;
if (ioctl(audiofd, SNDCTL_DSP_SETFRAGMENT, &fsize) == - 1)
{
perror("SNDCTL_DSP_SETFRAGMENT");
return(0);
}
// set 16-bit output
format = AFMT_S16_NE; // 16 bit signed "native"-endian
if (ioctl(audiofd, SNDCTL_DSP_SETFMT, &format) == - 1)
{
perror("SNDCTL_DSP_SETFMT");
return(0);
}
// now set stereo
stereo = 1;
if (ioctl(audiofd, SNDCTL_DSP_STEREO, &stereo) == - 1)
{
perror("SNDCTL_DSP_STEREO");
return(0);
}
// and the sample rate
rate = sample_rate;
if (ioctl(audiofd, SNDCTL_DSP_SPEED, &rate) == - 1)
{
perror("SNDCTL_DSP_SPEED");
return(0);
}
// and make sure that did what we wanted
ioctl(audiofd, SNDCTL_DSP_GETBLKSIZE, &fsize);
break;
case 3: // PulseAudio
sample_spec.format = PA_SAMPLE_S16NE;
sample_spec.rate = sample_rate;
sample_spec.channels = 2;
my_pa_context = NULL;
my_pa_stream = NULL;
my_pa_mainloop = NULL;
my_pa_mainloop_api = NULL;
#if !PULSE_USE_SIMPLE
// get default channel mapping
pa_channel_map_init_auto(&chanmap, sample_spec.channels, PA_CHANNEL_MAP_WAVEEX);
if (!(my_pa_mainloop = pa_mainloop_new()))
{
fprintf(stderr, "pa_mainloop_new() failed\n");
return 0;
}
my_pa_mainloop_api = pa_mainloop_get_api(my_pa_mainloop);
/* if (pa_signal_init(my_pa_mainloop_api) != 0)
{
fprintf(stderr, "pa_signal_init() failed\n");
return 0;
}*/
/* Create a new connection context */
if (!(my_pa_context = pa_context_new(my_pa_mainloop_api, "Audio Overload")))
{
fprintf(stderr, "pa_context_new() failed\n");
return 0;
}
/* set the context state CB */
// pa_context_set_state_callback(my_pa_context, context_state_callback, NULL);
/* Connect the context */
if (pa_context_connect(my_pa_context, NULL, (pa_context_flags_t)0, NULL) < 0)
{
fprintf(stderr, "pa_context_connect() failed: %s", pa_strerror(pa_context_errno(my_pa_context)));
return 0;
}
do
{
pa_mainloop_iterate(my_pa_mainloop, 1, NULL);
state = pa_context_get_state(my_pa_context);
if (!PA_CONTEXT_IS_GOOD((pa_context_state_t)state))
{
printf("PA CONTEXT NOT GOOD\n");
hw_present = 0;
return 0;
}
} while (state != PA_CONTEXT_READY);
if (!(my_pa_stream = pa_stream_new(my_pa_context, "Audio Overload", &sample_spec, &chanmap)))
{
fprintf(stderr, "pa_stream_new() failed: %s\n", pa_strerror(pa_context_errno(my_pa_context)));
return 0;
}
memset(&my_pa_attr, 0, sizeof(my_pa_attr));
my_pa_attr.tlength = nDSoundSegLen * 4 * 4;
my_pa_attr.prebuf = -1;
my_pa_attr.maxlength = -1;
my_pa_attr.minreq = nDSoundSegLen * 4 * 2;
if ((err = pa_stream_connect_playback(my_pa_stream, NULL, &my_pa_attr, PA_STREAM_ADJUST_LATENCY, NULL, NULL)) < 0)
{
fprintf(stderr, "pa_stream_connect_playback() failed: %s\n", pa_strerror(pa_context_errno(my_pa_context)));
return 0;
}
do
{
pa_mainloop_iterate(my_pa_mainloop, 1, NULL);
state = pa_stream_get_state(my_pa_stream);
if (!PA_STREAM_IS_GOOD((pa_stream_state_t)state))
{
printf("PA STREAM NOT GOOD\n");
hw_present = 0;
return 0;
}
} while (state != PA_STREAM_READY);
// printf("PulseAudio setup OK so far, len %d\n", nDSoundSegLen*4);
#else
my_simple = NULL;
#endif
break;
}
hw_present = 1;
return (1);
}
static int
PULSEAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
{
struct SDL_PrivateAudioData *h = NULL;
Uint16 test_format = 0;
pa_sample_spec paspec;
pa_buffer_attr paattr;
pa_channel_map pacmap;
pa_stream_flags_t flags = 0;
int state = 0;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
h = this->hidden;
paspec.format = PA_SAMPLE_INVALID;
/* Try for a closest match on audio format */
for (test_format = SDL_FirstAudioFormat(this->spec.format);
(paspec.format == PA_SAMPLE_INVALID) && test_format;) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
switch (test_format) {
case AUDIO_U8:
paspec.format = PA_SAMPLE_U8;
break;
case AUDIO_S16LSB:
paspec.format = PA_SAMPLE_S16LE;
break;
case AUDIO_S16MSB:
paspec.format = PA_SAMPLE_S16BE;
break;
case AUDIO_S32LSB:
paspec.format = PA_SAMPLE_S32LE;
break;
case AUDIO_S32MSB:
paspec.format = PA_SAMPLE_S32BE;
break;
case AUDIO_F32LSB:
paspec.format = PA_SAMPLE_FLOAT32LE;
break;
case AUDIO_F32MSB:
paspec.format = PA_SAMPLE_FLOAT32BE;
break;
default:
paspec.format = PA_SAMPLE_INVALID;
break;
}
if (paspec.format == PA_SAMPLE_INVALID) {
test_format = SDL_NextAudioFormat();
}
}
if (paspec.format == PA_SAMPLE_INVALID) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Couldn't find any hardware audio formats");
}
this->spec.format = test_format;
/* Calculate the final parameters for this audio specification */
#ifdef PA_STREAM_ADJUST_LATENCY
this->spec.samples /= 2; /* Mix in smaller chunck to avoid underruns */
#endif
SDL_CalculateAudioSpec(&this->spec);
/* Allocate mixing buffer */
h->mixlen = this->spec.size;
h->mixbuf = (Uint8 *) SDL_AllocAudioMem(h->mixlen);
if (h->mixbuf == NULL) {
PULSEAUDIO_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(h->mixbuf, this->spec.silence, this->spec.size);
paspec.channels = this->spec.channels;
paspec.rate = this->spec.freq;
/* Reduced prebuffering compared to the defaults. */
#ifdef PA_STREAM_ADJUST_LATENCY
/* 2x original requested bufsize */
paattr.tlength = h->mixlen * 4;
paattr.prebuf = -1;
paattr.maxlength = -1;
/* -1 can lead to pa_stream_writable_size() >= mixlen never being true */
paattr.minreq = h->mixlen;
flags = PA_STREAM_ADJUST_LATENCY;
#else
paattr.tlength = h->mixlen*2;
paattr.prebuf = h->mixlen*2;
paattr.maxlength = h->mixlen*2;
paattr.minreq = h->mixlen;
#endif
/* The SDL ALSA output hints us that we use Windows' channel mapping */
/* http://bugzilla.libsdl.org/show_bug.cgi?id=110 */
PULSEAUDIO_pa_channel_map_init_auto(&pacmap, this->spec.channels,
PA_CHANNEL_MAP_WAVEEX);
/* Set up a new main loop */
