static void
gst_deinterleave_set_pads_caps (GstDeinterleave * self, GstCaps * caps)
{
GList *l;
GstStructure *s;
gint i;
for (l = self->srcpads, i = 0; l; l = l->next, i++) {
GstPad *pad = GST_PAD (l->data);
GstCaps *srccaps;
/* Set channel position if we know it */
if (self->keep_positions) {
GstAudioChannelPosition pos[1] = { GST_AUDIO_CHANNEL_POSITION_NONE };
srccaps = gst_caps_copy (caps);
s = gst_caps_get_structure (srccaps, 0);
if (self->pos)
gst_audio_set_channel_positions (s, &self->pos[i]);
else
gst_audio_set_channel_positions (s, pos);
} else {
srccaps = caps;
}
gst_pad_set_caps (pad, srccaps);
if (self->keep_positions)
gst_caps_unref (srccaps);
}
}
static gboolean
gst_a52dec_reneg (GstA52Dec * a52dec, GstPad * pad)
{
GstAudioChannelPosition *pos;
gint channels = gst_a52dec_channels (a52dec->using_channels, &pos);
GstCaps *caps = NULL;
gboolean result = FALSE;
if (!channels)
goto done;
GST_INFO_OBJECT (a52dec, "reneg channels:%d rate:%d",
channels, a52dec->sample_rate);
caps = gst_caps_new_simple ("audio/x-raw-float",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, SAMPLE_WIDTH,
"channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, a52dec->sample_rate, NULL);
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
g_free (pos);
if (!gst_pad_set_caps (pad, caps))
goto done;
result = TRUE;
done:
if (caps)
gst_caps_unref (caps);
return result;
}
gint
main (gint argc, gchar * argv[])
{
gchar *str;
GstCaps *caps;
GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
};
/* register multichannel type */
gst_init (&argc, &argv);
gst_audio_channel_position_get_type ();
/* test some caps-string conversions */
caps = gst_caps_new_simple ("audio/x-raw-int",
"channels", G_TYPE_INT, 2, NULL);
str = gst_caps_to_string (caps);
g_print ("Test caps #1: %s\n", str);
g_free (str);
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
str = gst_caps_to_string (caps);
g_print ("Test caps #2: %s\n", str);
g_free (str);
gst_caps_unref (caps);
return 0;
}
static gboolean
gst_dtsdec_renegotiate (GstDtsDec * dts)
{
GstAudioChannelPosition *pos;
GstCaps *caps = gst_caps_from_string (DTS_CAPS);
gint channels = gst_dtsdec_channels (dts->using_channels, &pos);
gboolean result = FALSE;
if (!channels)
goto done;
GST_INFO ("dtsdec renegotiate, channels=%d, rate=%d",
channels, dts->sample_rate);
gst_caps_set_simple (caps,
"channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, (gint) dts->sample_rate, NULL);
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
g_free (pos);
if (!gst_pad_set_caps (dts->srcpad, caps))
goto done;
result = TRUE;
done:
if (caps) {
gst_caps_unref (caps);
}
return result;
}
/* if channels are less than or equal to 8, we set a default layout,
* otherwise set layout to an array of GST_AUDIO_CHANNEL_POSITION_NONE */
void
gst_jack_set_layout_on_caps (GstCaps ** caps, gint channels)
{
int c;
GValue pos = { 0 };
GValue chanpos = { 0 };
gst_caps_unref (*caps);
if (channels <= 8) {
g_assert (channels >= 1);
gst_audio_set_channel_positions (gst_caps_get_structure (*caps, 0),
default_positions[channels - 1]);
} else {
g_value_init (&chanpos, GST_TYPE_ARRAY);
g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
for (c = 0; c < channels; c++) {
g_value_set_enum (&pos, GST_AUDIO_CHANNEL_POSITION_NONE);
gst_value_array_append_value (&chanpos, &pos);
}
g_value_unset (&pos);
gst_structure_set_value (gst_caps_get_structure (*caps, 0),
"channel-positions", &chanpos);
g_value_unset (&chanpos);
}
gst_caps_ref (*caps);
}
static void
sink_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
gpointer user_data)
{
gint i;
gfloat *data;
GstCaps *caps;
gint n = GPOINTER_TO_INT (user_data);
fail_unless (GST_IS_BUFFER (buffer));
fail_unless_equals_int (GST_BUFFER_SIZE (buffer),
48000 * 2 * sizeof (gfloat));
fail_unless_equals_int (GST_BUFFER_DURATION (buffer), GST_SECOND);
caps = gst_caps_new_simple ("audio/x-raw-float",
