void AudioDriverPulseAudio::detect_channels(bool capture) {
pa_channel_map_init_stereo(capture ? &pa_rec_map : &pa_map);
String device = capture ? capture_device_name : device_name;
if (device == "Default") {
// Get the default output device name
pa_status = 0;
pa_operation *pa_op = pa_context_get_server_info(pa_ctx, &AudioDriverPulseAudio::pa_server_info_cb, (void *)this);
if (pa_op) {
while (pa_status == 0) {
int ret = pa_mainloop_iterate(pa_ml, 1, NULL);
if (ret < 0) {
ERR_PRINT("pa_mainloop_iterate error");
}
}
pa_operation_unref(pa_op);
} else {
ERR_PRINT("pa_context_get_server_info error");
}
}
char dev[1024];
if (device == "Default") {
strcpy(dev, capture ? capture_default_device.utf8().get_data() : default_device.utf8().get_data());
} else {
strcpy(dev, device.utf8().get_data());
}
// Now using the device name get the amount of channels
pa_status = 0;
pa_operation *pa_op;
if (capture) {
pa_op = pa_context_get_source_info_by_name(pa_ctx, dev, &AudioDriverPulseAudio::pa_source_info_cb, (void *)this);
} else {
pa_op = pa_context_get_sink_info_by_name(pa_ctx, dev, &AudioDriverPulseAudio::pa_sink_info_cb, (void *)this);
}
if (pa_op) {
while (pa_status == 0) {
int ret = pa_mainloop_iterate(pa_ml, 1, NULL);
if (ret < 0) {
ERR_PRINT("pa_mainloop_iterate error");
}
}
pa_operation_unref(pa_op);
} else {
if (capture) {
ERR_PRINT("pa_context_get_source_info_by_name error");
} else {
ERR_PRINT("pa_context_get_sink_info_by_name error");
}
}
}
void RageSoundDriver_PulseAudio::m_InitStream(void)
{
int error;
pa_sample_spec ss;
pa_channel_map map;
/* init sample spec */
ss.format = PA_SAMPLE_S16LE;
ss.channels = 2;
ss.rate = PREFSMAN->m_iSoundPreferredSampleRate;
if(ss.rate == 0)
{
ss.rate = 44100;
}
/* init channel map */
pa_channel_map_init_stereo(&map);
/* check sample spec */
if(!pa_sample_spec_valid(&ss))
{
if(asprintf(&m_Error, "invalid sample spec!") == -1)
{
m_Error = nullptr;
}
m_Sem.Post();
return;
}
/* log the used sample spec */
char specstring[PA_SAMPLE_SPEC_SNPRINT_MAX];
pa_sample_spec_snprint(specstring, sizeof(specstring), &ss);
LOG->Trace("Pulse: using sample spec: %s", specstring);
/* create the stream */
LOG->Trace("Pulse: pa_stream_new()...");
m_PulseStream = pa_stream_new(m_PulseCtx, "Stepmania Audio", &ss, &map);
if(m_PulseStream == nullptr)
{
if(asprintf(&m_Error, "pa_stream_new(): %s", pa_strerror(pa_context_errno(m_PulseCtx))) == -1)
{
m_Error = nullptr;
}
m_Sem.Post();
return;
}
/* set the write callback, it will be called when the sound server
* needs data */
pa_stream_set_write_callback(m_PulseStream, StaticStreamWriteCb, this);
/* set the state callback, it will be called the the stream state will
* change */
pa_stream_set_state_callback(m_PulseStream, StaticStreamStateCb, this);
/* configure attributes of the stream */
pa_buffer_attr attr;
memset(&attr, 0x00, sizeof(attr));
/* tlength: Target length of the buffer.
*
* "The server tries to assure that at least tlength bytes are always
* available in the per-stream server-side playback buffer. It is
* recommended to set this to (uint32_t) -1, which will initialize
* this to a value that is deemed sensible by the server. However,
* this value will default to something like 2s, i.e. for applications
* that have specific latency requirements this value should be set to
* the maximum latency that the application can deal with."
*
* We don't want the default here, we want a small latency.
* We use pa_usec_to_bytes() to convert a latency to a buffer size.
*/
attr.tlength = pa_usec_to_bytes(20*PA_USEC_PER_MSEC, &ss);
/* maxlength: Maximum length of the buffer
*
* "Setting this to (uint32_t) -1 will initialize this to the maximum
* value supported by server, which is recommended."
*
* (uint32_t)-1 is NOT working here, setting it to tlength*2, like
* openal-soft-pulseaudio does.
*/
attr.maxlength = attr.tlength*2;
/* minreq: Minimum request
*
* "The server does not request less than minreq bytes from the client,
* instead waits until the buffer is free enough to request more bytes
* at once. It is recommended to set this to (uint32_t) -1, which will
* initialize this to a value that is deemed sensible by the server."
*
* (uint32_t)-1 is NOT working here, setting it to 0, like
* openal-soft-pulseaudio does.
*/
attr.minreq = 0;
/* prebuf: Pre-buffering
*
* "The server does not start with playback before at least prebuf
* bytes are available in the buffer. It is recommended to set this
* to (uint32_t) -1, which will initialize this to the same value as
* tlength"
*/
attr.prebuf = (uint32_t)-1;
/* log the used target buffer length */
LOG->Trace("Pulse: using target buffer length of %i bytes", attr.tlength);
/* connect the stream for playback */
LOG->Trace("Pulse: pa_stream_connect_playback()...");
error = pa_stream_connect_playback(m_PulseStream, nullptr, &attr,
PA_STREAM_AUTO_TIMING_UPDATE, nullptr, nullptr);
if(error < 0)
{
if(asprintf(&m_Error, "pa_stream_connect_playback(): %s",
pa_strerror(pa_context_errno(m_PulseCtx))) == -1)
{
m_Error = nullptr;
}
m_Sem.Post();
return;
}
m_SampleRate = ss.rate;
}
int pa__init(pa_module* m)
{
pa_assert(m);
/* this example uses fixed sample spec and channel map
*
* however, usually we to the following instead:
* - set sample spec and chennel map to default values:
* m->core->default_sample_spec
* m->core->default_channel_map
*
* - overwrite them with pa_modargs_get_sample_spec_and_channel_map()
* if module was loaded with corresponding arguments
*
* - finally adjust values to nearest supported form
*/
pa_sample_spec sample_spec;
sample_spec.format = PA_SAMPLE_FLOAT32LE;
sample_spec.rate = 44100;
sample_spec.channels = 2;
pa_channel_map channel_map;
pa_channel_map_init_stereo(&channel_map);
/* get module arguments (key-value list passed to load-module) */
