static void
sink_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
gpointer user_data)
{
gint i;
GstMapInfo map;
gfloat *data;
GstCaps *caps, *ccaps;
gint n = GPOINTER_TO_INT (user_data);
guint64 mask;
fail_unless (GST_IS_BUFFER (buffer));
gst_buffer_map (buffer, &map, GST_MAP_READ);
data = (gfloat *) map.data;
fail_unless_equals_int (map.size, 48000 * 2 * sizeof (gfloat));
fail_unless_equals_int (GST_BUFFER_DURATION (buffer), GST_SECOND);
if (n == 0) {
GstAudioChannelPosition pos[2] =
{ GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE };
gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask);
} else if (n == 1) {
GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
};
gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask);
} else if (n == 2) {
GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER
};
gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask);
} else {
g_assert_not_reached ();
}
caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (F32),
"channels", G_TYPE_INT, 2, "rate", G_TYPE_INT, 48000,
"layout", G_TYPE_STRING, "interleaved",
"channel-mask", GST_TYPE_BITMASK, mask, NULL);
ccaps = gst_pad_get_current_caps (pad);
fail_unless (gst_caps_is_equal (caps, ccaps));
gst_caps_unref (ccaps);
gst_caps_unref (caps);
#ifdef HAVE_VALGRIND
if (!(RUNNING_ON_VALGRIND))
#endif
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);
}
gst_buffer_unmap (buffer, &map);
have_data++;
}
static gboolean
gst_interleave_channel_positions_to_mask (GValueArray * positions,
gint default_ordering_map[64], guint64 * mask)
{
gint i;
guint channels;
GstAudioChannelPosition *pos;
gboolean ret;
channels = positions->n_values;
pos = g_new (GstAudioChannelPosition, channels);
for (i = 0; i < channels; i++) {
GValue *val;
val = g_value_array_get_nth (positions, i);
pos[i] = g_value_get_enum (val);
}
/* sort the default ordering map according to the position order */
for (i = 0; i < channels; i++) {
default_ordering_map[i] = i;
}
g_qsort_with_data (default_ordering_map, channels,
sizeof (*default_ordering_map), compare_positions, pos);
ret = gst_audio_channel_positions_to_mask (pos, channels, FALSE, mask);
g_free (pos);
return ret;
}
static GstCaps *
gst_fdkaacenc_get_caps (GstAudioEncoder * enc, GstCaps * filter)
{
GstCaps *res, *caps;
gint i;
caps = gst_caps_new_empty ();
for (i = 0; i < G_N_ELEMENTS (channel_layouts); i++) {
guint64 channel_mask;
GstCaps *tmp =
gst_caps_make_writable (gst_pad_get_pad_template_caps
(GST_AUDIO_ENCODER_SINK_PAD (enc)));
if (channel_layouts[i].channels == 1) {
gst_caps_set_simple (tmp, "channels", G_TYPE_INT,
channel_layouts[i].channels, NULL);
} else {
gst_audio_channel_positions_to_mask (channel_layouts[i].positions,
channel_layouts[i].channels, FALSE, &channel_mask);
gst_caps_set_simple (tmp, "channels", G_TYPE_INT,
channel_layouts[i].channels, "channel-mask", GST_TYPE_BITMASK,
channel_mask, NULL);
}
gst_caps_append (caps, tmp);
}
res = gst_audio_encoder_proxy_getcaps (enc, caps, filter);
gst_caps_unref (caps);
return res;
}
static gboolean
gst_interleave_channel_positions_to_mask (GValueArray * positions,
guint64 * mask)
{
gint i;
guint channels;
GstAudioChannelPosition *pos;
gboolean ret;
channels = positions->n_values;
pos = g_new (GstAudioChannelPosition, channels);
for (i = 0; i < channels; i++) {
GValue *val;
val = g_value_array_get_nth (positions, i);
pos[i] = g_value_get_enum (val);
}
ret = gst_audio_channel_positions_to_mask (pos, channels, FALSE, mask);
g_free (pos);
return ret;
}