"width", G_TYPE_INT, 32,
"channels", G_TYPE_INT, 2,
"rate", G_TYPE_INT, 48000, "endianness", G_TYPE_INT, G_BYTE_ORDER, NULL);
if (n == 0) {
GstAudioChannelPosition pos[2] =
{ GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE };
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
} else if (n == 1) {
GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
};
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
} else if (n == 2) {
GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER
};
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
}
fail_unless (gst_caps_is_equal (caps, GST_BUFFER_CAPS (buffer)));
gst_caps_unref (caps);
data = (gfloat *) GST_BUFFER_DATA (buffer);
for (i = 0; i < 48000 * 2; i += 2) {
fail_unless_equals_float (data[i], -1.0);
fail_unless_equals_float (data[i + 1], 1.0);
}
have_data++;
}
static void
gst_deinterleave_add_new_pads (GstDeinterleave * self, GstCaps * caps)
{
GstPad *pad;
guint i;
for (i = 0; i < self->channels; i++) {
gchar *name = g_strdup_printf ("src%d", i);
GstCaps *srccaps;
GstStructure *s;
pad = gst_pad_new_from_static_template (&src_template, name);
g_free (name);
/* Set channel position if we know it */
if (self->keep_positions) {
GstAudioChannelPosition pos[1] = { GST_AUDIO_CHANNEL_POSITION_NONE };
srccaps = gst_caps_copy (caps);
s = gst_caps_get_structure (srccaps, 0);
if (self->pos)
gst_audio_set_channel_positions (s, &self->pos[i]);
else
gst_audio_set_channel_positions (s, pos);
} else {
srccaps = caps;
}
gst_pad_set_caps (pad, srccaps);
gst_pad_use_fixed_caps (pad);
gst_pad_set_query_function (pad,
GST_DEBUG_FUNCPTR (gst_deinterleave_src_query));
gst_pad_set_active (pad, TRUE);
gst_element_add_pad (GST_ELEMENT (self), pad);
self->srcpads = g_list_prepend (self->srcpads, gst_object_ref (pad));
if (self->keep_positions)
gst_caps_unref (srccaps);
}
gst_element_no_more_pads (GST_ELEMENT (self));
self->srcpads = g_list_reverse (self->srcpads);
}
static void
src_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
gpointer user_data)
{
gint n = GPOINTER_TO_INT (user_data);
GstCaps *caps;
gfloat *data;
gint i;
if (GST_PAD_CAPS (pad))
caps = gst_caps_ref (GST_PAD_CAPS (pad));
else {
caps = gst_caps_new_simple ("audio/x-raw-float",
"width", G_TYPE_INT, 32,
"channels", G_TYPE_INT, 1,
"rate", G_TYPE_INT, 48000, "endianness", G_TYPE_INT, G_BYTE_ORDER,
NULL);
if (n == 2) {
GstAudioChannelPosition pos[1] =
{ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT };
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
} else if (n == 3) {
GstAudioChannelPosition pos[1] =
{ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT };
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
}
}
data = g_new (gfloat, 48000);
GST_BUFFER_MALLOCDATA (buffer) = (guint8 *) data;
GST_BUFFER_DATA (buffer) = (guint8 *) data;
GST_BUFFER_SIZE (buffer) = 48000 * sizeof (gfloat);
GST_BUFFER_OFFSET (buffer) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
GST_BUFFER_OFFSET_END (buffer) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_DURATION (buffer) = GST_SECOND;
GST_BUFFER_CAPS (buffer) = caps;
for (i = 0; i < 48000; i++)
data[i] = (n % 2 == 0) ? -1.0 : 1.0;
}
static gboolean
gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps)
{
GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
GstStructure *structure = gst_caps_get_structure (caps, 0);
/* Check if we can set the caps here already */
if (gst_structure_get_int (structure, "channels", &dec->channels) &&
gst_structure_get_int (structure, "rate", &dec->sample_rate) &&
gst_structure_get_int (structure, "width", &dec->depth)) {
GstCaps *caps;
GstAudioChannelPosition *pos;
caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, dec->sample_rate,
"channels", G_TYPE_INT, dec->channels,
"depth", G_TYPE_INT, dec->depth,
"width", G_TYPE_INT, 32,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE, NULL);