pa_modargs *args;
if (!(args = pa_modargs_new(m->argument, example_sink_modargs))) {
pa_log("[example sink] failed to parse module arguments");
goto error;
}
/* create and initialize module-specific data */
struct example_sink_userdata *u = pa_xnew0(struct example_sink_userdata, 1);
pa_assert(u);
m->userdata = u;
u->module = m;
u->rtpoll = pa_rtpoll_new();
pa_thread_mq_init(&u->thread_mq, m->core->mainloop, u->rtpoll);
u->output_file = pa_modargs_get_value(args, "output_file", "/dev/null");
u->output_fd = open(u->output_file, O_WRONLY | O_CREAT | O_TRUNC);
if (u->output_fd == -1) {
pa_log("[example sink] can't open output file %s", u->output_file);
goto error;
}
/* create and initialize sink */
pa_sink_new_data data;
pa_sink_new_data_init(&data);
data.driver = "example_sink";
data.module = m;
pa_sink_new_data_set_name(
&data,
pa_modargs_get_value(args, "sink_name", "example_sink"));
pa_sink_new_data_set_sample_spec(&data, &sample_spec);
pa_sink_new_data_set_channel_map(&data, &channel_map);
if (pa_modargs_get_proplist(
args,
"sink_properties",
data.proplist,
PA_UPDATE_REPLACE) < 0) {
pa_log("[example sink] invalid sink properties");
pa_sink_new_data_done(&data);
goto error;
}
u->sink = pa_sink_new(m->core, &data, PA_SINK_LATENCY);
pa_sink_new_data_done(&data);
if (!u->sink) {
pa_log("[example sink] failed to create sink");
goto error;
}
/* setup sink callbacks */
u->sink->parent.process_msg = process_message;
u->sink->userdata = u;
/* setup sink event loop */
pa_sink_set_asyncmsgq(u->sink, u->thread_mq.inq);
pa_sink_set_rtpoll(u->sink, u->rtpoll);
/* start thread for sink event loop and sample reader */
if (!(u->thread = pa_thread_new("example_sink", thread_loop, u))) {
pa_log("[example sink] failed to create thread");
goto error;
}
pa_sink_put(u->sink);
pa_modargs_free(args);
return 0;
error:
if (args) {
pa_modargs_free(args);
}
pa__done(m);
return -1;
}
int pa__init(pa_module *m)
{
pa_modargs *ma;
const char *master_sink_name;
const char *master_source_name;
const char *max_hw_frag_size_str;
const char *aep_runtime;
pa_source *master_source;
struct userdata *u;
pa_proplist *p;
pa_sink *master_sink;
const char *raw_sink_name;
const char *raw_source_name;
const char *voice_sink_name;
const char *voice_source_name;
const char *dbus_type;
int max_hw_frag_size = 3840;
pa_assert(m);
if (!(ma = pa_modargs_new(m->argument, valid_modargs)))
{
pa_log_error("Failed to parse module arguments");
goto fail;
}
voice_turn_sidetone_down();
master_sink_name = pa_modargs_get_value(ma, "master_sink", NULL);
master_source_name = pa_modargs_get_value(ma, "master_source", NULL);
raw_sink_name = pa_modargs_get_value(ma, "raw_sink_name", "sink.voice.raw");
raw_source_name = pa_modargs_get_value(ma, "raw_source_name",
"source.voice.raw");
voice_sink_name = pa_modargs_get_value(ma, "voice_sink_name", "sink.voice");
voice_source_name = pa_modargs_get_value(ma, "voice_source_name",
"source.voice");
dbus_type = pa_modargs_get_value(ma, "dbus_type", "session");
max_hw_frag_size_str = pa_modargs_get_value(ma, "max_hw_frag_size", "3840");
aep_runtime = pa_modargs_get_value(ma, "aep_runtime",
"bbaid1n-wr0-h9a22b--dbxpb--");
voice_set_aep_runtime_switch(aep_runtime);
pa_log_debug("Got arguments: master_sink=\"%s\" master_source=\"%s\" raw_sink_name=\"%s\" raw_source_name=\"%s\" dbus_type=\"%s\" max_hw_frag_size=\"%s\". ",
master_sink_name, master_source_name, raw_sink_name,
raw_source_name, dbus_type, max_hw_frag_size_str);
if (!(master_sink = pa_namereg_get(m->core, master_sink_name, PA_NAMEREG_SINK)))
{
pa_log("Master sink \"%s\" not found", master_sink_name);
goto fail;
}
if (!(master_source = pa_namereg_get(m->core, master_source_name, PA_NAMEREG_SOURCE)))
{
pa_log( "Master source \"%s\" not found", master_source_name);
goto fail;
}
if (master_sink->sample_spec.format != master_source->sample_spec.format &&
master_sink->sample_spec.rate != master_source->sample_spec.rate &&
master_sink->sample_spec.channels != master_source->sample_spec.channels)
{
pa_log("Master source and sink must have same sample spec");
goto fail;
}
if (pa_atoi(max_hw_frag_size_str, &max_hw_frag_size) < 0 ||
max_hw_frag_size < 960 ||
max_hw_frag_size > 128*960)
{
pa_log("Bad value for max_hw_frag_size: %s", max_hw_frag_size_str);
goto fail;
}
m->userdata = u = pa_xnew0(struct userdata, 1);
u->core = m->core;
u->module = m;
u->modargs = ma;
u->master_sink = master_sink;
u->master_source = master_source;
u->mainloop_handler = voice_mainloop_handler_new(u);;
u->ul_timing_advance = 500; // = 500 micro seconds, seems to be a good default value
pa_channel_map_init_mono(&u->mono_map);
pa_channel_map_init_stereo(&u->stereo_map);
u->hw_sample_spec.format = PA_SAMPLE_S16NE;
u->hw_sample_spec.rate = SAMPLE_RATE_HW_HZ;
u->hw_sample_spec.channels = 2;
u->hw_mono_sample_spec.format = PA_SAMPLE_S16NE;
u->hw_mono_sample_spec.rate = SAMPLE_RATE_HW_HZ;
u->hw_mono_sample_spec.channels = 1;
u->aep_sample_spec.format = PA_SAMPLE_S16NE;
u->aep_sample_spec.rate = SAMPLE_RATE_AEP_HZ;
u->aep_sample_spec.channels = 1;
pa_channel_map_init_mono(&u->aep_channel_map);
// The result is rounded down incorrectly thus +1
u->aep_fragment_size = pa_usec_to_bytes(PERIOD_AEP_USECS+1, &u->aep_sample_spec);
u->aep_hw_fragment_size = pa_usec_to_bytes(PERIOD_AEP_USECS+1, &u->hw_sample_spec);
u->hw_fragment_size = pa_usec_to_bytes(PERIOD_MASTER_USECS+1, &u->hw_sample_spec);
u->hw_fragment_size_max = max_hw_frag_size;
if (0 != (u->hw_fragment_size_max % u->hw_fragment_size))
u->hw_fragment_size_max += u->hw_fragment_size - (u->hw_fragment_size_max % u->hw_fragment_size);
u->aep_hw_mono_fragment_size = pa_usec_to_bytes(PERIOD_AEP_USECS+1, &u->hw_mono_sample_spec);
u->hw_mono_fragment_size = pa_usec_to_bytes(PERIOD_MASTER_USECS+1, &u->hw_mono_sample_spec);