static GstCaps *
gst_openal_helper_probe_caps (ALCcontext * context)
{
static const struct
{
gint count;
GstAudioChannelPosition positions[8];
} chans[] = {
{
1, {
GST_AUDIO_CHANNEL_POSITION_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_LFE1,
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_LFE1,
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_LFE1,
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;
guint64 channel_mask;
GstCaps *caps;
ALCcontext *old;
old = pushContext (context);
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 value = alGetEnumValue (fmt32[i]);
if (checkALError () != AL_NO_ERROR || value == 0 || value == -1)
continue;
structure =
gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
GST_AUDIO_NE (F32), "rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
OPENAL_MAX_RATE, "channels", G_TYPE_INT, chans[i].count, NULL);
if (chans[i].count > 2) {
gst_audio_channel_positions_to_mask (chans[i].positions,
chans[i].count, FALSE, &channel_mask);
gst_structure_set (structure, "channel-mask", GST_TYPE_BITMASK,
channel_mask, NULL);
}
gst_caps_append_structure (caps, structure);
}
}
for (i = 0; fmt16[i]; i++) {
ALenum value = alGetEnumValue (fmt16[i]);
if (checkALError () != AL_NO_ERROR || value == 0 || value == -1)
continue;
structure =
gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
GST_AUDIO_NE (S16), "rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
OPENAL_MAX_RATE, "channels", G_TYPE_INT, chans[i].count, NULL);
if (chans[i].count > 2) {
gst_audio_channel_positions_to_mask (chans[i].positions, chans[i].count,
FALSE, &channel_mask);
gst_structure_set (structure, "channel-mask", GST_TYPE_BITMASK,
channel_mask, NULL);
}
gst_caps_append_structure (caps, structure);
}
for (i = 0; fmt8[i]; i++) {
ALenum value = alGetEnumValue (fmt8[i]);
if (checkALError () != AL_NO_ERROR || value == 0 || value == -1)
continue;
structure =
gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
G_STRINGIFY (U8), "rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
OPENAL_MAX_RATE, "channels", G_TYPE_INT, chans[i].count, NULL);
if (chans[i].count > 2) {
gst_audio_channel_positions_to_mask (chans[i].positions, chans[i].count,
FALSE, &channel_mask);
gst_structure_set (structure, "channel-mask", GST_TYPE_BITMASK,
channel_mask, NULL);
}
gst_caps_append_structure (caps, structure);
}
} else {
if (alIsExtensionPresent ("AL_EXT_FLOAT32")) {
structure =
gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
GST_AUDIO_NE (F32), "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);
}
structure =
gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
GST_AUDIO_NE (S16), "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);
structure =
gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
G_STRINGIFY (U8), "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);
}
if (alIsExtensionPresent ("AL_EXT_double")) {
structure =
gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
GST_AUDIO_NE (F64), "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);
}
if (alIsExtensionPresent ("AL_EXT_IMA4")) {
structure =
gst_structure_new ("audio/x-adpcm", "layout", G_TYPE_STRING, "ima",
"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);
}
if (alIsExtensionPresent ("AL_EXT_ALAW")) {
structure =
gst_structure_new ("audio/x-alaw", "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);
}
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 value = alGetEnumValue (fmtmulaw[i]);
if (checkALError () != AL_NO_ERROR || value == 0 || value == -1)