if (!(h->mainloop = PULSEAUDIO_pa_mainloop_new())) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("pa_mainloop_new() failed");
}
h->mainloop_api = PULSEAUDIO_pa_mainloop_get_api(h->mainloop);
h->context = PULSEAUDIO_pa_context_new(h->mainloop_api, getAppName());
if (!h->context) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("pa_context_new() failed");
}
/* Connect to the PulseAudio server */
if (PULSEAUDIO_pa_context_connect(h->context, NULL, 0, NULL) < 0) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Could not setup connection to PulseAudio");
}
do {
if (PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("pa_mainloop_iterate() failed");
}
state = PULSEAUDIO_pa_context_get_state(h->context);
if (!PA_CONTEXT_IS_GOOD(state)) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Could not connect to PulseAudio");
}
} while (state != PA_CONTEXT_READY);
h->stream = PULSEAUDIO_pa_stream_new(
h->context,
"Simple DirectMedia Layer", /* stream description */
&paspec, /* sample format spec */
&pacmap /* channel map */
);
if (h->stream == NULL) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Could not set up PulseAudio stream");
}
if (PULSEAUDIO_pa_stream_connect_playback(h->stream, NULL, &paattr, flags,
NULL, NULL) < 0) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Could not connect PulseAudio stream");
}
do {
if (PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("pa_mainloop_iterate() failed");
}
state = PULSEAUDIO_pa_stream_get_state(h->stream);
if (!PA_STREAM_IS_GOOD(state)) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Could not create to PulseAudio stream");
}
} while (state != PA_STREAM_READY);
/* We're ready to rock and roll. :-) */
return 0;
}
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 gboolean
gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
pa_buffer_attr wanted;
const pa_buffer_attr *actual;
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_stream_flags_t flags;
pa_operation *o;
GstAudioClock *clock;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
{
GstAudioRingBufferSpec s = *spec;
const pa_channel_map *m;
m = pa_stream_get_channel_map (pulsesrc->stream);
gst_pulse_channel_map_to_gst (m, &s);
gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
(pulsesrc)->ringbuffer, s.info.position);
}
/* enable event notifications */
GST_LOG_OBJECT (pulsesrc, "subscribing to context events");
if (!(o = pa_context_subscribe (pulsesrc->context,
PA_SUBSCRIPTION_MASK_SOURCE_OUTPUT, NULL, NULL))) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_context_subscribe() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
pa_operation_unref (o);
wanted.maxlength = -1;
wanted.tlength = -1;
wanted.prebuf = 0;
wanted.minreq = -1;
wanted.fragsize = spec->segsize;
GST_INFO_OBJECT (pulsesrc, "maxlength: %d", wanted.maxlength);
GST_INFO_OBJECT (pulsesrc, "tlength: %d", wanted.tlength);
GST_INFO_OBJECT (pulsesrc, "prebuf: %d", wanted.prebuf);
GST_INFO_OBJECT (pulsesrc, "minreq: %d", wanted.minreq);
GST_INFO_OBJECT (pulsesrc, "fragsize: %d", wanted.fragsize);
flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
PA_STREAM_NOT_MONOTONIC | PA_STREAM_ADJUST_LATENCY |
PA_STREAM_START_CORKED;
if (pulsesrc->mute_set && pulsesrc->mute)
flags |= PA_STREAM_START_MUTED;
if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &wanted,
flags) < 0) {
goto connect_failed;
}
/* our clock will now start from 0 again */
clock = GST_AUDIO_CLOCK (GST_AUDIO_BASE_SRC (pulsesrc)->clock);
gst_audio_clock_reset (clock, 0);
pulsesrc->corked = TRUE;
for (;;) {
pa_stream_state_t state;
state = pa_stream_get_state (pulsesrc->stream);
if (!PA_STREAM_IS_GOOD (state))
goto stream_is_bad;
if (state == PA_STREAM_READY)
break;
/* Wait until the stream is ready */
pa_threaded_mainloop_wait (pulsesrc->mainloop);
}
pulsesrc->stream_connected = TRUE;
/* store the source output index so it can be accessed via a property */
pulsesrc->source_output_idx = pa_stream_get_index (pulsesrc->stream);
g_object_notify (G_OBJECT (pulsesrc), "source-output-index");
if (pulsesrc->volume_set) {
gst_pulsesrc_set_stream_volume (pulsesrc, pulsesrc->volume);
pulsesrc->volume_set = FALSE;
}
/* get the actual buffering properties now */
actual = pa_stream_get_buffer_attr (pulsesrc->stream);
GST_INFO_OBJECT (pulsesrc, "maxlength: %d", actual->maxlength);
GST_INFO_OBJECT (pulsesrc, "tlength: %d (wanted: %d)",
actual->tlength, wanted.tlength);
GST_INFO_OBJECT (pulsesrc, "prebuf: %d", actual->prebuf);
GST_INFO_OBJECT (pulsesrc, "minreq: %d (wanted %d)", actual->minreq,
wanted.minreq);
GST_INFO_OBJECT (pulsesrc, "fragsize: %d (wanted %d)",
actual->fragsize, wanted.fragsize);
if (actual->fragsize >= wanted.fragsize) {
spec->segsize = actual->fragsize;
} else {
spec->segsize = actual->fragsize * (wanted.fragsize / actual->fragsize);
}
spec->segtotal = actual->maxlength / spec->segsize;
if (!pulsesrc->paused) {
GST_DEBUG_OBJECT (pulsesrc, "uncorking because we are playing");
gst_pulsesrc_set_corked (pulsesrc, FALSE, FALSE);
}
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return TRUE;
/* ERRORS */
connect_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
stream_is_bad:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
unlock_and_fail:
{
gst_pulsesrc_destroy_stream (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return FALSE;
}
}
static ALCboolean pulse_reset_playback(ALCdevice *device) //{{{
{
pulse_data *data = device->ExtraData;
pa_stream_flags_t flags = 0;
pa_stream_state_t state;
pa_channel_map chanmap;
ppa_threaded_mainloop_lock(data->loop);
if(!ConfigValueExists(NULL, "format"))
{
pa_operation *o;
struct {
pa_threaded_mainloop *loop;
char *name;
} server_data;
