static void pa_adrian_ec_fixate_spec(pa_sample_spec *source_ss, pa_channel_map *source_map,
pa_sample_spec *sink_ss, pa_channel_map *sink_map)
{
source_ss->format = PA_SAMPLE_S16NE;
source_ss->channels = 1;
pa_channel_map_init_mono(source_map);
*sink_ss = *source_ss;
*sink_map = *source_map;
}
static void pa_adrian_ec_fixate_spec(pa_sample_spec *rec_ss, pa_channel_map *rec_map,
pa_sample_spec *play_ss, pa_channel_map *play_map,
pa_sample_spec *out_ss, pa_channel_map *out_map)
{
out_ss->format = PA_SAMPLE_S16NE;
out_ss->channels = 1;
pa_channel_map_init_mono(out_map);
*play_ss = *out_ss;
*play_map = *out_map;
*rec_ss = *out_ss;
*rec_map = *out_map;
}
static inline void init_pulse(struct pa_fft *pa_fft)
{
/* PA spec */
fprintf(stderr, "device = %s\n", pa_fft->dev);
if (!pa_fft->dev) {
fprintf(stderr, "Warning: no device specified! It's highly possible "
"Pulseaudio will attempt to use the microphone!\n");
}
pa_fft->ss.format = PA_SAMPLE_FLOAT32LE;
pa_fft->ss.rate = 44100;
pa_fft->ss.channels = 1;
pa_channel_map_init_mono(&pa_fft->map);
if (!(pa_fft->s = pa_simple_new(NULL, "pa_fft", PA_STREAM_RECORD, pa_fft->dev,
"record", &pa_fft->ss, &pa_fft->map, NULL,
&pa_fft->error))) {
fprintf(stderr, __FILE__": pa_simple_new() failed: %s\n",
pa_strerror(pa_fft->error));
pa_fft->cont = 0;
return;
}
}
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;
}
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) {
pa_modargs *ma = NULL;
struct userdata *u;
const char *sink_name, *source_name, *dbus_type;
pa_sink *sink = NULL;
pa_source *source = NULL;
pa_assert(m);
if (!(ma = pa_modargs_new(m->argument, valid_modargs))) {
pa_log_error("Failed to parse module arguments");
goto fail;
}
sink_name = pa_modargs_get_value(ma, "sink", NULL);
source_name = pa_modargs_get_value(ma, "source", NULL);
dbus_type = pa_modargs_get_value(ma, "dbus_type", "session");
pa_log_debug("Got arguments: sink=\"%s\" source=\"%s\" dbus_type=\"%s\"",
sink_name, source_name, dbus_type);
u = pa_xnew0(struct userdata, 1);
m->userdata = u;
u->core = m->core;
u->module = m;
u->ss.format = PA_SAMPLE_S16NE;
u->ss.rate = CMTSPEECH_SAMPLERATE;
u->ss.channels = 1;
pa_channel_map_init_mono(&u->map);
/* The result is rounded down incorrectly thus +1 */
u->dl_frame_size = pa_usec_to_bytes(VOICE_SINK_FRAMESIZE+1, &u->ss);
u->ul_frame_size = pa_usec_to_bytes(VOICE_SOURCE_FRAMESIZE+1, &u->ss);
if (!(source = pa_namereg_get(m->core, source_name, PA_NAMEREG_SOURCE))) {
pa_log_error("Source \"%s\" not found", source_name);
goto fail;
}
if (!(sink = pa_namereg_get(m->core, sink_name, PA_NAMEREG_SINK))) {
pa_log_error("Sink \"%s\" not found", sink_name);
goto fail;
}
u->sink_name = pa_xstrdup(sink_name);
u->source_name = pa_xstrdup(source_name);
if (cmtspeech_check_source_api(source))
goto fail;
if (cmtspeech_check_sink_api(sink))
goto fail;
u->sink_input = NULL;
u->source_output = NULL;
u->local_sideinfoq = pa_queue_new();
u->voice_sideinfoq = NULL;
u->continuous_dl_stream = false,
u->dl_memblockq =
pa_memblockq_new("cmtspeech dl_memblockq", 0, 4*u->dl_frame_size, 0, &u->ss, 0, 0, 0, NULL);
u->mainloop_handler = cmtspeech_mainloop_handler_new(u);
if (cmtspeech_dbus_init(u, dbus_type))
goto fail;
if (cmtspeech_connection_init(u))
goto fail;
pa_modargs_free(ma);
return 0;
fail:
if (ma)
pa_modargs_free(ma);
pa__done(m);
return -1;
}