continue;
structure =
gst_structure_new ("audio/x-mulaw", "rate", GST_TYPE_INT_RANGE,
OPENAL_MIN_RATE, OPENAL_MAX_RATE, "channels", G_TYPE_INT,
chans[i].count, NULL);
if (chans[i].count > 2) {
gst_audio_channel_positions_to_mask (chans[i].positions, chans[i].count,
FALSE, &channel_mask);
gst_structure_set (structure, "channel-mask", GST_TYPE_BITMASK,
channel_mask, NULL);
}
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, context);
return caps;
}
static gboolean
gst_fdkaacenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstFdkAacEnc *self = GST_FDKAACENC (enc);
gboolean ret = FALSE;
GstCaps *allowed_caps;
GstCaps *src_caps;
AACENC_ERROR err;
gint transmux = 0, aot = AOT_AAC_LC;
gint mpegversion = 4;
CHANNEL_MODE channel_mode;
AACENC_InfoStruct enc_info = { 0 };
gint bitrate;
if (self->enc) {
/* drain */
gst_fdkaacenc_handle_frame (enc, NULL);
aacEncClose (&self->enc);
}
allowed_caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (self));
GST_DEBUG_OBJECT (self, "allowed caps: %" GST_PTR_FORMAT, allowed_caps);
if (allowed_caps && gst_caps_get_size (allowed_caps) > 0) {
GstStructure *s = gst_caps_get_structure (allowed_caps, 0);
const gchar *str = NULL;
if ((str = gst_structure_get_string (s, "stream-format"))) {
if (strcmp (str, "adts") == 0) {
GST_DEBUG_OBJECT (self, "use ADTS format for output");
transmux = 2;
} else if (strcmp (str, "adif") == 0) {
GST_DEBUG_OBJECT (self, "use ADIF format for output");
transmux = 1;
} else if (strcmp (str, "raw") == 0) {
GST_DEBUG_OBJECT (self, "use RAW format for output");
transmux = 0;
}
}
gst_structure_get_int (s, "mpegversion", &mpegversion);
}
if (allowed_caps)
gst_caps_unref (allowed_caps);
err = aacEncOpen (&self->enc, 0, GST_AUDIO_INFO_CHANNELS (info));
if (err != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to open encoder: %d\n", err);
return FALSE;
}
aot = AOT_AAC_LC;
if ((err = aacEncoder_SetParam (self->enc, AACENC_AOT, aot)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set AOT %d: %d\n", aot, err);
return FALSE;
}
if ((err = aacEncoder_SetParam (self->enc, AACENC_SAMPLERATE,
GST_AUDIO_INFO_RATE (info))) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set sample rate %d: %d\n",
GST_AUDIO_INFO_RATE (info), err);
return FALSE;
}
if (GST_AUDIO_INFO_CHANNELS (info) == 1) {
channel_mode = MODE_1;
self->need_reorder = FALSE;
self->aac_positions = NULL;
} else {
guint64 in_channel_mask, out_channel_mask;
gint i;
for (i = 0; i < G_N_ELEMENTS (channel_layouts); i++) {
if (channel_layouts[i].channels != GST_AUDIO_INFO_CHANNELS (info))
continue;
gst_audio_channel_positions_to_mask (&GST_AUDIO_INFO_POSITION (info, 0),
GST_AUDIO_INFO_CHANNELS (info), FALSE, &in_channel_mask);
gst_audio_channel_positions_to_mask (channel_layouts[i].positions,
channel_layouts[i].channels, FALSE, &out_channel_mask);
if (in_channel_mask == out_channel_mask) {
channel_mode = channel_layouts[i].mode;
self->need_reorder =
memcmp (channel_layouts[i].positions,
&GST_AUDIO_INFO_POSITION (info, 0),
GST_AUDIO_INFO_CHANNELS (info) *
sizeof (GstAudioChannelPosition)) != 0;
self->aac_positions = channel_layouts[i].positions;
break;
}
}
if (i == G_N_ELEMENTS (channel_layouts)) {
GST_ERROR_OBJECT (self, "Couldn't find a valid channel layout");
return FALSE;
}
}
if ((err = aacEncoder_SetParam (self->enc, AACENC_CHANNELMODE,
channel_mode)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set channel mode %d: %d", channel_mode,
err);
return FALSE;
}
/* MPEG channel order */