server_data.loop = data->loop;
server_data.name = NULL;
o = ppa_context_get_server_info(data->context, server_info_callback, &server_data);
while(ppa_operation_get_state(o) == PA_OPERATION_RUNNING)
ppa_threaded_mainloop_wait(data->loop);
ppa_operation_unref(o);
if(server_data.name)
{
o = ppa_context_get_sink_info_by_name(data->context, server_data.name, sink_info_callback, device);
while(ppa_operation_get_state(o) == PA_OPERATION_RUNNING)
ppa_threaded_mainloop_wait(data->loop);
ppa_operation_unref(o);
free(server_data.name);
}
}
if(!ConfigValueExists(NULL, "frequency"))
flags |= PA_STREAM_FIX_RATE;
data->frame_size = aluBytesFromFormat(device->Format) *
aluChannelsFromFormat(device->Format);
data->stream_name = "Playback Stream";
data->attr.minreq = -1;
data->attr.prebuf = -1;
data->attr.fragsize = -1;
data->attr.tlength = device->UpdateSize * device->NumUpdates *
data->frame_size;
data->attr.maxlength = data->attr.tlength;
switch(aluBytesFromFormat(device->Format))
{
case 1:
data->spec.format = PA_SAMPLE_U8;
break;
case 2:
data->spec.format = PA_SAMPLE_S16NE;
break;
case 4:
data->spec.format = PA_SAMPLE_FLOAT32NE;
break;
default:
AL_PRINT("Unknown format: 0x%x\n", device->Format);
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
data->spec.rate = device->Frequency;
data->spec.channels = aluChannelsFromFormat(device->Format);
if(ppa_sample_spec_valid(&data->spec) == 0)
{
AL_PRINT("Invalid sample format\n");
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
if(!ppa_channel_map_init_auto(&chanmap, data->spec.channels, PA_CHANNEL_MAP_WAVEEX))
{
AL_PRINT("Couldn't build map for channel count (%d)!\n", data->spec.channels);
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
SetDefaultWFXChannelOrder(device);
data->stream = ppa_stream_new(data->context, data->stream_name, &data->spec, &chanmap);
if(!data->stream)
{
AL_PRINT("pa_stream_new() failed: %s\n",
ppa_strerror(ppa_context_errno(data->context)));
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
ppa_stream_set_state_callback(data->stream, stream_state_callback, device);
if(ppa_stream_connect_playback(data->stream, NULL, &data->attr, flags, NULL, NULL) < 0)
{
AL_PRINT("Stream did not connect: %s\n",
ppa_strerror(ppa_context_errno(data->context)));
ppa_stream_unref(data->stream);
data->stream = NULL;
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
while((state=ppa_stream_get_state(data->stream)) != PA_STREAM_READY)
{
if(!PA_STREAM_IS_GOOD(state))
{
AL_PRINT("Stream did not get ready: %s\n",
ppa_strerror(ppa_context_errno(data->context)));
ppa_stream_unref(data->stream);
data->stream = NULL;
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
ppa_threaded_mainloop_wait(data->loop);
}
ppa_stream_set_state_callback(data->stream, stream_state_callback2, device);
data->spec = *(ppa_stream_get_sample_spec(data->stream));
if(device->Frequency != data->spec.rate)
{
pa_operation *o;
/* Server updated our playback rate, so modify the buffer attribs
* accordingly. */
data->attr.tlength = (ALuint64)(data->attr.tlength/data->frame_size) *
data->spec.rate / device->Frequency * data->frame_size;
data->attr.maxlength = data->attr.tlength;
o = ppa_stream_set_buffer_attr(data->stream, &data->attr,
stream_success_callback, device);
while(ppa_operation_get_state(o) == PA_OPERATION_RUNNING)
ppa_threaded_mainloop_wait(data->loop);
ppa_operation_unref(o);
device->Frequency = data->spec.rate;
}
stream_buffer_attr_callback(data->stream, device);
#if PA_CHECK_VERSION(0,9,15)
if(ppa_stream_set_buffer_attr_callback)
ppa_stream_set_buffer_attr_callback(data->stream, stream_buffer_attr_callback, device);
#endif
stream_write_callback(data->stream, data->attr.tlength, device);
ppa_stream_set_write_callback(data->stream, stream_write_callback, device);
ppa_threaded_mainloop_unlock(data->loop);
return ALC_TRUE;
} //}}}
static ALCboolean pulse_reset_playback(ALCdevice *device) //{{{
{
pulse_data *data = device->ExtraData;
pa_stream_state_t state;
pa_channel_map chanmap;
ppa_threaded_mainloop_lock(data->loop);
data->frame_size = aluBytesFromFormat(device->Format) *
aluChannelsFromFormat(device->Format);
data->attr.minreq = data->frame_size * device->UpdateSize;
data->attr.prebuf = -1;
data->attr.maxlength = -1;
data->attr.fragsize = -1;
data->attr.tlength = data->attr.minreq * device->NumUpdates;
data->stream_name = "Playback Stream";
switch(aluBytesFromFormat(device->Format))
{
case 1:
data->spec.format = PA_SAMPLE_U8;
break;
case 2:
data->spec.format = PA_SAMPLE_S16NE;
break;
case 4:
data->spec.format = PA_SAMPLE_FLOAT32NE;
break;
default:
AL_PRINT("Unknown format: 0x%x\n", device->Format);
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
data->spec.rate = device->Frequency;
data->spec.channels = aluChannelsFromFormat(device->Format);
if(ppa_sample_spec_valid(&data->spec) == 0)
{
AL_PRINT("Invalid sample format\n");
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
#ifdef _WIN32
if(!ppa_channel_map_init_auto(&chanmap, data->spec.channels, PA_CHANNEL_MAP_WAVEEX))
{
AL_PRINT("Couldn't build map for channel count (%d)!", data->spec.channels);
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
#else
switch(data->spec.channels)
{
case 1:
ppa_channel_map_parse(&chanmap, "mono");
break;
case 2:
ppa_channel_map_parse(&chanmap, "front-left,front-right");
break;
case 4:
ppa_channel_map_parse(&chanmap, "front-left,front-right,rear-left,rear-right");
break;
case 6:
ppa_channel_map_parse(&chanmap, "front-left,front-right,rear-left,rear-right,front-center,lfe");
break;
case 7:
ppa_channel_map_parse(&chanmap, "front-left,front-right,front-center,lfe,rear-center,side-left,side-right");