/* If we already have the channel layout set from upstream
* take this */
if (gst_structure_has_field (structure, "channel-positions")) {
pos = gst_audio_get_channel_positions (structure);
if (pos != NULL && dec->channels > 2) {
GstStructure *new_str = gst_caps_get_structure (caps, 0);
gst_audio_set_channel_positions (new_str, pos);
dec->channel_mask =
gst_wavpack_get_channel_mask_from_positions (pos, dec->channels);
}
if (pos != NULL)
g_free (pos);
}
GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
/* should always succeed */
gst_pad_set_caps (dec->srcpad, caps);
gst_caps_unref (caps);
/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
* is decoded or after the format has changed */
gst_wavpack_dec_post_tags (dec);
}
gst_object_unref (dec);
return TRUE;
}
static gboolean
gst_dvdlpcmdec_set_outcaps (GstDvdLpcmDec * dvdlpcmdec)
{
gboolean res = TRUE;
GstCaps *src_caps;
GstAudioChannelPosition *pos;
/* Build caps to set on the src pad, which we know from the incoming caps */
src_caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, dvdlpcmdec->rate,
"channels", G_TYPE_INT, dvdlpcmdec->channels,
"endianness", G_TYPE_INT, G_BIG_ENDIAN,
"depth", G_TYPE_INT, dvdlpcmdec->out_width,
"width", G_TYPE_INT, dvdlpcmdec->out_width,
"signed", G_TYPE_BOOLEAN, TRUE, NULL);
pos = get_audio_channel_positions (dvdlpcmdec);
if (pos) {
gst_audio_set_channel_positions (gst_caps_get_structure (src_caps, 0), pos);
g_free (pos);
}
GST_DEBUG_OBJECT (dvdlpcmdec, "Set rate %d, channels %d, width %d (out %d)",
dvdlpcmdec->rate, dvdlpcmdec->channels, dvdlpcmdec->width,
dvdlpcmdec->out_width);
res = gst_pad_set_caps (dvdlpcmdec->srcpad, src_caps);
if (res) {
GST_DEBUG_OBJECT (dvdlpcmdec, "Successfully set output caps: %"
GST_PTR_FORMAT, src_caps);
gst_dvdlpcmdec_send_tags (dvdlpcmdec);
} else {
GST_DEBUG_OBJECT (dvdlpcmdec, "Failed to set output caps: %"
GST_PTR_FORMAT, src_caps);
}
gst_caps_unref (src_caps);
return res;
}
GstRingBufferSpec *
gst_pulse_channel_map_to_gst (const pa_channel_map * map,
GstRingBufferSpec * spec)
{
int i;
GstAudioChannelPosition *pos;
gboolean invalid = FALSE;
g_return_val_if_fail (map->channels == spec->channels, NULL);
pos = g_new0 (GstAudioChannelPosition, spec->channels + 1);
for (i = 0; i < spec->channels; i++) {
if (map->map[i] == PA_CHANNEL_POSITION_INVALID) {
invalid = TRUE;
break;
} else if ((int) map->map[i] < (int) GST_AUDIO_CHANNEL_POSITION_NUM) {
pos[i] = pa_to_gst_pos[map->map[i] + 1];
} else {
invalid = TRUE;
break;
}
}
if (!invalid && !gst_audio_check_channel_positions (pos, spec->channels))
invalid = TRUE;
if (invalid) {
for (i = 0; i < spec->channels; i++)
pos[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
}
gst_audio_set_channel_positions (gst_caps_get_structure (spec->caps, 0), pos);
g_free (pos);
return spec;
}
static GstFlowReturn
vorbis_handle_identification_packet (GstVorbisDec * vd)
{
GstCaps *caps;
const GstAudioChannelPosition *pos = NULL;
gint width = GST_VORBIS_DEC_DEFAULT_SAMPLE_WIDTH;
switch (vd->vi.channels) {
case 1:
case 2:
/* nothing */
break;
case 3:
case 4:
case 5:
case 6:
case 7:
case 8:
pos = gst_vorbis_channel_positions[vd->vi.channels - 1];
break;
default:{
gint i;
GstAudioChannelPosition *posn =
g_new (GstAudioChannelPosition, vd->vi.channels);
GST_ELEMENT_WARNING (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("Using NONE channel layout for more than 8 channels"));
for (i = 0; i < vd->vi.channels; i++)
posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
pos = posn;
}
}
/* negotiate width with downstream */
caps = gst_pad_get_allowed_caps (vd->srcpad);
if (caps) {
if (!gst_caps_is_empty (caps)) {
GstStructure *s;
s = gst_caps_get_structure (caps, 0);
/* template ensures 16 or 32 */
gst_structure_get_int (s, "width", &width);
GST_INFO_OBJECT (vd, "using %s with %d channels and %d bit audio depth",
gst_structure_get_name (s), vd->vi.channels, width);