if ((err = aacEncoder_SetParam (self->enc, AACENC_CHANNELORDER,
0)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set channel order %d: %d", channel_mode,
err);
return FALSE;
}
bitrate = self->bitrate;
/* See
* http://wiki.hydrogenaud.io/index.php?title=Fraunhofer_FDK_AAC#Recommended_Sampling_Rate_and_Bitrate_Combinations
*/
if (bitrate == 0) {
if (GST_AUDIO_INFO_CHANNELS (info) == 1) {
if (GST_AUDIO_INFO_RATE (info) < 16000) {
bitrate = 8000;
} else if (GST_AUDIO_INFO_RATE (info) == 16000) {
bitrate = 16000;
} else if (GST_AUDIO_INFO_RATE (info) < 32000) {
bitrate = 24000;
} else if (GST_AUDIO_INFO_RATE (info) == 32000) {
bitrate = 32000;
} else if (GST_AUDIO_INFO_RATE (info) <= 44100) {
bitrate = 56000;
} else {
bitrate = 160000;
}
} else if (GST_AUDIO_INFO_CHANNELS (info) == 2) {
if (GST_AUDIO_INFO_RATE (info) < 16000) {
bitrate = 16000;
} else if (GST_AUDIO_INFO_RATE (info) == 16000) {
bitrate = 24000;
} else if (GST_AUDIO_INFO_RATE (info) < 22050) {
bitrate = 32000;
} else if (GST_AUDIO_INFO_RATE (info) < 32000) {
bitrate = 40000;
} else if (GST_AUDIO_INFO_RATE (info) == 32000) {
bitrate = 96000;
} else if (GST_AUDIO_INFO_RATE (info) <= 44100) {
bitrate = 112000;
} else {
bitrate = 320000;
}
} else {
/* 5, 5.1 */
if (GST_AUDIO_INFO_RATE (info) < 32000) {
bitrate = 160000;
} else if (GST_AUDIO_INFO_RATE (info) <= 44100) {
bitrate = 240000;
} else {
bitrate = 320000;
}
}
}
if ((err = aacEncoder_SetParam (self->enc, AACENC_TRANSMUX,
transmux)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set transmux %d: %d", transmux, err);
return FALSE;
}
if ((err = aacEncoder_SetParam (self->enc, AACENC_BITRATE,
bitrate)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set bitrate %d: %d", bitrate, err);
return FALSE;
}
if ((err = aacEncEncode (self->enc, NULL, NULL, NULL, NULL)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to initialize encoder: %d", err);
return FALSE;
}
if ((err = aacEncInfo (self->enc, &enc_info)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to get encoder info: %d", err);
return FALSE;
}
gst_audio_encoder_set_frame_max (enc, 1);
gst_audio_encoder_set_frame_samples_min (enc, enc_info.frameLength);
gst_audio_encoder_set_frame_samples_max (enc, enc_info.frameLength);
gst_audio_encoder_set_hard_min (enc, FALSE);
self->outbuf_size = enc_info.maxOutBufBytes;
self->samples_per_frame = enc_info.frameLength;
src_caps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, mpegversion,
"channels", G_TYPE_INT, GST_AUDIO_INFO_CHANNELS (info),
"framed", G_TYPE_BOOLEAN, TRUE,
"rate", G_TYPE_INT, GST_AUDIO_INFO_RATE (info), NULL);
/* raw */
if (transmux == 0) {
GstBuffer *codec_data =
gst_buffer_new_wrapped (g_memdup (enc_info.confBuf, enc_info.confSize),
enc_info.confSize);
gst_caps_set_simple (src_caps, "codec_data", GST_TYPE_BUFFER, codec_data,
"stream-format", G_TYPE_STRING, "raw", NULL);
gst_buffer_unref (codec_data);
} else if (transmux == 1) {
gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "adif",
NULL);
} else if (transmux == 2) {
gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "adts",
NULL);
} else {
g_assert_not_reached ();
}
gst_codec_utils_aac_caps_set_level_and_profile (src_caps, enc_info.confBuf,
enc_info.confSize);
ret = gst_audio_encoder_set_output_format (enc, src_caps);
gst_caps_unref (src_caps);
return ret;
}
/**
* gst_audio_info_to_caps:
* @info: a #GstAudioInfo
*
* Convert the values of @info into a #GstCaps.
*
* Returns: (transfer full): the new #GstCaps containing the
* info of @info.