u->voice_ul_fragment_size = pa_usec_to_bytes(PERIOD_CMT_USECS+1, &u->aep_sample_spec);
pa_silence_memchunk_get(&u->core->silence_cache,
u->core->mempool,
&u->aep_silence_memchunk,
&u->aep_sample_spec,
u->aep_fragment_size);
voice_memchunk_pool_load(u);
if (voice_init_raw_sink(u, raw_sink_name))
goto fail;
pa_sink_put(u->raw_sink);
if (voice_init_voip_sink(u, voice_sink_name))
goto fail;
pa_sink_put(u->voip_sink);
if (voice_init_aep_sink_input(u))
goto fail;
pa_atomic_store(&u->mixer_state, PROP_MIXER_TUNING_PRI);
u->alt_mixer_compensation = PA_VOLUME_NORM;
if (voice_init_hw_sink_input(u))
goto fail;
u->sink_temp_buff = pa_xmalloc(2 * u->hw_fragment_size_max);
u->sink_temp_buff_len = 2 * u->hw_fragment_size_max;
u->dl_memblockq =
pa_memblockq_new(0, 2 * u->voice_ul_fragment_size, 0,
pa_frame_size(&u->aep_sample_spec), 0, 0, 0, NULL);
if (voice_init_raw_source(u, raw_source_name))
goto fail;
pa_source_put(u->raw_source);
if (voice_init_voip_source(u, voice_source_name))
goto fail;
pa_source_put(u->voip_source);
if (voice_init_hw_source_output(u))
goto fail;
u->hw_source_memblockq =
pa_memblockq_new(0, 2 * u->hw_fragment_size_max, 0,
pa_frame_size(&u->hw_sample_spec), 0, 0, 0, NULL);
u->ul_memblockq =
pa_memblockq_new(0, 2 * u->voice_ul_fragment_size, 0,
pa_frame_size(&u->aep_sample_spec), 0, 0, 0, NULL);
u->cs_call_sink_input = 0;
u->dl_sideinfo_queue = pa_queue_new();
u->linear_q15_master_volume_L = INT16_MAX;
u->linear_q15_master_volume_R = INT16_MAX;
u->field_2CC = 0;
voice_aep_ear_ref_init(u);
if (voice_convert_init(u))
goto fail;
if (voice_init_event_forwarder(u, dbus_type) || voice_init_cmtspeech(u))
goto fail;
if (!(u->wb_mic_iir_eq = iir_eq_new(u->hw_fragment_size / 2,
master_source->sample_spec.channels)))
goto fail;
if (!(u->nb_mic_iir_eq = iir_eq_new( u->aep_fragment_size / 2, 1)))
goto fail;
if (!(u->wb_ear_iir_eq = fir_eq_new(master_sink->sample_spec.rate,
master_sink->sample_spec.channels)))
goto fail;
if (!(u->nb_ear_iir_eq = iir_eq_new(u->aep_fragment_size / 2, 1)))
goto fail;
u->input_task_active = FALSE;
u->xprot_watchdog = TRUE;
u->ambient_temp = 30;
if (!(u->xprot = xprot_new()))
goto fail;
u->aep_enable = FALSE;
u->wb_meq_enable = FALSE;
u->wb_eeq_enable = FALSE;
u->nb_meq_enable = FALSE;
u->nb_eeq_enable = FALSE;
u->xprot_enable = FALSE;
u->updating_parameters = FALSE;
u->sink_proplist_changed_slot =
pa_hook_connect(&m->core->hooks[PA_CORE_HOOK_SINK_PROPLIST_CHANGED],
0, (pa_hook_cb_t)sink_proplist_changed_cb, u);;
u->source_proplist_changed_slot =
pa_hook_connect( &m->core->hooks[PA_CORE_HOOK_SOURCE_PROPLIST_CHANGED], 0,
(pa_hook_cb_t)source_proplist_changed_cb, u);
u->mode_accessory_hwid_hash = 0;
p = pa_proplist_new();
pa_proplist_sets(p, PA_NOKIA_PROP_AUDIO_MODE, "ihf");
pa_proplist_sets(p, PA_NOKIA_PROP_AUDIO_ACCESSORY_HWID, "");
pa_sink_update_proplist( master_sink, PA_UPDATE_REPLACE, p);
pa_proplist_free(p);
pa_source_output_put(u->hw_source_output);
pa_sink_input_put(u->hw_sink_input);
pa_sink_input_put(u->aep_sink_input);
u->sink_subscription = pa_subscription_new(m->core, PA_SUBSCRIPTION_MASK_SINK,
sink_subscribe_cb, u);
return 0;
fail:
if (ma)
pa_modargs_free(ma);
pa__done(m);
return -1;
}
static gboolean
gst_pulsesrc_create_stream (GstPulseSrc * pulsesrc, GstCaps * caps)
{
pa_channel_map channel_map;
GstStructure *s;
gboolean need_channel_layout = FALSE;
GstRingBufferSpec spec;
const gchar *name;
memset (&spec, 0, sizeof (GstRingBufferSpec));
spec.latency_time = GST_SECOND;
if (!gst_ring_buffer_parse_caps (&spec, caps)) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
("Can't parse caps."), (NULL));
goto fail;
}
/* Keep the refcount of the caps at 1 to make them writable */
gst_caps_unref (spec.caps);
if (!gst_pulse_fill_sample_spec (&spec, &pulsesrc->sample_spec)) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
("Invalid sample specification."), (NULL));
goto fail;
}
pa_threaded_mainloop_lock (pulsesrc->mainloop);
if (!pulsesrc->context) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Bad context"), (NULL));
goto unlock_and_fail;
}
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_has_field (s, "channel-layout") ||
!gst_pulse_gst_to_channel_map (&channel_map, &spec)) {
if (spec.channels == 1)
pa_channel_map_init_mono (&channel_map);
else if (spec.channels == 2)
pa_channel_map_init_stereo (&channel_map);
else
need_channel_layout = TRUE;
}
name = "Record Stream";
if (pulsesrc->proplist) {
if (!(pulsesrc->stream = pa_stream_new_with_proplist (pulsesrc->context,
name, &pulsesrc->sample_spec,
(need_channel_layout) ? NULL : &channel_map,
pulsesrc->proplist))) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to create stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
} else if (!(pulsesrc->stream = pa_stream_new (pulsesrc->context,
name, &pulsesrc->sample_spec,
(need_channel_layout) ? NULL : &channel_map))) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to create stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
if (need_channel_layout) {
const pa_channel_map *m = pa_stream_get_channel_map (pulsesrc->stream);
gst_pulse_channel_map_to_gst (m, &spec);
caps = spec.caps;
}
GST_DEBUG_OBJECT (pulsesrc, "Caps are %" GST_PTR_FORMAT, caps);
pa_stream_set_state_callback (pulsesrc->stream, gst_pulsesrc_stream_state_cb,
pulsesrc);
pa_stream_set_read_callback (pulsesrc->stream, gst_pulsesrc_stream_request_cb,
pulsesrc);
pa_stream_set_underflow_callback (pulsesrc->stream,
gst_pulsesrc_stream_underflow_cb, pulsesrc);
pa_stream_set_overflow_callback (pulsesrc->stream,
gst_pulsesrc_stream_overflow_cb, pulsesrc);
pa_stream_set_latency_update_callback (pulsesrc->stream,
gst_pulsesrc_stream_latency_update_cb, pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return TRUE;