}
gst_caps_unref (caps);
}
vd->width = width >> 3;
/* select a copy_samples function, this way we can have specialized versions
* for mono/stereo and avoid the depth switch in tremor case */
vd->copy_samples = get_copy_sample_func (vd->vi.channels, vd->width);
caps = gst_caps_copy (gst_pad_get_pad_template_caps (vd->srcpad));
gst_caps_set_simple (caps, "rate", G_TYPE_INT, vd->vi.rate,
"channels", G_TYPE_INT, vd->vi.channels,
"width", G_TYPE_INT, width, NULL);
if (pos) {
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
}
if (vd->vi.channels > 8) {
g_free ((GstAudioChannelPosition *) pos);
}
gst_pad_set_caps (vd->srcpad, caps);
gst_caps_unref (caps);
return GST_FLOW_OK;
}
static gboolean
gst_rtp_g722_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpG722Depay *rtpg722depay;
gint clock_rate, payload, samplerate;
gint channels;
GstCaps *srccaps;
gboolean res;
const gchar *channel_order;
const GstRTPChannelOrder *order;
rtpg722depay = GST_RTP_G722_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
payload = 96;
gst_structure_get_int (structure, "payload", &payload);
switch (payload) {
case GST_RTP_PAYLOAD_G722:
channels = 1;
clock_rate = 8000;
samplerate = 16000;
break;
default:
/* no fixed mapping, we need clock-rate */
channels = 0;
clock_rate = 0;
samplerate = 0;
break;
}
/* caps can overwrite defaults */
clock_rate =
gst_rtp_g722_depay_parse_int (structure, "clock-rate", clock_rate);
if (clock_rate == 0)
goto no_clockrate;
if (clock_rate == 8000)
samplerate = 16000;
if (samplerate == 0)
samplerate = clock_rate;
channels =
gst_rtp_g722_depay_parse_int (structure, "encoding-params", channels);
if (channels == 0) {
channels = gst_rtp_g722_depay_parse_int (structure, "channels", channels);
if (channels == 0) {
/* channels defaults to 1 otherwise */
channels = 1;
}
}
depayload->clock_rate = clock_rate;
rtpg722depay->rate = samplerate;
rtpg722depay->channels = channels;
srccaps = gst_caps_new_simple ("audio/G722",
"rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL);
/* add channel positions */
channel_order = gst_structure_get_string (structure, "channel-order");
order = gst_rtp_channels_get_by_order (channels, channel_order);
if (order) {
gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0),
order->pos);
} else {
GstAudioChannelPosition *pos;
GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
(NULL), ("Unknown channel order '%s' for %d channels",
GST_STR_NULL (channel_order), channels));
/* create default NONE layout */
pos = gst_rtp_channels_create_default (channels);
gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos);
g_free (pos);
}
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
return res;
/* ERRORS */
no_clockrate:
{
GST_ERROR_OBJECT (depayload, "no clock-rate specified");
return FALSE;
}
}
static gboolean
gst_lv2_filter_setup (GstAudioFilter * gsp, const GstAudioInfo * info)
{
GstLV2Filter *self;
GstLV2FilterClass *oclass;
GstAudioFilterClass *audiofilter_class;
gint i;
audiofilter_class = GST_AUDIO_FILTER_GET_CLASS (gsp);
self = (GstLV2Filter *) gsp;
oclass = (GstLV2FilterClass *) audiofilter_class;
g_return_val_if_fail (self->activated == FALSE, FALSE);
GST_DEBUG_OBJECT (self, "instantiating the plugin at %d Hz",
GST_AUDIO_INFO_RATE (info));
if (self->instance)
lilv_instance_free (self->instance);
if (!(self->instance =
lilv_plugin_instantiate (oclass->plugin, GST_AUDIO_INFO_RATE (info),
NULL)))
goto no_instance;
/* connect the control ports */
for (i = 0; i < oclass->control_in_ports->len; i++)
lilv_instance_connect_port (self->instance,
g_array_index (oclass->control_in_ports, GstLV2FilterPort, i).index,
&(self->ports.control.in[i]));
for (i = 0; i < oclass->control_out_ports->len; i++)
lilv_instance_connect_port (self->instance,
g_array_index (oclass->control_out_ports, GstLV2FilterPort, i).index,
&(self->ports.control.out[i]));