break;
case 8:
ppa_channel_map_parse(&chanmap, "front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right");
break;
default:
AL_PRINT("Got unhandled channel count (%d)!", data->spec.channels);
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
#endif
data->stream = ppa_stream_new(data->context, data->stream_name, &data->spec, &chanmap);
if(!data->stream)
{
AL_PRINT("pa_stream_new() failed: %s\n",
ppa_strerror(ppa_context_errno(data->context)));
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
ppa_stream_set_state_callback(data->stream, stream_state_callback, device);
ppa_stream_set_write_callback(data->stream, stream_write_callback, device);
if(ppa_stream_connect_playback(data->stream, NULL, &data->attr, PA_STREAM_ADJUST_LATENCY, NULL, NULL) < 0)
{
AL_PRINT("Stream did not connect: %s\n",
ppa_strerror(ppa_context_errno(data->context)));
ppa_stream_unref(data->stream);
data->stream = NULL;
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
while((state=ppa_stream_get_state(data->stream)) != PA_STREAM_READY)
{
if(!PA_STREAM_IS_GOOD(state))
{
AL_PRINT("Stream did not get ready: %s\n",
ppa_strerror(ppa_context_errno(data->context)));
ppa_stream_unref(data->stream);
data->stream = NULL;
ppa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
ppa_threaded_mainloop_wait(data->loop);
ppa_threaded_mainloop_accept(data->loop);
}
ppa_stream_set_state_callback(data->stream, stream_state_callback2, device);
stream_buffer_attr_callback(data->stream, device);
ppa_stream_set_buffer_attr_callback(data->stream, stream_buffer_attr_callback, device);
ppa_threaded_mainloop_unlock(data->loop);
return ALC_TRUE;
} //}}}
static void rdpsnd_pulse_open(rdpsndDevicePlugin* device, rdpsndFormat* format)
{
rdpsndPulsePlugin* pulse = (rdpsndPulsePlugin*)device;
pa_stream_state_t state;
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);
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);
if (pa_stream_connect_playback(pulse->stream,
pulse->device_name, NULL, PA_STREAM_INTERPOLATE_TIMING|PA_STREAM_AUTO_TIMING_UPDATE, 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)
{
memset(&pulse->adpcm, 0, sizeof(ADPCM));
DEBUG_SVC("connected");
}
else
{
rdpsnd_pulse_close(device);
}
}
static int init(struct ao *ao)
{
pa_proplist *proplist = NULL;
pa_format_info *format = NULL;
struct priv *priv = ao->priv;
char *sink = priv->cfg_sink && priv->cfg_sink[0] ? priv->cfg_sink : ao->device;
if (pa_init_boilerplate(ao) < 0)
return -1;
pa_threaded_mainloop_lock(priv->mainloop);
if (!(proplist = pa_proplist_new())) {
MP_ERR(ao, "Failed to allocate proplist\n");
goto unlock_and_fail;
}
(void)pa_proplist_sets(proplist, PA_PROP_MEDIA_ICON_NAME, ao->client_name);
if (!(format = pa_format_info_new()))
goto unlock_and_fail;
if (!set_format(ao, format)) {
ao->channels = (struct mp_chmap) MP_CHMAP_INIT_STEREO;
ao->samplerate = 48000;
ao->format = AF_FORMAT_FLOAT;
if (!set_format(ao, format)) {
MP_ERR(ao, "Invalid audio format\n");
goto unlock_and_fail;
}
}
if (!(priv->stream = pa_stream_new_extended(priv->context, "audio stream",
&format, 1, proplist)))
goto unlock_and_fail;
pa_format_info_free(format);
format = NULL;
pa_proplist_free(proplist);
proplist = NULL;
pa_stream_set_state_callback(priv->stream, stream_state_cb, ao);
pa_stream_set_write_callback(priv->stream, stream_request_cb, ao);
pa_stream_set_latency_update_callback(priv->stream,
stream_latency_update_cb, ao);
int buf_size = af_fmt_seconds_to_bytes(ao->format, priv->cfg_buffer / 1000.0,
ao->channels.num, ao->samplerate);
pa_buffer_attr bufattr = {
.maxlength = -1,
.tlength = buf_size > 0 ? buf_size : (uint32_t)-1,
.prebuf = -1,
.minreq = -1,
.fragsize = -1,
};
int flags = PA_STREAM_NOT_MONOTONIC;
if (!priv->cfg_latency_hacks)
flags |= PA_STREAM_INTERPOLATE_TIMING|PA_STREAM_AUTO_TIMING_UPDATE;
if (pa_stream_connect_playback(priv->stream, sink, &bufattr,
flags, NULL, NULL) < 0)
goto unlock_and_fail;
/* Wait until the stream is ready */
while (1) {
int state = pa_stream_get_state(priv->stream);
if (state == PA_STREAM_READY)
break;
if (!PA_STREAM_IS_GOOD(state))
goto unlock_and_fail;
pa_threaded_mainloop_wait(priv->mainloop);
}
if (pa_stream_is_suspended(priv->stream)) {
MP_ERR(ao, "The stream is suspended. Bailing out.\n");
goto unlock_and_fail;
}
pa_threaded_mainloop_unlock(priv->mainloop);
return 0;
unlock_and_fail:
pa_threaded_mainloop_unlock(priv->mainloop);
if (format)
pa_format_info_free(format);
if (proplist)
pa_proplist_free(proplist);
uninit(ao);
return -1;
}
static void cork(struct ao *ao, bool pause)
{
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
priv->retval = 0;
if (!waitop(priv, pa_stream_cork(priv->stream, pause, success_cb, ao)) ||
!priv->retval)
GENERIC_ERR_MSG("pa_stream_cork() failed");
}
// Play the specified data to the pulseaudio server
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
if (pa_stream_write(priv->stream, data[0], samples * ao->sstride, NULL, 0,
PA_SEEK_RELATIVE) < 0) {
GENERIC_ERR_MSG("pa_stream_write() failed");
samples = -1;
}
if (flags & AOPLAY_FINAL_CHUNK) {
// Force start in case the stream was too short for prebuf
pa_operation *op = pa_stream_trigger(priv->stream, NULL, NULL);
pa_operation_unref(op);
}
pa_threaded_mainloop_unlock(priv->mainloop);
return samples;
}
// Reset the audio stream, i.e. flush the playback buffer on the server side
static void reset(struct ao *ao)
{
// pa_stream_flush() works badly if not corked
cork(ao, true);
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
priv->retval = 0;
if (!waitop(priv, pa_stream_flush(priv->stream, success_cb, ao)) ||
!priv->retval)