unlock_and_fail:
gst_pulsesrc_destroy_stream (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
fail:
return FALSE;
}
/**
* Setup a new stream based on the properties of the given audio_buf
*/
static void
stream_setup(pa_audio_mode_t *pam, audio_buf_t *ab)
{
pa_stream *s;
char buf[100];
int flags = 0;
#if PA_API_VERSION >= 12
pa_proplist *pl;
media_pipe_t *mp = ab->ab_mp;
#endif
pa_channel_map map;
pa_cvolume cv;
memset(&pam->ss, 0, sizeof(pa_sample_spec));
pam->ss.format = ab->ab_isfloat ? PA_SAMPLE_FLOAT32NE : PA_SAMPLE_S16NE;
pam->ss.rate = ab->ab_samplerate;
switch(ab->ab_format) {
case AM_FORMAT_PCM_STEREO:
pam->ss.channels = 2;
pa_channel_map_init_stereo(&map);
break;
case AM_FORMAT_PCM_5DOT0:
pam->ss.channels = 5;
pa_channel_map_init(&map);
map.channels = 5;
map.map[0] = PA_CHANNEL_POSITION_LEFT;
map.map[1] = PA_CHANNEL_POSITION_RIGHT;
map.map[2] = PA_CHANNEL_POSITION_CENTER;
map.map[3] = PA_CHANNEL_POSITION_SIDE_LEFT;
map.map[4] = PA_CHANNEL_POSITION_SIDE_RIGHT;
break;
case AM_FORMAT_PCM_5DOT1:
pam->ss.channels = 6;
pa_channel_map_init(&map);
map.channels = 6;
map.map[0] = PA_CHANNEL_POSITION_LEFT;
map.map[1] = PA_CHANNEL_POSITION_RIGHT;
map.map[2] = PA_CHANNEL_POSITION_CENTER;
map.map[3] = PA_CHANNEL_POSITION_LFE;
map.map[4] = PA_CHANNEL_POSITION_SIDE_LEFT;
map.map[5] = PA_CHANNEL_POSITION_SIDE_RIGHT;
break;
case AM_FORMAT_PCM_7DOT1:
pam->ss.channels = 8;
pa_channel_map_init(&map);
map.channels = 8;
map.map[0] = PA_CHANNEL_POSITION_LEFT;
map.map[1] = PA_CHANNEL_POSITION_RIGHT;
map.map[2] = PA_CHANNEL_POSITION_CENTER;
map.map[3] = PA_CHANNEL_POSITION_LFE;
map.map[4] = PA_CHANNEL_POSITION_SIDE_LEFT;
map.map[5] = PA_CHANNEL_POSITION_SIDE_RIGHT;
map.map[6] = PA_CHANNEL_POSITION_REAR_LEFT;
map.map[7] = PA_CHANNEL_POSITION_REAR_RIGHT;
break;
case AM_FORMAT_PCM_6DOT1:
pam->ss.channels = 7;
pa_channel_map_init(&map);
map.channels = 7;
map.map[0] = PA_CHANNEL_POSITION_LEFT;
map.map[1] = PA_CHANNEL_POSITION_RIGHT;
map.map[2] = PA_CHANNEL_POSITION_CENTER;
map.map[3] = PA_CHANNEL_POSITION_LFE;
map.map[4] = PA_CHANNEL_POSITION_SIDE_LEFT;
map.map[5] = PA_CHANNEL_POSITION_SIDE_RIGHT;
map.map[6] = PA_CHANNEL_POSITION_REAR_CENTER;
break;
default:
abort();
}
TRACE(TRACE_DEBUG, "PA", "Created stream %s",
pa_sample_spec_snprint(buf, sizeof(buf), &pam->ss));
#if PA_API_VERSION >= 12
pl = pa_proplist_new();
if(mp->mp_flags & MP_VIDEO)
pa_proplist_sets(pl, PA_PROP_MEDIA_ROLE, "video");
else
pa_proplist_sets(pl, PA_PROP_MEDIA_ROLE, "music");
s = pa_stream_new_with_proplist(pam->context, "Showtime playback",
&pam->ss, &map, pl);
pa_proplist_free(pl);
#else
s = pa_stream_new(pam->context, "Showtime playback", &pam->ss, &map);
#endif
pa_stream_set_state_callback(s, stream_state_callback, pam);
pa_stream_set_write_callback(s, stream_write_callback, pam);
flags |= PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_INTERPOLATE_TIMING;
memset(&cv, 0, sizeof(cv));
pa_cvolume_set(&cv, pam->ss.channels, pam->mastervol);
#if 1
pa_buffer_attr pba = {0};
pba.fragsize = (uint32_t)-1;
pba.maxlength = 16 * 1024;
pba.minreq = 3 * 1024;
pba.prebuf = 8 * 1024;
pba.tlength = 12 * 1024;
#endif
pa_stream_connect_playback(s, NULL, &pba, flags, &cv, NULL);
pam->stream = s;
pam->cur_rate = ab->ab_samplerate;
pam->cur_format = ab->ab_format;
pam->cur_isfloat = ab->ab_isfloat;
}
static int
pulseaudio_audio_reconfig(audio_decoder_t *ad)
{
decoder_t *d = (decoder_t *)ad;
int i;
pa_threaded_mainloop_lock(mainloop);
if(pulseaudio_make_context_ready()) {
pa_threaded_mainloop_unlock(mainloop);
return -1;
}
if(d->s) {
pa_stream_disconnect(d->s);
pa_stream_unref(d->s);
}
pa_channel_map map;
ad->ad_out_sample_rate = ad->ad_in_sample_rate;
d->ss.rate = ad->ad_in_sample_rate;
switch(ad->ad_in_sample_format) {
case AV_SAMPLE_FMT_S32:
case AV_SAMPLE_FMT_S32P:
ad->ad_out_sample_format = AV_SAMPLE_FMT_S32;
d->ss.format = PA_SAMPLE_S32NE;
d->framesize = sizeof(int32_t);
break;
case AV_SAMPLE_FMT_S16:
case AV_SAMPLE_FMT_S16P:
ad->ad_out_sample_format = AV_SAMPLE_FMT_S16;
d->ss.format = PA_SAMPLE_S16NE;
d->framesize = sizeof(int16_t);
break;
default:
ad->ad_out_sample_format = AV_SAMPLE_FMT_FLT;
d->ss.format = PA_SAMPLE_FLOAT32NE;
d->framesize = sizeof(float);
break;
}
switch(ad->ad_in_channel_layout) {
case AV_CH_LAYOUT_MONO:
d->ss.channels = 1;
ad->ad_out_channel_layout = AV_CH_LAYOUT_MONO;
pa_channel_map_init_mono(&map);
break;
case AV_CH_LAYOUT_STEREO:
d->ss.channels = 2;
ad->ad_out_channel_layout = AV_CH_LAYOUT_STEREO;
pa_channel_map_init_stereo(&map);
default:
pa_channel_map_init(&map);
for(i = 0; i < sizeof(av2pa_map) / sizeof(av2pa_map[0]); i++) {
if(ad->ad_in_channel_layout & av2pa_map[i].avmask) {
ad->ad_out_channel_layout |= av2pa_map[i].avmask;
map.map[map.channels++] = av2pa_map[i].papos;
}
}
d->ss.channels = map.channels;
break;
}
d->framesize *= d->ss.channels;
ad->ad_tile_size = pa_context_get_tile_size(ctx, &d->ss) / d->framesize;
char buf[100];
char buf2[PA_CHANNEL_MAP_SNPRINT_MAX];
TRACE(TRACE_DEBUG, "PA", "Created stream %s [%s] (tilesize=%d)",
pa_sample_spec_snprint(buf, sizeof(buf), &d->ss),
pa_channel_map_snprint(buf2, sizeof(buf2), &map),
ad->ad_tile_size);
#if PA_API_VERSION >= 12
pa_proplist *pl = pa_proplist_new();
media_pipe_t *mp = ad->ad_mp;
if(mp->mp_flags & MP_VIDEO)
pa_proplist_sets(pl, PA_PROP_MEDIA_ROLE, "video");
else
pa_proplist_sets(pl, PA_PROP_MEDIA_ROLE, "music");
d->s = pa_stream_new_with_proplist(ctx, "Showtime playback",
&d->ss, &map, pl);
pa_proplist_free(pl);
#else
d->s = pa_stream_new(ctx, "Showtime playback", &ss, &map);
#endif
int flags = 0;
pa_stream_set_state_callback(d->s, stream_state_callback, d);
pa_stream_set_write_callback(d->s, stream_write_callback, d);
flags |= PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_INTERPOLATE_TIMING;