/* FIXME Handle audio channel positionning while negotiating CAPS */
#if 0
/* set input group pad audio channel position */
for (i = 0; i < oclass->in_groups->len; ++i) {
group = &g_array_index (oclass->in_groups, GstLV2FilterGroup, i);
if (group->has_roles) {
if ((positions = gst_lv2_filter_build_positions (group))) {
if ((pad = gst_element_get_static_pad (GST_ELEMENT (gsp),
lilv_node_as_string (group->symbol)))) {
GST_INFO_OBJECT (self, "set audio channel positions on sink pad %s",
lilv_node_as_string (group->symbol));
s = gst_caps_get_structure (caps, 0);
gst_audio_set_channel_positions (s, positions);
gst_object_unref (pad);
}
g_free (positions);
positions = NULL;
}
}
}
/* set output group pad audio channel position */
for (i = 0; i < oclass->out_groups->len; ++i) {
group = &g_array_index (oclass->out_groups, GstLV2FilterGroup, i);
if (group->has_roles) {
if ((positions = gst_lv2_filter_build_positions (group))) {
if ((pad = gst_element_get_static_pad (GST_ELEMENT (gsp),
lilv_node_as_string (group->symbol)))) {
GST_INFO_OBJECT (self, "set audio channel positions on src pad %s",
lilv_node_as_string (group->symbol));
s = gst_caps_get_structure (caps, 0);
gst_audio_set_channel_positions (s, positions);
gst_object_unref (pad);
}
g_free (positions);
positions = NULL;
}
}
}
#endif
lilv_instance_activate (self->instance);
self->activated = TRUE;
return TRUE;
no_instance:
{
GST_ERROR_OBJECT (gsp, "could not create instance");
return FALSE;
}
}
void test_multichannel_checks()
{
GstAudioChannelPosition pos_2_mixed[2] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_MONO,
GST_AUDIO_CHANNEL_POSITION_NONE
};
GstAudioChannelPosition pos_2_none[2] = {
GST_AUDIO_CHANNEL_POSITION_NONE,
GST_AUDIO_CHANNEL_POSITION_NONE
};
GstAudioChannelPosition pos_2_flr[2] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
};
GstAudioChannelPosition pos_2_frr[2] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
};
GstStructure *s;
xmlfile = "test_multichannel_checks";
std_log(LOG_FILENAME_LINE, "Test Started test_multichannel_checks");
s = gst_structure_new ("audio/x-raw-int", "channels", G_TYPE_INT, 2, NULL);
/* this should not work and issue a warning: FRONT_MONO + NONE */
// _gst_check_expecting_log = TRUE;
// gst_audio_set_channel_positions (s, pos_2_mixed);
// _gst_check_expecting_log = FALSE;
// fail_if (structure_contains_channel_positions (s));
/* this should work: NONE + NONE */
gst_audio_set_channel_positions (s, pos_2_none);
fail_unless (structure_contains_channel_positions (s));
gst_structure_remove_field (s, "channel-positions");
/* this should also work: FRONT_LEFT + FRONT_RIGHT */
gst_audio_set_channel_positions (s, pos_2_flr);
fail_unless (structure_contains_channel_positions (s));
gst_structure_remove_field (s, "channel-positions");
/* this should not work and issue a warning: FRONT_RIGHT twice */
// _gst_check_expecting_log = TRUE;
// gst_audio_set_channel_positions (s, pos_2_frr);
// _gst_check_expecting_log = FALSE;
/* FIXME: did I misunderstand _set_structure_channel_positions_list? */
#if 0
/* this should not work and issue a warning: FRONT_RIGHT twice */
_gst_check_expecting_log = TRUE;
gst_audio_set_structure_channel_positions_list (s, pos_2_frr, 2);
_gst_check_expecting_log = FALSE;
/* this should not work and issue a warning: FRONT_MONO + NONE */
_gst_check_expecting_log = TRUE;
gst_audio_set_structure_channel_positions_list (s, pos_2_mixed, 2);
_gst_check_expecting_log = FALSE;
/* this should not work either (channel count mismatch) */
_gst_check_expecting_log = TRUE;
gst_audio_set_structure_channel_positions_list (s, pos_2_none, 44);
_gst_check_expecting_log = FALSE;
fail_if (structure_contains_channel_positions (s));
#endif
gst_structure_free (s);
std_log(LOG_FILENAME_LINE, "Test Successful");
create_xml(0);
}
static void
caps_add_channel_configuration (GstCaps * caps,
const GstStructure * in_structure, gint min_chans, gint max_chans)