GENERIC_ERR_MSG("pa_stream_flush() failed");
cork(ao, false);
}
static int
rdpsnd_pulse_open(rdpsndDevicePlugin * devplugin)
{
struct pulse_device_data * pulse_data;
pa_stream_state_t state;
pa_buffer_attr buffer_attr = { 0 };
pulse_data = (struct pulse_device_data *) devplugin->device_data;
if (!pulse_data->context)
return 1;
/* Since rdpsnd_main calls set_format() after open(), but we need the
format spec to open the stream, we will defer the open request if
initial set_format request is not yet received */
if (!pulse_data->sample_spec.rate || pulse_data->stream)
return 0;
LLOGLN(10, ("rdpsnd_pulse_open:"));
pa_threaded_mainloop_lock(pulse_data->mainloop);
pulse_data->stream = pa_stream_new(pulse_data->context, "freerdp",
&pulse_data->sample_spec, NULL);
if (!pulse_data->stream)
{
pa_threaded_mainloop_unlock(pulse_data->mainloop);
LLOGLN(0, ("rdpsnd_pulse_open: pa_stream_new failed (%d)",
pa_context_errno(pulse_data->context)));
return 1;
}
pa_stream_set_state_callback(pulse_data->stream,
rdpsnd_pulse_stream_state_callback, devplugin);
pa_stream_set_write_callback(pulse_data->stream,
rdpsnd_pulse_stream_request_callback, devplugin);
buffer_attr.maxlength = (uint32_t) -1;
buffer_attr.tlength = (uint32_t) -1;//pa_usec_to_bytes(2000000, &pulse_data->sample_spec);
buffer_attr.prebuf = (uint32_t) -1;
buffer_attr.minreq = (uint32_t) -1;
buffer_attr.fragsize = (uint32_t) -1;
if (pa_stream_connect_playback(pulse_data->stream,
pulse_data->device_name[0] ? pulse_data->device_name : NULL,
&buffer_attr, 0, NULL, NULL) < 0)
{
pa_threaded_mainloop_unlock(pulse_data->mainloop);
LLOGLN(0, ("rdpsnd_pulse_open: pa_stream_connect_playback failed (%d)",
pa_context_errno(pulse_data->context)));
return 1;
}
for (;;)
{
state = pa_stream_get_state(pulse_data->stream);
if (state == PA_STREAM_READY)
break;
if (!PA_STREAM_IS_GOOD(state))
{
LLOGLN(0, ("rdpsnd_pulse_open: bad stream state (%d)",
pa_context_errno(pulse_data->context)));
break;
}
pa_threaded_mainloop_wait(pulse_data->mainloop);
}
pa_threaded_mainloop_unlock(pulse_data->mainloop);
if (state == PA_STREAM_READY)
{
memset(&pulse_data->adpcm, 0, sizeof(rdpsndDspAdpcm));
LLOGLN(0, ("rdpsnd_pulse_open: connected"));
return 0;
}
else
{
rdpsnd_pulse_close(devplugin);
return 1;
}
}
static ALCboolean pulse_open_capture(ALCdevice *device, const ALCchar *device_name) //{{{
{
pulse_data *data;
pa_stream_state_t state;
if(!pa_handle)
return ALC_FALSE;
if(!device_name)
device_name = pulse_capture_device;
else if(strcmp(device_name, pulse_capture_device) != 0)
return ALC_FALSE;
if(pulse_open(device, device_name) == ALC_FALSE)
return ALC_FALSE;
data = device->ExtraData;
ppa_threaded_mainloop_lock(data->loop);
data->samples = device->UpdateSize * device->NumUpdates;
data->frame_size = aluBytesFromFormat(device->Format) *
aluChannelsFromFormat(device->Format);
if(!(data->ring = CreateRingBuffer(data->frame_size, data->samples)))
{
ppa_threaded_mainloop_unlock(data->loop);
pulse_close(device);
return ALC_FALSE;
}
data->attr.minreq = -1;
data->attr.prebuf = -1;
data->attr.maxlength = -1;
data->attr.tlength = -1;
data->attr.fragsize = data->frame_size * data->samples / 2;
data->stream_name = "Capture Stream";
data->spec.rate = device->Frequency;
data->spec.channels = aluChannelsFromFormat(device->Format);
switch(aluBytesFromFormat(device->Format))
{
case 1:
data->spec.format = PA_SAMPLE_U8;
break;
case 2:
data->spec.format = PA_SAMPLE_S16NE;
break;
case 4:
data->spec.format = PA_SAMPLE_FLOAT32NE;
break;
default:
AL_PRINT("Unknown format: 0x%x\n", device->Format);
ppa_threaded_mainloop_unlock(data->loop);
pulse_close(device);
return ALC_FALSE;
}
if(ppa_sample_spec_valid(&data->spec) == 0)
{
AL_PRINT("Invalid sample format\n");
ppa_threaded_mainloop_unlock(data->loop);
pulse_close(device);
return ALC_FALSE;
}
data->stream = ppa_stream_new(data->context, data->stream_name, &data->spec, NULL);
if(!data->stream)
{
AL_PRINT("pa_stream_new() failed: %s\n",
ppa_strerror(ppa_context_errno(data->context)));
ppa_threaded_mainloop_unlock(data->loop);
pulse_close(device);
return ALC_FALSE;
}
if(ppa_stream_connect_record(data->stream, NULL, &data->attr, PA_STREAM_ADJUST_LATENCY) < 0)
{
AL_PRINT("Stream did not connect: %s\n",
ppa_strerror(ppa_context_errno(data->context)));
ppa_stream_unref(data->stream);
ppa_threaded_mainloop_unlock(data->loop);
data->stream = NULL;
pulse_close(device);
return ALC_FALSE;
}
while((state=ppa_stream_get_state(data->stream)) != PA_STREAM_READY)
{
if(!PA_STREAM_IS_GOOD(state))
{
AL_PRINT("Stream did not get ready: %s\n",
ppa_strerror(ppa_context_errno(data->context)));
ppa_stream_unref(data->stream);
ppa_threaded_mainloop_unlock(data->loop);
data->stream = NULL;
pulse_close(device);
return ALC_FALSE;
}
ppa_threaded_mainloop_unlock(data->loop);
Sleep(1);
ppa_threaded_mainloop_lock(data->loop);
}
ppa_threaded_mainloop_unlock(data->loop);
return ALC_TRUE;
} //}}}
static ALCenum pulse_open_capture(ALCdevice *device, const ALCchar *device_name) //{{{
{
char *pulse_name = NULL;
pulse_data *data;
pa_stream_flags_t flags = 0;
pa_stream_state_t state;
pa_channel_map chanmap;
if(!allCaptureDevNameMap)
probe_devices(AL_TRUE);
if(!device_name)
device_name = pulse_device;
else if(strcmp(device_name, pulse_device) != 0)
{
ALuint i;
for(i = 0;i < numCaptureDevNames;i++)
{
if(strcmp(device_name, allCaptureDevNameMap[i].name) == 0)
{
pulse_name = allCaptureDevNameMap[i].device_name;
break;
}
}
if(i == numCaptureDevNames)
return ALC_INVALID_VALUE;
}
if(pulse_open(device, device_name) == ALC_FALSE)
return ALC_INVALID_VALUE;
data = device->ExtraData;
pa_threaded_mainloop_lock(data->loop);
data->samples = device->UpdateSize * device->NumUpdates;