pa_stream_connect_playback(d->s, NULL, NULL, flags, NULL, NULL);
while(1) {
switch(pa_stream_get_state(d->s)) {
case PA_STREAM_UNCONNECTED:
case PA_STREAM_CREATING:
pa_threaded_mainloop_wait(mainloop);
continue;
case PA_STREAM_READY:
pa_threaded_mainloop_unlock(mainloop);
return 0;
case PA_STREAM_TERMINATED:
case PA_STREAM_FAILED:
pa_threaded_mainloop_unlock(mainloop);
return 1;
}
}
}
static gboolean
gst_pulsesrc_create_stream (GstPulseSrc * pulsesrc, GstCaps ** caps)
{
pa_channel_map channel_map;
const pa_channel_map *m;
GstStructure *s;
gboolean need_channel_layout = FALSE;
GstAudioRingBufferSpec spec;
const gchar *name;
s = gst_caps_get_structure (*caps, 0);
gst_structure_get_int (s, "channels", &spec.info.channels);
if (!gst_structure_has_field (s, "channel-mask")) {
if (spec.info.channels == 1) {
pa_channel_map_init_mono (&channel_map);
} else if (spec.info.channels == 2) {
gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK,
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) |
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT), NULL);
pa_channel_map_init_stereo (&channel_map);
} else {
need_channel_layout = TRUE;
gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK,
G_GUINT64_CONSTANT (0), NULL);
}
}
memset (&spec, 0, sizeof (GstAudioRingBufferSpec));
spec.latency_time = GST_SECOND;
if (!gst_audio_ring_buffer_parse_caps (&spec, *caps))
goto invalid_caps;
/* Keep the refcount of the caps at 1 to make them writable */
gst_caps_unref (spec.caps);
if (!need_channel_layout
&& !gst_pulse_gst_to_channel_map (&channel_map, &spec)) {
need_channel_layout = TRUE;
gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK,
G_GUINT64_CONSTANT (0), NULL);
memset (spec.info.position, 0xff, sizeof (spec.info.position));
}
if (!gst_pulse_fill_sample_spec (&spec, &pulsesrc->sample_spec))
goto invalid_spec;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
if (!pulsesrc->context)
goto bad_context;
name = "Record Stream";
if (pulsesrc->proplist) {
if (!(pulsesrc->stream = pa_stream_new_with_proplist (pulsesrc->context,
name, &pulsesrc->sample_spec,
(need_channel_layout) ? NULL : &channel_map,
pulsesrc->proplist)))
goto create_failed;
} else if (!(pulsesrc->stream = pa_stream_new (pulsesrc->context,
name, &pulsesrc->sample_spec,
(need_channel_layout) ? NULL : &channel_map)))
goto create_failed;
m = pa_stream_get_channel_map (pulsesrc->stream);
gst_pulse_channel_map_to_gst (m, &spec);
gst_audio_channel_positions_to_valid_order (spec.info.position,
spec.info.channels);
gst_caps_unref (*caps);
*caps = gst_audio_info_to_caps (&spec.info);
GST_DEBUG_OBJECT (pulsesrc, "Caps are %" GST_PTR_FORMAT, *caps);
pa_stream_set_state_callback (pulsesrc->stream, gst_pulsesrc_stream_state_cb,
pulsesrc);
pa_stream_set_read_callback (pulsesrc->stream, gst_pulsesrc_stream_request_cb,
pulsesrc);
pa_stream_set_underflow_callback (pulsesrc->stream,
gst_pulsesrc_stream_underflow_cb, pulsesrc);
pa_stream_set_overflow_callback (pulsesrc->stream,
gst_pulsesrc_stream_overflow_cb, pulsesrc);
pa_stream_set_latency_update_callback (pulsesrc->stream,
gst_pulsesrc_stream_latency_update_cb, pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return TRUE;
/* ERRORS */
invalid_caps:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
("Can't parse caps."), (NULL));
goto fail;
}
invalid_spec:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
("Invalid sample specification."), (NULL));
goto fail;
}
bad_context:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Bad context"), (NULL));
goto unlock_and_fail;
}
create_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to create 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);
fail:
return FALSE;
}
}
int pa__init(pa_module*m) {
pa_modargs *ma = NULL;
struct userdata *u;
const char *master_sink_name;
const char *master_source_name;
const char *raw_sink_name;
const char *raw_source_name;
const char *voice_sink_name;
const char *voice_source_name;
const char *max_hw_frag_size_str;
int max_hw_frag_size = 3840;
pa_sink *master_sink;
pa_source *master_source;
pa_assert(m);
if (!(ma = pa_modargs_new(m->argument, valid_modargs))) {
pa_log("Failed to parse module arguments");
goto fail;
}
master_sink_name = pa_modargs_get_value(ma, "master_sink", NULL);
master_source_name = pa_modargs_get_value(ma, "master_source", NULL);
raw_sink_name = pa_modargs_get_value(ma, "raw_sink_name", "sink.voice.raw");
raw_source_name = pa_modargs_get_value(ma, "raw_source_name", "source.voice.raw");
voice_sink_name = pa_modargs_get_value(ma, "voice_sink_name", "sink.voice");
voice_source_name = pa_modargs_get_value(ma, "voice_source_name", "source.voice");
max_hw_frag_size_str = pa_modargs_get_value(ma, "max_hw_frag_size", "3840");
pa_log_debug("Got arguments: master_sink=\"%s\" master_source=\"%s\" "
"raw_sink_name=\"%s\" raw_source_name=\"%s\" max_hw_frag_size=\"%s\".",
master_sink_name, master_source_name,
raw_sink_name, raw_source_name,
max_hw_frag_size_str);
if (!(master_sink = pa_namereg_get(m->core, master_sink_name, PA_NAMEREG_SINK))) {
pa_log("Master sink \"%s\" not found", master_sink_name);
goto fail;
}
if (!(master_source = pa_namereg_get(m->core, master_source_name, PA_NAMEREG_SOURCE))) {
pa_log("Master source \"%s\" not found", master_source_name);
goto fail;
}
if (master_sink->sample_spec.format != master_source->sample_spec.format &&
master_sink->sample_spec.rate != master_source->sample_spec.rate &&
master_sink->sample_spec.channels != master_source->sample_spec.channels) {
pa_log("Master source and sink must have same sample spec");
goto fail;
}
if (pa_atoi(max_hw_frag_size_str, &max_hw_frag_size) < 0 ||
max_hw_frag_size < 960 ||
max_hw_frag_size > 128*960) {
pa_log("Bad value for max_hw_frag_size: %s", max_hw_frag_size_str);
goto fail;