{
GstAudioChannelPosition pos[8] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT
};
GstStructure *s = NULL;
gint c;
if (min_chans == max_chans && max_chans <= 2) {
s = get_channel_free_structure (in_structure);
gst_structure_set (s, "channels", G_TYPE_INT, max_chans, NULL);
gst_caps_append_structure (caps, s);
return;
}
g_assert (min_chans >= 1);
/* mono and stereo don't need channel configurations */
if (min_chans == 2) {
s = get_channel_free_structure (in_structure);
gst_structure_set (s, "channels", G_TYPE_INT, 2, NULL);
gst_caps_append_structure (caps, s);
} else if (min_chans == 1 && max_chans >= 2) {
s = get_channel_free_structure (in_structure);
gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
gst_caps_append_structure (caps, s);
}
/* don't know whether to use 2.1 or 3.0 here - but I suspect
* alsa might work around that/fix it somehow. Can we tell alsa
* what our channel layout is like? */
if (max_chans >= 3 && min_chans <= 3) {
GstAudioChannelPosition pos_21[3] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE
};
s = get_channel_free_structure (in_structure);
gst_structure_set (s, "channels", G_TYPE_INT, 3, NULL);
gst_audio_set_channel_positions (s, pos_21);
gst_caps_append_structure (caps, s);
}
/* everything else (4, 6, 8 channels) needs a channel layout */
for (c = MAX (4, min_chans); c <= 8; c += 2) {
if (max_chans >= c) {
s = get_channel_free_structure (in_structure);
gst_structure_set (s, "channels", G_TYPE_INT, c, NULL);
gst_audio_set_channel_positions (s, pos);
gst_caps_append_structure (caps, s);
}
}
for (c = MAX (9, min_chans); c <= max_chans; ++c) {
GstAudioChannelPosition *ch_layout;
guint i;
ch_layout = g_new (GstAudioChannelPosition, c);
for (i = 0; i < c; ++i) {
ch_layout[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
}
s = get_channel_free_structure (in_structure);
gst_structure_set (s, "channels", G_TYPE_INT, c, NULL);
gst_audio_set_channel_positions (s, ch_layout);
gst_caps_append_structure (caps, s);
g_free (ch_layout);
}
}
static gboolean
dmo_audiodec_sink_setcaps (GstPad * pad, GstCaps * caps)
{
DMOAudioDec *dec = (DMOAudioDec *) gst_pad_get_parent (pad);
DMOAudioDecClass *klass = (DMOAudioDecClass *) G_OBJECT_GET_CLASS (dec);
GstStructure *s = gst_caps_get_structure (caps, 0);
gchar *dll;
gint size;
WAVEFORMATEX * hdr = NULL;
const GValue *v;
GstBuffer *extradata = NULL;
GstCaps *out;
gboolean ret = FALSE;
guint32 fourcc;
GST_DEBUG_OBJECT (dec, "setcaps called with %" GST_PTR_FORMAT, caps);
Check_FS_Segment ();
if (dec->ctx) {
DMO_AudioDecoder_Destroy (dec->ctx);
dec->ctx = NULL;
}
/* read data */
if (!gst_structure_get_int (s, "bitrate", &dec->bitrate) ||
!gst_structure_get_int (s, "block_align", &dec->block_align) ||
!gst_structure_get_int (s, "rate", &dec->rate) ||
!gst_structure_get_int (s, "channels", &dec->channels) ||
!gst_structure_get_int (s, "depth", &dec->depth)) {
goto beach;
}
if ((v = gst_structure_get_value (s, "codec_data")))
extradata = gst_value_get_buffer (v);
if (!gst_structure_get_fourcc (s, "format", &fourcc))
fourcc = klass->entry->format;
/* set up dll initialization */
dll = g_strdup_printf ("%s.dll", klass->entry->dll);
size = sizeof (WAVEFORMATEX) +
(extradata ? GST_BUFFER_SIZE (extradata) : 0);
hdr = g_malloc0 (size);
if (extradata) { /* Codec data is appended after our header */
memcpy (((guchar *) hdr) + sizeof (WAVEFORMATEX),
GST_BUFFER_DATA (extradata), GST_BUFFER_SIZE (extradata));
hdr->cbSize = GST_BUFFER_SIZE (extradata);
}
hdr->wFormatTag = fourcc;
hdr->nChannels = dec->channels;
hdr->nSamplesPerSec = dec->rate;
hdr->nAvgBytesPerSec = dec->bitrate / 8;
hdr->nBlockAlign = dec->block_align;
hdr->wBitsPerSample = dec->depth;
GST_DEBUG ("Will now open %s using %d bps %d channels", dll, dec->bitrate,
dec->channels);
if (!(dec->ctx = DMO_AudioDecoder_Open (dll, &klass->entry->guid, hdr))) {