data->frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
data->samples = maxu(data->samples, 100 * device->Frequency / 1000);
if(!(data->ring = CreateRingBuffer(data->frame_size, data->samples)))
{
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
data->attr.minreq = -1;
data->attr.prebuf = -1;
data->attr.maxlength = data->samples * data->frame_size;
data->attr.tlength = -1;
data->attr.fragsize = minu(data->samples, 50*device->Frequency/1000) *
data->frame_size;
data->spec.rate = device->Frequency;
data->spec.channels = ChannelsFromDevFmt(device->FmtChans);
switch(device->FmtType)
{
case DevFmtUByte:
data->spec.format = PA_SAMPLE_U8;
break;
case DevFmtShort:
data->spec.format = PA_SAMPLE_S16NE;
break;
case DevFmtFloat:
data->spec.format = PA_SAMPLE_FLOAT32NE;
break;
case DevFmtByte:
case DevFmtUShort:
ERR("Capture format type %#x capture not supported on PulseAudio\n", device->FmtType);
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
if(pa_sample_spec_valid(&data->spec) == 0)
{
ERR("Invalid sample format\n");
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
if(!pa_channel_map_init_auto(&chanmap, data->spec.channels, PA_CHANNEL_MAP_WAVEEX))
{
ERR("Couldn't build map for channel count (%d)!\n", data->spec.channels);
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
data->stream = pa_stream_new(data->context, "Capture Stream", &data->spec, &chanmap);
if(!data->stream)
{
ERR("pa_stream_new() failed: %s\n",
pa_strerror(pa_context_errno(data->context)));
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
pa_stream_set_state_callback(data->stream, stream_state_callback, data->loop);
flags |= PA_STREAM_START_CORKED|PA_STREAM_ADJUST_LATENCY;
if(pa_stream_connect_record(data->stream, pulse_name, &data->attr, flags) < 0)
{
ERR("Stream did not connect: %s\n",
pa_strerror(pa_context_errno(data->context)));
pa_stream_unref(data->stream);
data->stream = NULL;
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
while((state=pa_stream_get_state(data->stream)) != PA_STREAM_READY)
{
if(!PA_STREAM_IS_GOOD(state))
{
ERR("Stream did not get ready: %s\n",
pa_strerror(pa_context_errno(data->context)));
pa_stream_unref(data->stream);
data->stream = NULL;
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
pa_threaded_mainloop_wait(data->loop);
}
pa_stream_set_state_callback(data->stream, stream_state_callback2, device);
pa_threaded_mainloop_unlock(data->loop);
return ALC_NO_ERROR;
fail:
pulse_close(device);
return ALC_INVALID_VALUE;
} //}}}
static int PULSE_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
int state;
Uint16 test_format;
pa_sample_spec paspec;
pa_buffer_attr paattr;
pa_channel_map pacmap;
pa_stream_flags_t flags = 0;
paspec.format = PA_SAMPLE_INVALID;
for ( test_format = SDL_FirstAudioFormat(spec->format); test_format; ) {
switch ( test_format ) {
case AUDIO_U8:
paspec.format = PA_SAMPLE_U8;
break;
case AUDIO_S16LSB:
paspec.format = PA_SAMPLE_S16LE;
break;
case AUDIO_S16MSB:
paspec.format = PA_SAMPLE_S16BE;
break;
}
if ( paspec.format != PA_SAMPLE_INVALID )
break;
}
if (paspec.format == PA_SAMPLE_INVALID ) {
SDL_SetError("Couldn't find any suitable audio formats");
return(-1);
}
spec->format = test_format;
paspec.channels = spec->channels;
paspec.rate = spec->freq;
/* Calculate the final parameters for this audio specification */
#ifdef PA_STREAM_ADJUST_LATENCY
spec->samples /= 2; /* Mix in smaller chunck to avoid underruns */
#endif
SDL_CalculateAudioSpec(spec);
/* Allocate mixing buffer */
mixlen = spec->size;
mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
if ( mixbuf == NULL ) {
return(-1);
}
SDL_memset(mixbuf, spec->silence, spec->size);
/* Reduced prebuffering compared to the defaults. */
#ifdef PA_STREAM_ADJUST_LATENCY
paattr.tlength = mixlen * 4; /* 2x original requested bufsize */
paattr.prebuf = -1;
paattr.maxlength = -1;
paattr.minreq = mixlen; /* -1 can lead to pa_stream_writable_size()
>= mixlen never becoming true */
flags = PA_STREAM_ADJUST_LATENCY;
#else
paattr.tlength = mixlen*2;
paattr.prebuf = mixlen*2;
paattr.maxlength = mixlen*2;
paattr.minreq = mixlen;
#endif
/* The SDL ALSA output hints us that we use Windows' channel mapping */
/* http://bugzilla.libsdl.org/show_bug.cgi?id=110 */
SDL_NAME(pa_channel_map_init_auto)(
&pacmap, spec->channels, PA_CHANNEL_MAP_WAVEEX);
/* Set up a new main loop */
if (!(mainloop = SDL_NAME(pa_mainloop_new)())) {
PULSE_CloseAudio(this);
SDL_SetError("pa_mainloop_new() failed");
return(-1);
}
mainloop_api = SDL_NAME(pa_mainloop_get_api)(mainloop);
if (!(context = SDL_NAME(pa_context_new)(mainloop_api, get_progname()))) {
PULSE_CloseAudio(this);
SDL_SetError("pa_context_new() failed");
return(-1);
}
/* Connect to the PulseAudio server */
if (SDL_NAME(pa_context_connect)(context, NULL, 0, NULL) < 0) {
PULSE_CloseAudio(this);
SDL_SetError("Could not setup connection to PulseAudio");
return(-1);
}
do {
if (SDL_NAME(pa_mainloop_iterate)(mainloop, 1, NULL) < 0) {
PULSE_CloseAudio(this);
SDL_SetError("pa_mainloop_iterate() failed");
return(-1);
}
state = SDL_NAME(pa_context_get_state)(context);
if (!PA_CONTEXT_IS_GOOD(state)) {
PULSE_CloseAudio(this);
SDL_SetError("Could not connect to PulseAudio");
return(-1);
}
} while (state != PA_CONTEXT_READY);
stream = SDL_NAME(pa_stream_new)(
context,
"Simple DirectMedia Layer", /* stream description */
&paspec, /* sample format spec */
&pacmap /* channel map */
);
if ( stream == NULL ) {
PULSE_CloseAudio(this);
SDL_SetError("Could not setup PulseAudio stream");
return(-1);
}
if (SDL_NAME(pa_stream_connect_playback)(stream, NULL, &paattr, flags,
NULL, NULL) < 0) {