}
m->userdata = u = pa_xnew0(struct userdata, 1);
u->modargs = ma;
u->core = m->core;
u->module = m;
u->master_sink = master_sink;
u->master_source = master_source;
set_hooks(u);
u->mainloop_handler = voice_mainloop_handler_new(u);
u->ul_timing_advance = 500; // = 500 micro seconds, seems to be a good default value
pa_channel_map_init_mono(&u->mono_map);
pa_channel_map_init_stereo(&u->stereo_map);
u->hw_sample_spec.format = PA_SAMPLE_S16NE;
u->hw_sample_spec.rate = VOICE_SAMPLE_RATE_HW_HZ;
u->hw_sample_spec.channels = 2;
u->hw_mono_sample_spec.format = PA_SAMPLE_S16NE;
u->hw_mono_sample_spec.rate = VOICE_SAMPLE_RATE_HW_HZ;
u->hw_mono_sample_spec.channels = 1;
u->aep_sample_spec.format = PA_SAMPLE_S16NE;
u->aep_sample_spec.rate = VOICE_SAMPLE_RATE_AEP_HZ;
u->aep_sample_spec.channels = 1;
pa_channel_map_init_mono(&u->aep_channel_map);
// The result is rounded down incorrectly thus +1
u->aep_fragment_size = pa_usec_to_bytes(VOICE_PERIOD_AEP_USECS+1, &u->aep_sample_spec);
u->aep_hw_fragment_size = pa_usec_to_bytes(VOICE_PERIOD_AEP_USECS+1, &u->hw_sample_spec);
u->hw_fragment_size = pa_usec_to_bytes(VOICE_PERIOD_MASTER_USECS+1, &u->hw_sample_spec);
u->hw_fragment_size_max = max_hw_frag_size;
if (0 != (u->hw_fragment_size_max % u->hw_fragment_size))
u->hw_fragment_size_max += u->hw_fragment_size - (u->hw_fragment_size_max % u->hw_fragment_size);
u->aep_hw_mono_fragment_size = pa_usec_to_bytes(VOICE_PERIOD_AEP_USECS+1, &u->hw_mono_sample_spec);
u->hw_mono_fragment_size = pa_usec_to_bytes(VOICE_PERIOD_MASTER_USECS+1, &u->hw_mono_sample_spec);
u->voice_ul_fragment_size = pa_usec_to_bytes(VOICE_PERIOD_CMT_USECS+1, &u->aep_sample_spec);
pa_silence_memchunk_get(&u->core->silence_cache,
u->core->mempool,
&u->aep_silence_memchunk,
& u->aep_sample_spec,
u->aep_fragment_size);
voice_memchunk_pool_load(u);
if (voice_init_raw_sink(u, raw_sink_name))
goto fail;
u->call_state_tracker = pa_call_state_tracker_get(m->core);
pa_atomic_store(&u->mixer_state, PROP_MIXER_TUNING_PRI);
pa_call_state_tracker_set_active(u->call_state_tracker, FALSE);
u->alt_mixer_compensation = PA_VOLUME_NORM;
if (voice_init_hw_sink_input(u))
goto fail;
/* This must be set before calling pa_sink_put(), because pa_sink_put() has
* assertion
* "!(s->flags & PA_SINK_SHARE_VOLUME_WITH_MASTER) || s->flat_sink_input".
*/
u->raw_sink->flat_sink_input = u->hw_sink_input;
/* This must be called before calling voice_init_voip_sink(), because
* pa_sink_input_new() has assertion
* "PA_SINK_IS_LINKED(pa_sink_get_state(data->sink))". */
pa_sink_put(u->raw_sink);
/* This must be called before calling voice_init_aep_sink_input(), because
* the flat volume logic will otherwise mess up the aep sink input's volume
* when pa_sink_input_put(u->hw_sink_input) is called. */
pa_sink_input_put(u->hw_sink_input);
if (voice_init_voip_sink(u, voice_sink_name))
goto fail;
if (voice_init_aep_sink_input(u))
goto fail;
u->sink_temp_buff = pa_xmalloc(2*u->hw_fragment_size_max);
u->sink_temp_buff_len = 2*u->hw_fragment_size_max;
if (voice_init_raw_source(u, raw_source_name))
goto fail;
pa_source_put(u->raw_source);
if (voice_init_voip_source(u, voice_source_name))
goto fail;
pa_source_put(u->voip_source);
if (voice_init_hw_source_output(u))
goto fail;
/* TODO: Guess we should use max_hw_frag_size here */
u->hw_source_memblockq = // 8 * 5ms = 40ms
pa_memblockq_new(0, 2*u->hw_fragment_size_max, 0, pa_frame_size(&u->hw_sample_spec), 0, 0, 0, NULL);
u->ul_memblockq =
pa_memblockq_new(0, 2*u->voice_ul_fragment_size, 0, pa_frame_size(&u->aep_sample_spec), 0, 0, 0, NULL);
u->dl_sideinfo_queue = pa_queue_new();
u->ul_deadline = 0;
u->linear_q15_master_volume_L = INT16_MAX;
u->linear_q15_master_volume_R = INT16_MAX;
voice_aep_ear_ref_init(u);
if (voice_convert_init(u))
goto fail;
/* IHF mode is the default and this initialization is consistent with it. */
u->active_mic_channel = MIC_CH0;
meego_parameter_request_updates("voice", (pa_hook_cb_t)voice_parameter_cb, PA_HOOK_NORMAL, FALSE, u);
meego_parameter_request_updates("alsa", (pa_hook_cb_t)alsa_parameter_cb, PA_HOOK_NORMAL, FALSE, u);
meego_parameter_request_updates("aep", (pa_hook_cb_t)aep_parameter_cb, PA_HOOK_LATE, FALSE, u);
/* aep-s-i */
/* voip-sink ---\ hw-sink-input */
/* > optimized mix -------------> master-sink */
/* | */
/* raw-sink */
/* */
/* voip-src <--- hw-source-output */
/* < mux <------------- master-src */
/* raw-src <--- */
u->voip_sink->flat_sink_input = u->aep_sink_input;
pa_sink_put(u->voip_sink);
pa_source_output_put(u->hw_source_output);
pa_sink_input_put(u->aep_sink_input);
u->sink_subscription = pa_subscription_new(m->core, PA_SUBSCRIPTION_MASK_SINK | PA_SUBSCRIPTION_MASK_SINK_INPUT, master_sink_volume_subscribe_cb, u);
u->previous_master_source_state = pa_source_get_state(u->master_source);
u->source_change_subscription = pa_subscription_new(m->core, PA_SUBSCRIPTION_MASK_SOURCE, master_source_state_subscribe_cb, u);
return 0;
fail:
pa__done(m);
return -1;
}
int pa__init(pa_module*m) {
struct userdata *u;
pa_sample_spec ss, sink_input_ss;
pa_channel_map map, sink_input_map;
pa_modargs *ma;
pa_sink *master=NULL;
pa_sink_input_new_data sink_input_data;
pa_sink_new_data sink_data;
bool use_volume_sharing = true;
bool force_flat_volume = false;
pa_memchunk silence;
const char *hrir_file;
unsigned i, j, found_channel_left, found_channel_right;
float *hrir_data;
pa_sample_spec hrir_ss;
pa_channel_map hrir_map;
pa_sample_spec hrir_temp_ss;
pa_memchunk hrir_temp_chunk, hrir_temp_chunk_resampled;
pa_resampler *resampler;
size_t hrir_copied_length, hrir_total_length;
hrir_temp_chunk.memblock = NULL;
hrir_temp_chunk_resampled.memblock = NULL;
pa_assert(m);
if (!(ma = pa_modargs_new(m->argument, valid_modargs))) {
pa_log("Failed to parse module arguments.");