*/
GstCaps *
gst_audio_info_to_caps (const GstAudioInfo * info)
{
GstCaps *caps;
const gchar *format;
const gchar *layout;
GstAudioFlags flags;
g_return_val_if_fail (info != NULL, NULL);
g_return_val_if_fail (info->finfo != NULL, NULL);
g_return_val_if_fail (info->finfo->format != GST_AUDIO_FORMAT_UNKNOWN, NULL);
format = gst_audio_format_to_string (info->finfo->format);
g_return_val_if_fail (format != NULL, NULL);
if (info->layout == GST_AUDIO_LAYOUT_INTERLEAVED)
layout = "interleaved";
else if (info->layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED)
layout = "non-interleaved";
else
g_return_val_if_reached (NULL);
flags = info->flags;
if ((flags & GST_AUDIO_FLAG_UNPOSITIONED) && info->channels > 1
&& info->position[0] != GST_AUDIO_CHANNEL_POSITION_NONE) {
flags &= ~GST_AUDIO_FLAG_UNPOSITIONED;
g_warning ("Unpositioned audio channel position flag set but "
"channel positions present");
} else if (!(flags & GST_AUDIO_FLAG_UNPOSITIONED) && info->channels > 1
&& info->position[0] == GST_AUDIO_CHANNEL_POSITION_NONE) {
flags |= GST_AUDIO_FLAG_UNPOSITIONED;
g_warning ("Unpositioned audio channel position flag not set "
"but no channel positions present");
}
caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, format,
"layout", G_TYPE_STRING, layout,
"rate", G_TYPE_INT, info->rate,
"channels", G_TYPE_INT, info->channels, NULL);
if (info->channels > 1
|| info->position[0] != GST_AUDIO_CHANNEL_POSITION_MONO) {
guint64 channel_mask = 0;
if ((flags & GST_AUDIO_FLAG_UNPOSITIONED)) {
channel_mask = 0;
} else {
if (!gst_audio_channel_positions_to_mask (info->position, info->channels,
TRUE, &channel_mask))
goto invalid_channel_positions;
}
if (info->channels == 1
&& info->position[0] == GST_AUDIO_CHANNEL_POSITION_MONO) {
/* Default mono special case */
} else {
gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, channel_mask,
NULL);
}
}
return caps;
invalid_channel_positions:
{
GST_ERROR ("Invalid channel positions");
gst_caps_unref (caps);
return NULL;
}
}
static gboolean
gst_osx_audio_sink_allowed_caps (GstOsxAudioSink * osxsink)
{
gint i, channels;
gboolean spdif_allowed;
AudioChannelLayout *layout;
GstElementClass *element_class;
GstPadTemplate *pad_template;
GstCaps *caps, *in_caps;
guint64 channel_mask = 0;
GstAudioChannelPosition *pos = osxsink->channel_positions;
/* First collect info about the HW capabilites and preferences */
spdif_allowed =
gst_core_audio_audio_device_is_spdif_avail (osxsink->device_id);
layout = gst_core_audio_audio_device_get_channel_layout (osxsink->device_id);
GST_DEBUG_OBJECT (osxsink, "Selected device ID: %u SPDIF allowed: %d",
(unsigned) osxsink->device_id, spdif_allowed);
if (layout) {
channels = MIN (layout->mNumberChannelDescriptions,
GST_OSX_AUDIO_MAX_CHANNEL);
} else {
GST_WARNING_OBJECT (osxsink, "This driver does not support "
"kAudioDevicePropertyPreferredChannelLayout.");
channels = 2;
}
switch (channels) {
case 0:
pos[0] = GST_AUDIO_CHANNEL_POSITION_NONE;
break;
case 1:
pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
break;
case 2:
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
channel_mask |= GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT);
channel_mask |= GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT);
break;
default:
channels = MIN (layout->mNumberChannelDescriptions,
GST_OSX_AUDIO_MAX_CHANNEL);
for (i = 0; i < channels; i++) {
switch (layout->mChannelDescriptions[i].mChannelLabel) {
case kAudioChannelLabel_Left:
pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