PULSE_CloseAudio(this);
SDL_SetError("Could not connect PulseAudio stream");
return(-1);
}
do {
if (SDL_NAME(pa_mainloop_iterate)(mainloop, 1, NULL) < 0) {
PULSE_CloseAudio(this);
SDL_SetError("pa_mainloop_iterate() failed");
return(-1);
}
state = SDL_NAME(pa_stream_get_state)(stream);
if (!PA_STREAM_IS_GOOD(state)) {
PULSE_CloseAudio(this);
SDL_SetError("Could not create to PulseAudio stream");
return(-1);
}
} while (state != PA_STREAM_READY);
return(0);
}
static int PULSE_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
int state;
Uint16 test_format;
pa_sample_spec paspec;
pa_buffer_attr paattr;
pa_channel_map pacmap;
pa_stream_flags_t flags = 0;
paspec.format = PA_SAMPLE_INVALID;
for ( test_format = SDL_FirstAudioFormat(spec->format); test_format; ) {
switch ( test_format ) {
case AUDIO_U8:
paspec.format = PA_SAMPLE_U8;
break;
case AUDIO_S16LSB:
paspec.format = PA_SAMPLE_S16LE;
break;
case AUDIO_S16MSB:
paspec.format = PA_SAMPLE_S16BE;
break;
}
if ( paspec.format != PA_SAMPLE_INVALID )
break;
test_format = SDL_NextAudioFormat();
}
if (paspec.format == PA_SAMPLE_INVALID ) {
SDL_SetError("Couldn't find any suitable audio formats");
return(-1);
}
spec->format = test_format;
paspec.channels = spec->channels;
paspec.rate = spec->freq;
#ifdef PA_STREAM_ADJUST_LATENCY
spec->samples /= 2;
#endif
SDL_CalculateAudioSpec(spec);
mixlen = spec->size;
mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
if ( mixbuf == NULL ) {
return(-1);
}
SDL_memset(mixbuf, spec->silence, spec->size);
#ifdef PA_STREAM_ADJUST_LATENCY
paattr.tlength = mixlen * 4;
paattr.prebuf = -1;
paattr.maxlength = -1;
paattr.minreq = mixlen;
flags = PA_STREAM_ADJUST_LATENCY;
#else
paattr.tlength = mixlen*2;
paattr.prebuf = mixlen*2;
paattr.maxlength = mixlen*2;
paattr.minreq = mixlen;
#endif
SDL_NAME(pa_channel_map_init_auto)(
&pacmap, spec->channels, PA_CHANNEL_MAP_WAVEEX);
if (!(mainloop = SDL_NAME(pa_mainloop_new)())) {
PULSE_CloseAudio(this);
SDL_SetError("pa_mainloop_new() failed");
return(-1);
}
if (this->hidden->caption == NULL) {
char *title = NULL;
SDL_WM_GetCaption(&title, NULL);
PULSE_SetCaption(this, title);
}
mainloop_api = SDL_NAME(pa_mainloop_get_api)(mainloop);
if (!(context = SDL_NAME(pa_context_new)(mainloop_api,
this->hidden->caption))) {
PULSE_CloseAudio(this);
SDL_SetError("pa_context_new() failed");
return(-1);
}
if (SDL_NAME(pa_context_connect)(context, NULL, 0, NULL) < 0) {
PULSE_CloseAudio(this);
SDL_SetError("Could not setup connection to PulseAudio");
return(-1);
}
do {
if (SDL_NAME(pa_mainloop_iterate)(mainloop, 1, NULL) < 0) {
PULSE_CloseAudio(this);
SDL_SetError("pa_mainloop_iterate() failed");
return(-1);
}
state = SDL_NAME(pa_context_get_state)(context);
if (!PA_CONTEXT_IS_GOOD(state)) {
PULSE_CloseAudio(this);
SDL_SetError("Could not connect to PulseAudio");
return(-1);
}
} while (state != PA_CONTEXT_READY);
stream = SDL_NAME(pa_stream_new)(
context,
"Simple DirectMedia Layer",
&paspec,
&pacmap
);
if ( stream == NULL ) {
PULSE_CloseAudio(this);
SDL_SetError("Could not setup PulseAudio stream");
return(-1);
}
if (SDL_NAME(pa_stream_connect_playback)(stream, NULL, &paattr, flags,
NULL, NULL) < 0) {
PULSE_CloseAudio(this);
SDL_SetError("Could not connect PulseAudio stream");
return(-1);
}
do {
if (SDL_NAME(pa_mainloop_iterate)(mainloop, 1, NULL) < 0) {
PULSE_CloseAudio(this);
SDL_SetError("pa_mainloop_iterate() failed");
return(-1);
}
state = SDL_NAME(pa_stream_get_state)(stream);
if (!PA_STREAM_IS_GOOD(state)) {
PULSE_CloseAudio(this);
SDL_SetError("Could not create to PulseAudio stream");
return(-1);
}
} while (state != PA_STREAM_READY);
return(0);
}
static ALCboolean pulse_open_capture( ALCdevice* device, const ALCchar* device_name ) //{{{
{
char* pulse_name = NULL;
pulse_data* data;
pa_stream_flags_t flags = 0;
pa_stream_state_t state;
pa_channel_map chanmap;
if ( !pulse_load() )
{
return ALC_FALSE;
}
if ( !allCaptureDevNameMap )
{
probe_devices( AL_TRUE );
}
if ( !device_name )
{
device_name = allCaptureDevNameMap[0].name;
}
else
{
ALuint i;
for ( i = 0; i < numCaptureDevNames; i++ )
{
if ( strcmp( device_name, allCaptureDevNameMap[i].name ) == 0 )
{
pulse_name = allCaptureDevNameMap[i].device_name;
break;
}
}
if ( i == numCaptureDevNames )
{
return ALC_FALSE;
}
}
if ( pulse_open( device, device_name ) == ALC_FALSE )
{
return ALC_FALSE;
}
data = device->ExtraData;
ppa_threaded_mainloop_lock( data->loop );
data->samples = device->UpdateSize * device->NumUpdates;
data->frame_size = aluFrameSizeFromFormat( device->Format );
if ( !( data->ring = CreateRingBuffer( data->frame_size, data->samples ) ) )
{
ppa_threaded_mainloop_unlock( data->loop );
goto fail;
}
data->attr.minreq = -1;
data->attr.prebuf = -1;
data->attr.maxlength = data->frame_size * data->samples;
data->attr.tlength = -1;
data->attr.fragsize = min( data->frame_size * data->samples,
10 * device->Frequency / 1000 );
data->spec.rate = device->Frequency;
data->spec.channels = aluChannelsFromFormat( device->Format );
switch ( aluBytesFromFormat( device->Format ) )
{
case 1:
data->spec.format = PA_SAMPLE_U8;
break;
case 2:
data->spec.format = PA_SAMPLE_S16NE;
break;
case 4:
data->spec.format = PA_SAMPLE_FLOAT32NE;
break;
default:
AL_PRINT( "Unknown format: 0x%x\n", device->Format );
ppa_threaded_mainloop_unlock( data->loop );
goto fail;
}
if ( ppa_sample_spec_valid( &data->spec ) == 0 )
{
AL_PRINT( "Invalid sample format\n" );
ppa_threaded_mainloop_unlock( data->loop );
goto fail;
}
if ( !ppa_channel_map_init_auto( &chanmap, data->spec.channels, PA_CHANNEL_MAP_WAVEEX ) )
{
AL_PRINT( "Couldn't build map for channel count (%d)!\n", data->spec.channels );