goto fail;
}
if (!(master = pa_namereg_get(m->core, pa_modargs_get_value(ma, "master", NULL), PA_NAMEREG_SINK))) {
pa_log("Master sink not found");
goto fail;
}
pa_assert(master);
u = pa_xnew0(struct userdata, 1);
u->module = m;
m->userdata = u;
/* Initialize hrir and input buffer */
/* this is the hrir file for the left ear! */
if (!(hrir_file = pa_modargs_get_value(ma, "hrir", NULL))) {
pa_log("The mandatory 'hrir' module argument is missing.");
goto fail;
}
if (pa_sound_file_load(master->core->mempool, hrir_file, &hrir_temp_ss, &hrir_map, &hrir_temp_chunk, NULL) < 0) {
pa_log("Cannot load hrir file.");
goto fail;
}
/* sample spec / map of hrir */
hrir_ss.format = PA_SAMPLE_FLOAT32;
hrir_ss.rate = master->sample_spec.rate;
hrir_ss.channels = hrir_temp_ss.channels;
/* sample spec of sink */
ss = hrir_ss;
map = hrir_map;
if (pa_modargs_get_sample_spec_and_channel_map(ma, &ss, &map, PA_CHANNEL_MAP_DEFAULT) < 0) {
pa_log("Invalid sample format specification or channel map");
goto fail;
}
ss.format = PA_SAMPLE_FLOAT32;
hrir_ss.rate = ss.rate;
u->channels = ss.channels;
if (pa_modargs_get_value_boolean(ma, "use_volume_sharing", &use_volume_sharing) < 0) {
pa_log("use_volume_sharing= expects a boolean argument");
goto fail;
}
if (pa_modargs_get_value_boolean(ma, "force_flat_volume", &force_flat_volume) < 0) {
pa_log("force_flat_volume= expects a boolean argument");
goto fail;
}
if (use_volume_sharing && force_flat_volume) {
pa_log("Flat volume can't be forced when using volume sharing.");
goto fail;
}
/* sample spec / map of sink input */
pa_channel_map_init_stereo(&sink_input_map);
sink_input_ss.channels = 2;
sink_input_ss.format = PA_SAMPLE_FLOAT32;
sink_input_ss.rate = ss.rate;
u->sink_fs = pa_frame_size(&ss);
u->fs = pa_frame_size(&sink_input_ss);
/* Create sink */
pa_sink_new_data_init(&sink_data);
sink_data.driver = __FILE__;
sink_data.module = m;
if (!(sink_data.name = pa_xstrdup(pa_modargs_get_value(ma, "sink_name", NULL))))
sink_data.name = pa_sprintf_malloc("%s.vsurroundsink", master->name);
pa_sink_new_data_set_sample_spec(&sink_data, &ss);
pa_sink_new_data_set_channel_map(&sink_data, &map);
pa_proplist_sets(sink_data.proplist, PA_PROP_DEVICE_MASTER_DEVICE, master->name);
pa_proplist_sets(sink_data.proplist, PA_PROP_DEVICE_CLASS, "filter");
pa_proplist_sets(sink_data.proplist, "device.vsurroundsink.name", sink_data.name);
if (pa_modargs_get_proplist(ma, "sink_properties", sink_data.proplist, PA_UPDATE_REPLACE) < 0) {
pa_log("Invalid properties");
pa_sink_new_data_done(&sink_data);
goto fail;
}
if ((u->auto_desc = !pa_proplist_contains(sink_data.proplist, PA_PROP_DEVICE_DESCRIPTION))) {
const char *z;
z = pa_proplist_gets(master->proplist, PA_PROP_DEVICE_DESCRIPTION);
pa_proplist_setf(sink_data.proplist, PA_PROP_DEVICE_DESCRIPTION, "Virtual Surround Sink %s on %s", sink_data.name, z ? z : master->name);
}
u->sink = pa_sink_new(m->core, &sink_data, (master->flags & (PA_SINK_LATENCY|PA_SINK_DYNAMIC_LATENCY))
| (use_volume_sharing ? PA_SINK_SHARE_VOLUME_WITH_MASTER : 0));
pa_sink_new_data_done(&sink_data);
if (!u->sink) {
pa_log("Failed to create sink.");
goto fail;
}
u->sink->parent.process_msg = sink_process_msg_cb;
u->sink->set_state = sink_set_state_cb;
u->sink->update_requested_latency = sink_update_requested_latency_cb;
u->sink->request_rewind = sink_request_rewind_cb;
pa_sink_set_set_mute_callback(u->sink, sink_set_mute_cb);
if (!use_volume_sharing) {
pa_sink_set_set_volume_callback(u->sink, sink_set_volume_cb);
pa_sink_enable_decibel_volume(u->sink, true);
}
/* Normally this flag would be enabled automatically be we can force it. */
if (force_flat_volume)
u->sink->flags |= PA_SINK_FLAT_VOLUME;
u->sink->userdata = u;
pa_sink_set_asyncmsgq(u->sink, master->asyncmsgq);
/* Create sink input */
pa_sink_input_new_data_init(&sink_input_data);
sink_input_data.driver = __FILE__;
sink_input_data.module = m;
pa_sink_input_new_data_set_sink(&sink_input_data, master, false);
sink_input_data.origin_sink = u->sink;
pa_proplist_setf(sink_input_data.proplist, PA_PROP_MEDIA_NAME, "Virtual Surround Sink Stream from %s", pa_proplist_gets(u->sink->proplist, PA_PROP_DEVICE_DESCRIPTION));
pa_proplist_sets(sink_input_data.proplist, PA_PROP_MEDIA_ROLE, "filter");
pa_sink_input_new_data_set_sample_spec(&sink_input_data, &sink_input_ss);
pa_sink_input_new_data_set_channel_map(&sink_input_data, &sink_input_map);
pa_sink_input_new(&u->sink_input, m->core, &sink_input_data);
pa_sink_input_new_data_done(&sink_input_data);
if (!u->sink_input)
goto fail;
u->sink_input->pop = sink_input_pop_cb;
u->sink_input->process_rewind = sink_input_process_rewind_cb;
u->sink_input->update_max_rewind = sink_input_update_max_rewind_cb;
u->sink_input->update_max_request = sink_input_update_max_request_cb;
u->sink_input->update_sink_latency_range = sink_input_update_sink_latency_range_cb;
u->sink_input->update_sink_fixed_latency = sink_input_update_sink_fixed_latency_cb;
u->sink_input->kill = sink_input_kill_cb;
u->sink_input->attach = sink_input_attach_cb;
u->sink_input->detach = sink_input_detach_cb;
u->sink_input->state_change = sink_input_state_change_cb;
u->sink_input->moving = sink_input_moving_cb;
u->sink_input->volume_changed = use_volume_sharing ? NULL : sink_input_volume_changed_cb;
u->sink_input->mute_changed = sink_input_mute_changed_cb;
u->sink_input->userdata = u;
u->sink->input_to_master = u->sink_input;
pa_sink_input_get_silence(u->sink_input, &silence);
u->memblockq = pa_memblockq_new("module-virtual-surround-sink memblockq", 0, MEMBLOCKQ_MAXLENGTH, 0, &sink_input_ss, 1, 1, 0, &silence);
pa_memblock_unref(silence.memblock);
/* resample hrir */
resampler = pa_resampler_new(u->sink->core->mempool, &hrir_temp_ss, &hrir_map, &hrir_ss, &hrir_map,