GST_ERROR ("Failed to open DLL %s", dll);
g_free (dll);
g_free (hdr);
goto beach;
}
g_free (dll);
g_free (hdr);
DMO_AudioDecoder_GetOutputInfos (dec->ctx, &dec->out_buffer_size,
&dec->out_align);
DMO_AudioDecoder_GetInputInfos (dec->ctx, &dec->in_buffer_size,
&dec->in_align, &dec->lookahead);
/* negotiate output */
out = gst_caps_from_string (klass->entry->srccaps);
gst_caps_set_simple (out,
"width", G_TYPE_INT, dec->depth,
"depth", G_TYPE_INT, dec->depth,
"rate", G_TYPE_INT, dec->rate,
"channels", G_TYPE_INT, dec->channels,
NULL);
if (dec->channels > 2 && dec->channels <= 11) {
GstAudioChannelPosition pos[] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT
};
gst_audio_set_channel_positions (gst_caps_get_structure (out, 0), pos);
}
if (!gst_pad_set_caps (dec->srcpad, out)) {
gst_caps_unref (out);
GST_ERROR ("Failed to negotiate output");
goto beach;
}
gst_caps_unref (out);
ret = TRUE;
beach:
gst_object_unref (dec);
return ret;
}
static GstCaps *
gst_openal_helper_probe_caps (ALCcontext * ctx)
{
static const struct
{
gint count;
GstAudioChannelPosition pos[8];
} chans[] = {
{
1, {
GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
}
}, {
2, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
}
}, {
4, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT
}
}, {
6, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT
}
}, {
7, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT
}
}, {
8, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT
}
},
};
GstStructure *structure;
ALCcontext *old;
GstCaps *caps;
old = pushContext (ctx);
caps = gst_caps_new_empty ();
if (alIsExtensionPresent ("AL_EXT_MCFORMATS")) {
const char *fmt32[] = {
"AL_FORMAT_MONO_FLOAT32", "AL_FORMAT_STEREO_FLOAT32",
"AL_FORMAT_QUAD32", "AL_FORMAT_51CHN32", "AL_FORMAT_61CHN32",
"AL_FORMAT_71CHN32", NULL
}, *fmt16[] = {
"AL_FORMAT_MONO16", "AL_FORMAT_STEREO16", "AL_FORMAT_QUAD16",
"AL_FORMAT_51CHN16", "AL_FORMAT_61CHN16", "AL_FORMAT_71CHN16", NULL
},
*fmt8[] = {
"AL_FORMAT_MONO8", "AL_FORMAT_STEREO8", "AL_FORMAT_QUAD8",
"AL_FORMAT_51CHN8", "AL_FORMAT_61CHN8", "AL_FORMAT_71CHN8", NULL
};
int i;
if (alIsExtensionPresent ("AL_EXT_FLOAT32")) {
for (i = 0; fmt32[i]; i++) {
ALenum val = alGetEnumValue (fmt32[i]);
if (checkALError () != AL_NO_ERROR || val == 0 || val == -1)
continue;
structure = gst_structure_new ("audio/x-raw-float",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
OPENAL_MAX_RATE, "width", G_TYPE_INT, 32, NULL);
gst_structure_set (structure, "channels", G_TYPE_INT,
chans[i].count, NULL);
if (chans[i].count > 2)
gst_audio_set_channel_positions (structure, chans[i].pos);
gst_caps_append_structure (caps, structure);
}
}
for (i = 0; fmt16[i]; i++) {
ALenum val = alGetEnumValue (fmt16[i]);
if (checkALError () != AL_NO_ERROR || val == 0 || val == -1)
continue;
structure = gst_structure_new ("audio/x-raw-int",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
gst_structure_set (structure, "channels", G_TYPE_INT,
chans[i].count, NULL);
if (chans[i].count > 2)
gst_audio_set_channel_positions (structure, chans[i].pos);
gst_caps_append_structure (caps, structure);
}
for (i = 0; fmt8[i]; i++) {
ALenum val = alGetEnumValue (fmt8[i]);
if (checkALError () != AL_NO_ERROR || val == 0 || val == -1)
continue;
structure = gst_structure_new ("audio/x-raw-int",
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
"width", G_TYPE_INT, 8,
"depth", G_TYPE_INT, 8, "signed", G_TYPE_BOOLEAN, FALSE, NULL);
gst_structure_set (structure, "channels", G_TYPE_INT,
chans[i].count, NULL);
if (chans[i].count > 2)
gst_audio_set_channel_positions (structure, chans[i].pos);
gst_caps_append_structure (caps, structure);
}
} else {
if (alIsExtensionPresent ("AL_EXT_FLOAT32")) {
structure = gst_structure_new ("audio/x-raw-float",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
"width", G_TYPE_INT, 32, "channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