ppa_threaded_mainloop_unlock( data->loop );
goto fail;
}
data->stream = ppa_stream_new( data->context, "Capture Stream", &data->spec, &chanmap );
if ( !data->stream )
{
AL_PRINT( "pa_stream_new() failed: %s\n",
ppa_strerror( ppa_context_errno( data->context ) ) );
ppa_threaded_mainloop_unlock( data->loop );
goto fail;
}
ppa_stream_set_state_callback( data->stream, stream_state_callback, data->loop );
flags |= PA_STREAM_START_CORKED | PA_STREAM_ADJUST_LATENCY;
if ( ppa_stream_connect_record( data->stream, pulse_name, &data->attr, flags ) < 0 )
{
AL_PRINT( "Stream did not connect: %s\n",
ppa_strerror( ppa_context_errno( data->context ) ) );
ppa_stream_unref( data->stream );
data->stream = NULL;
ppa_threaded_mainloop_unlock( data->loop );
goto fail;
}
while ( ( state = ppa_stream_get_state( data->stream ) ) != PA_STREAM_READY )
{
if ( !PA_STREAM_IS_GOOD( state ) )
{
AL_PRINT( "Stream did not get ready: %s\n",
ppa_strerror( ppa_context_errno( data->context ) ) );
ppa_stream_unref( data->stream );
data->stream = NULL;
ppa_threaded_mainloop_unlock( data->loop );
goto fail;
}
ppa_threaded_mainloop_wait( data->loop );
}
ppa_stream_set_state_callback( data->stream, stream_state_callback2, device );
ppa_threaded_mainloop_unlock( data->loop );
return ALC_TRUE;
fail:
pulse_close( device );
return ALC_FALSE;
} //}}}
static int
tsmf_pulse_open_stream(TSMFPulseAudioDevice * pulse)
{
pa_stream_state_t state;
pa_buffer_attr buffer_attr = { 0 };
if (!pulse->context)
return 1;
LLOGLN(0, ("tsmf_pulse_open_stream:"));
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);
LLOGLN(0, ("tsmf_pulse_open_stream: pa_stream_new failed (%d)",
pa_context_errno(pulse->context)));
return 1;
}
pa_stream_set_state_callback(pulse->stream,
tsmf_pulse_stream_state_callback, pulse);
pa_stream_set_write_callback(pulse->stream,
tsmf_pulse_stream_request_callback, pulse);
buffer_attr.maxlength = pa_usec_to_bytes(500000, &pulse->sample_spec);
buffer_attr.tlength = pa_usec_to_bytes(250000, &pulse->sample_spec);
buffer_attr.prebuf = (uint32_t) -1;
buffer_attr.minreq = (uint32_t) -1;
buffer_attr.fragsize = (uint32_t) -1;
if (pa_stream_connect_playback(pulse->stream,
pulse->device[0] ? pulse->device : NULL, &buffer_attr,
PA_STREAM_ADJUST_LATENCY | PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE,
NULL, NULL) < 0)
{
pa_threaded_mainloop_unlock(pulse->mainloop);
LLOGLN(0, ("tsmf_pulse_open_stream: pa_stream_connect_playback failed (%d)",
pa_context_errno(pulse->context)));
return 1;
}
for (;;)
{
state = pa_stream_get_state(pulse->stream);
if (state == PA_STREAM_READY)
break;
if (!PA_STREAM_IS_GOOD(state))
{
LLOGLN(0, ("tsmf_pulse_open_stream: 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)
{
LLOGLN(0, ("tsmf_pulse_open_stream: connected"));
return 0;
}
else
{
tsmf_pulse_close_stream(pulse);
return 1;
}
}
pa_simple* pa_simple_new(
const char *server,
const char *name,
pa_stream_direction_t dir,
const char *dev,
const char *stream_name,
const pa_sample_spec *ss,
const pa_channel_map *map,
const pa_buffer_attr *attr,
int *rerror) {
pa_simple *p;
int error = PA_ERR_INTERNAL, r;
CHECK_VALIDITY_RETURN_ANY(rerror, !server || *server, PA_ERR_INVALID, NULL);
CHECK_VALIDITY_RETURN_ANY(rerror, dir == PA_STREAM_PLAYBACK || dir == PA_STREAM_RECORD, PA_ERR_INVALID, NULL);
CHECK_VALIDITY_RETURN_ANY(rerror, !dev || *dev, PA_ERR_INVALID, NULL);
CHECK_VALIDITY_RETURN_ANY(rerror, ss && pa_sample_spec_valid(ss), PA_ERR_INVALID, NULL);
CHECK_VALIDITY_RETURN_ANY(rerror, !map || (pa_channel_map_valid(map) && map->channels == ss->channels), PA_ERR_INVALID, NULL)
p = pa_xnew0(pa_simple, 1);
p->direction = dir;
if (!(p->mainloop = pa_threaded_mainloop_new()))
goto fail;
if (!(p->context = pa_context_new(pa_threaded_mainloop_get_api(p->mainloop), name)))
goto fail;
pa_context_set_state_callback(p->context, context_state_cb, p);
if (pa_context_connect(p->context, server, 0, NULL) < 0) {
error = pa_context_errno(p->context);
goto fail;
}
pa_threaded_mainloop_lock(p->mainloop);
if (pa_threaded_mainloop_start(p->mainloop) < 0)
goto unlock_and_fail;
for (;;) {
pa_context_state_t state;
state = pa_context_get_state(p->context);
if (state == PA_CONTEXT_READY)
break;
if (!PA_CONTEXT_IS_GOOD(state)) {
error = pa_context_errno(p->context);
goto unlock_and_fail;
}
/* Wait until the context is ready */
pa_threaded_mainloop_wait(p->mainloop);
}
if (!(p->stream = pa_stream_new(p->context, stream_name, ss, map))) {
error = pa_context_errno(p->context);
goto unlock_and_fail;
}
pa_stream_set_state_callback(p->stream, stream_state_cb, p);
pa_stream_set_read_callback(p->stream, stream_request_cb, p);
pa_stream_set_write_callback(p->stream, stream_request_cb, p);
pa_stream_set_latency_update_callback(p->stream, stream_latency_update_cb, p);
if (dir == PA_STREAM_PLAYBACK)
r = pa_stream_connect_playback(p->stream, dev, attr,
PA_STREAM_INTERPOLATE_TIMING
|PA_STREAM_ADJUST_LATENCY
|PA_STREAM_AUTO_TIMING_UPDATE, NULL, NULL);
else
r = pa_stream_connect_record(p->stream, dev, attr,
PA_STREAM_INTERPOLATE_TIMING
|PA_STREAM_ADJUST_LATENCY
|PA_STREAM_AUTO_TIMING_UPDATE);
if (r < 0) {
error = pa_context_errno(p->context);
goto unlock_and_fail;
}
for (;;) {
pa_stream_state_t state;
state = pa_stream_get_state(p->stream);
if (state == PA_STREAM_READY)
break;
if (!PA_STREAM_IS_GOOD(state)) {
error = pa_context_errno(p->context);
goto unlock_and_fail;
}
/* Wait until the stream is ready */
pa_threaded_mainloop_wait(p->mainloop);
}
pa_threaded_mainloop_unlock(p->mainloop);
return p;
unlock_and_fail:
pa_threaded_mainloop_unlock(p->mainloop);
fail:
if (rerror)
*rerror = error;
pa_simple_free(p);
return NULL;
}