PA_RESAMPLER_SRC_SINC_BEST_QUALITY, PA_RESAMPLER_NO_REMAP);
u->hrir_samples = hrir_temp_chunk.length / pa_frame_size(&hrir_temp_ss) * hrir_ss.rate / hrir_temp_ss.rate;
if (u->hrir_samples > 64) {
u->hrir_samples = 64;
pa_log("The (resampled) hrir contains more than 64 samples. Only the first 64 samples will be used to limit processor usage.");
}
hrir_total_length = u->hrir_samples * pa_frame_size(&hrir_ss);
u->hrir_channels = hrir_ss.channels;
u->hrir_data = (float *) pa_xmalloc(hrir_total_length);
hrir_copied_length = 0;
/* add silence to the hrir until we get enough samples out of the resampler */
while (hrir_copied_length < hrir_total_length) {
pa_resampler_run(resampler, &hrir_temp_chunk, &hrir_temp_chunk_resampled);
if (hrir_temp_chunk.memblock != hrir_temp_chunk_resampled.memblock) {
/* Silence input block */
pa_silence_memblock(hrir_temp_chunk.memblock, &hrir_temp_ss);
}
if (hrir_temp_chunk_resampled.memblock) {
/* Copy hrir data */
hrir_data = (float *) pa_memblock_acquire(hrir_temp_chunk_resampled.memblock);
if (hrir_total_length - hrir_copied_length >= hrir_temp_chunk_resampled.length) {
memcpy(u->hrir_data + hrir_copied_length, hrir_data, hrir_temp_chunk_resampled.length);
hrir_copied_length += hrir_temp_chunk_resampled.length;
} else {
memcpy(u->hrir_data + hrir_copied_length, hrir_data, hrir_total_length - hrir_copied_length);
hrir_copied_length = hrir_total_length;
}
pa_memblock_release(hrir_temp_chunk_resampled.memblock);
pa_memblock_unref(hrir_temp_chunk_resampled.memblock);
hrir_temp_chunk_resampled.memblock = NULL;
}
}
pa_resampler_free(resampler);
pa_memblock_unref(hrir_temp_chunk.memblock);
hrir_temp_chunk.memblock = NULL;
if (hrir_map.channels < map.channels) {
pa_log("hrir file does not have enough channels!");
goto fail;
}
normalize_hrir(u);
/* create mapping between hrir and input */
u->mapping_left = (unsigned *) pa_xnew0(unsigned, u->channels);
u->mapping_right = (unsigned *) pa_xnew0(unsigned, u->channels);
for (i = 0; i < map.channels; i++) {
found_channel_left = 0;
found_channel_right = 0;
for (j = 0; j < hrir_map.channels; j++) {
if (hrir_map.map[j] == map.map[i]) {
u->mapping_left[i] = j;
found_channel_left = 1;
}
if (hrir_map.map[j] == mirror_channel(map.map[i])) {
u->mapping_right[i] = j;
found_channel_right = 1;
}
}
if (!found_channel_left) {
pa_log("Cannot find mapping for channel %s", pa_channel_position_to_string(map.map[i]));
goto fail;
}
if (!found_channel_right) {
pa_log("Cannot find mapping for channel %s", pa_channel_position_to_string(mirror_channel(map.map[i])));
goto fail;
}
}
u->input_buffer = pa_xmalloc0(u->hrir_samples * u->sink_fs);
u->input_buffer_offset = 0;
pa_sink_put(u->sink);
pa_sink_input_put(u->sink_input);
pa_modargs_free(ma);
return 0;
fail:
if (hrir_temp_chunk.memblock)
pa_memblock_unref(hrir_temp_chunk.memblock);
if (hrir_temp_chunk_resampled.memblock)
pa_memblock_unref(hrir_temp_chunk_resampled.memblock);
if (ma)
pa_modargs_free(ma);
pa__done(m);
return -1;
}
Error AudioDriverPulseAudio::init_device() {
// If there is a specified device check that it is really present
if (device_name != "Default") {
Array list = get_device_list();
if (list.find(device_name) == -1) {
device_name = "Default";
new_device = "Default";
}
}
// Detect the amount of channels PulseAudio is using
// Note: If using an even amount of channels (2, 4, etc) channels and pa_map.channels will be equal,
// if not then pa_map.channels will have the real amount of channels PulseAudio is using and channels
// will have the amount of channels Godot is using (in this case it's pa_map.channels + 1)
detect_channels();
switch (pa_map.channels) {
case 1: // Mono
case 3: // Surround 2.1
case 5: // Surround 5.0
case 7: // Surround 7.0
channels = pa_map.channels + 1;
break;
case 2: // Stereo
case 4: // Surround 4.0
case 6: // Surround 5.1
case 8: // Surround 7.1
channels = pa_map.channels;
break;
default:
WARN_PRINTS("PulseAudio: Unsupported number of channels: " + itos(pa_map.channels));
pa_channel_map_init_stereo(&pa_map);
channels = 2;
break;
}
int latency = GLOBAL_DEF_RST("audio/output_latency", DEFAULT_OUTPUT_LATENCY);
buffer_frames = closest_power_of_2(latency * mix_rate / 1000);
pa_buffer_size = buffer_frames * pa_map.channels;
print_verbose("PulseAudio: detected " + itos(pa_map.channels) + " channels");
print_verbose("PulseAudio: audio buffer frames: " + itos(buffer_frames) + " calculated latency: " + itos(buffer_frames * 1000 / mix_rate) + "ms");
pa_sample_spec spec;
spec.format = PA_SAMPLE_S16LE;
spec.channels = pa_map.channels;
spec.rate = mix_rate;
pa_str = pa_stream_new(pa_ctx, "Sound", &spec, &pa_map);
if (pa_str == NULL) {
ERR_PRINTS("PulseAudio: pa_stream_new error: " + String(pa_strerror(pa_context_errno(pa_ctx))));
ERR_FAIL_V(ERR_CANT_OPEN);
}
pa_buffer_attr attr;
// set to appropriate buffer length (in bytes) from global settings
// Note: PulseAudio defaults to 4 fragments, which means that the actual
// latency is tlength / fragments. It seems that the PulseAudio has no way
// to get the fragments number so we're hardcoding this to the default of 4
const int fragments = 4;
attr.tlength = pa_buffer_size * sizeof(int16_t) * fragments;
// set them to be automatically chosen
attr.prebuf = (uint32_t)-1;
attr.maxlength = (uint32_t)-1;
attr.minreq = (uint32_t)-1;
const char *dev = device_name == "Default" ? NULL : device_name.utf8().get_data();
pa_stream_flags flags = pa_stream_flags(PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE);
int error_code = pa_stream_connect_playback(pa_str, dev, &attr, flags, NULL, NULL);
ERR_FAIL_COND_V(error_code < 0, ERR_CANT_OPEN);
samples_in.resize(buffer_frames * channels);
samples_out.resize(pa_buffer_size);
// Reset audio input to keep synchronisation.
input_position = 0;
input_size = 0;
return OK;
}