break;
case kAudioChannelLabel_Right:
pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
break;
case kAudioChannelLabel_Center:
pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
break;
case kAudioChannelLabel_LFEScreen:
pos[i] = GST_AUDIO_CHANNEL_POSITION_LFE1;
break;
case kAudioChannelLabel_LeftSurround:
pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
break;
case kAudioChannelLabel_RightSurround:
pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
break;
case kAudioChannelLabel_RearSurroundLeft:
pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
break;
case kAudioChannelLabel_RearSurroundRight:
pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
break;
case kAudioChannelLabel_CenterSurround:
pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
break;
default:
GST_WARNING_OBJECT (osxsink, "unrecognized channel: %d",
(int) layout->mChannelDescriptions[i].mChannelLabel);
channel_mask = 0;
channels = 2;
break;
}
}
}
g_free (layout);
/* Recover the template caps */
element_class = GST_ELEMENT_GET_CLASS (osxsink);
pad_template = gst_element_class_get_pad_template (element_class, "sink");
in_caps = gst_pad_template_get_caps (pad_template);
/* Create the allowed subset */
caps = gst_caps_new_empty ();
for (i = 0; i < gst_caps_get_size (in_caps); i++) {
GstStructure *in_s, *out_s;
in_s = gst_caps_get_structure (in_caps, i);
if (gst_structure_has_name (in_s, "audio/x-ac3") ||
gst_structure_has_name (in_s, "audio/x-dts")) {
if (spdif_allowed) {
gst_caps_append_structure (caps, gst_structure_copy (in_s));
}
}
gst_audio_channel_positions_to_mask (pos, channels, false, &channel_mask);
out_s = gst_structure_copy (in_s);
gst_structure_remove_fields (out_s, "channels", "channel-mask", NULL);
gst_structure_set (out_s, "channels", G_TYPE_INT, channels,
"channel-mask", GST_TYPE_BITMASK, channel_mask, NULL);
gst_caps_append_structure (caps, out_s);
}
if (osxsink->cached_caps) {
gst_caps_unref (osxsink->cached_caps);
}
osxsink->cached_caps = caps;
osxsink->channels = channels;
return TRUE;
}
static gboolean
gst_raw_audio_parse_config_to_caps (GstRawAudioParse * raw_audio_parse,
GstCaps ** caps, GstRawAudioParseConfig * config)
{
gboolean ret = TRUE;
GstAudioChannelPosition *channel_positions;
g_assert (caps != NULL);
if (config->bpf == 0) {
GST_ERROR_OBJECT (raw_audio_parse,
"cannot convert config to caps - config not filled with valid values");
*caps = NULL;
return FALSE;
}
channel_positions =
config->needs_channel_reordering ? &(config->
reordered_channel_positions[0]) : &(config->channel_positions[0]);
switch (config->format) {
case GST_RAW_AUDIO_PARSE_FORMAT_PCM:
{
GstAudioInfo info;
gst_audio_info_init (&info);
gst_audio_info_set_format (&info,
config->pcm_format,
config->sample_rate, config->num_channels, channel_positions);
*caps = gst_audio_info_to_caps (&info);
break;
}
case GST_RAW_AUDIO_PARSE_FORMAT_ALAW:
case GST_RAW_AUDIO_PARSE_FORMAT_MULAW:
{
guint64 channel_mask;
if (!gst_audio_channel_positions_to_mask (channel_positions,
config->num_channels, TRUE, &channel_mask)) {
GST_ERROR_OBJECT (raw_audio_parse, "invalid channel positions");
ret = FALSE;
break;
}
*caps = gst_caps_new_simple (
(config->format ==
GST_RAW_AUDIO_PARSE_FORMAT_ALAW) ? "audio/x-alaw" :
"audio/x-mulaw", "rate", G_TYPE_INT, config->sample_rate, "channels",
G_TYPE_INT, config->num_channels, "channel-mask", GST_TYPE_BITMASK,
channel_mask, NULL);
break;
}
default:
g_assert_not_reached ();
ret = FALSE;
}
if (!ret)
*caps = NULL;
return ret;
}