gst_caps_append_structure (caps, structure);
}
structure = gst_structure_new ("audio/x-raw-int",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16,
"signed", G_TYPE_BOOLEAN, TRUE,
"channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
gst_caps_append_structure (caps, structure);
structure = gst_structure_new ("audio/x-raw-int",
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
"width", G_TYPE_INT, 8,
"depth", G_TYPE_INT, 8,
"signed", G_TYPE_BOOLEAN, FALSE,
"channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
gst_caps_append_structure (caps, structure);
}
if (alIsExtensionPresent ("AL_EXT_MULAW_MCFORMATS")) {
const char *fmtmulaw[] = {
"AL_FORMAT_MONO_MULAW", "AL_FORMAT_STEREO_MULAW",
"AL_FORMAT_QUAD_MULAW", "AL_FORMAT_51CHN_MULAW",
"AL_FORMAT_61CHN_MULAW", "AL_FORMAT_71CHN_MULAW", NULL
};
int i;
for (i = 0; fmtmulaw[i]; i++) {
ALenum val = alGetEnumValue (fmtmulaw[i]);
if (checkALError () != AL_NO_ERROR || val == 0 || val == -1)
continue;
structure = gst_structure_new ("audio/x-mulaw",
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE, NULL);
gst_structure_set (structure, "channels", G_TYPE_INT,
chans[i].count, NULL);
if (chans[i].count > 2)
gst_audio_set_channel_positions (structure, chans[i].pos);
gst_caps_append_structure (caps, structure);
}
} else if (alIsExtensionPresent ("AL_EXT_MULAW")) {
structure = gst_structure_new ("audio/x-mulaw",
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
"channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
gst_caps_append_structure (caps, structure);
}
popContext (old, ctx);
return caps;
}
static GstFlowReturn
gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
{
const guint8 *data = GST_BUFFER_DATA (buf);
GstCaps *caps;
const GstAudioChannelPosition *pos = NULL;
if (!gst_opus_header_is_id_header (buf)) {
GST_ERROR_OBJECT (dec, "Header is not an Opus ID header");
return GST_FLOW_ERROR;
}
if (!(dec->n_channels == 0 || dec->n_channels == data[9])) {
GST_ERROR_OBJECT (dec, "Opus ID header has invalid channels");
return GST_FLOW_ERROR;
}
dec->n_channels = data[9];
dec->pre_skip = GST_READ_UINT16_LE (data + 10);
dec->r128_gain = GST_READ_UINT16_LE (data + 16);
dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain);
GST_INFO_OBJECT (dec,
"Found pre-skip of %u samples, R128 gain %d (volume %f)",
dec->pre_skip, dec->r128_gain, dec->r128_gain_volume);
dec->channel_mapping_family = data[18];
if (dec->channel_mapping_family == 0) {
/* implicit mapping */
GST_INFO_OBJECT (dec, "Channel mapping family 0, implicit mapping");
dec->n_streams = dec->n_stereo_streams = 1;
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
} else {
dec->n_streams = data[19];
dec->n_stereo_streams = data[20];
memcpy (dec->channel_mapping, data + 21, dec->n_channels);
if (dec->channel_mapping_family == 1) {
GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
switch (dec->n_channels) {
case 1:
case 2:
/* nothing */
break;
case 3:
case 4:
case 5:
case 6:
case 7:
case 8:
pos = gst_opus_channel_positions[dec->n_channels - 1];
break;
default:{
gint i;
GstAudioChannelPosition *posn =
g_new (GstAudioChannelPosition, dec->n_channels);
GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("Using NONE channel layout for more than 8 channels"));
for (i = 0; i < dec->n_channels; i++)
posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
pos = posn;
}
}
} else {
GST_INFO_OBJECT (dec, "Channel mapping family %d",
dec->channel_mapping_family);
}
}
caps = gst_opus_dec_negotiate (dec);
if (pos) {
GST_DEBUG_OBJECT (dec, "Setting channel positions on caps");
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
}
if (dec->n_channels > 8) {
g_free ((GstAudioChannelPosition *) pos);
}
GST_INFO_OBJECT (dec, "Setting src caps to %" GST_PTR_FORMAT, caps);
gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
gst_caps_unref (caps);
return GST_FLOW_OK;
}
