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faac: make template pad caps more accurate and remove custom getcaps

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Allows reusing baseclass caps query handling and simplifying negotiation
code.
This commit is contained in:
Thiago Santos 2015-08-17 07:15:00 -03:00
parent f3b18a29bf
commit ba4e6ee1be
1 changed files with 50 additions and 63 deletions
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113
ext/faac/gstfaac.c
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@ -55,12 +55,6 @@
"64000, " \
"88200, " \
"96000"
#define SINK_CAPS \
"audio/x-raw, " \
"format = (string) "GST_AUDIO_NE (S16) ", " \
"layout = (string) interleaved, " \
"rate = (int) {" SAMPLE_RATES "}, " \
"channels = (int) [ 1, 6 ] "
/* these don't seem to work? */
#if 0
@ -96,11 +90,6 @@ static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_ALWAYS,
GST_STATIC_CAPS (SRC_CAPS));
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (SINK_CAPS));
enum
{
PROP_0,
@ -124,9 +113,9 @@ static void gst_faac_set_property (GObject * object,
static void gst_faac_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstCaps *gst_faac_enc_generate_sink_caps (void);
static gboolean gst_faac_configure_source_pad (GstFaac * faac,
GstAudioInfo * info);
static GstCaps *gst_faac_getcaps (GstAudioEncoder * enc, GstCaps * filter);
static gboolean gst_faac_stop (GstAudioEncoder * enc);
static gboolean gst_faac_set_format (GstAudioEncoder * enc,
@ -194,14 +183,20 @@ gst_faac_class_init (GstFaacClass * klass)
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
GstCaps *sink_caps;
GstPadTemplate *sink_templ;
gobject_class->set_property = gst_faac_set_property;
gobject_class->get_property = gst_faac_get_property;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sink_template));
sink_caps = gst_faac_enc_generate_sink_caps ();
sink_templ = gst_pad_template_new ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, sink_caps);
gst_element_class_add_pad_template (gstelement_class, sink_templ);
gst_caps_unref (sink_caps);
gst_element_class_set_static_metadata (gstelement_class, "AAC audio encoder",
"Codec/Encoder/Audio",
@ -211,7 +206,6 @@ gst_faac_class_init (GstFaacClass * klass)
base_class->stop = GST_DEBUG_FUNCPTR (gst_faac_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_faac_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_faac_handle_frame);
base_class->getcaps = GST_DEBUG_FUNCPTR (gst_faac_getcaps);
/* properties */
g_object_class_install_property (gobject_class, PROP_QUALITY,
@ -299,58 +293,51 @@ static const GstAudioChannelPosition aac_channel_positions[][8] = {
};
static GstCaps *
gst_faac_getcaps (GstAudioEncoder * enc, GstCaps * filter)
gst_faac_enc_generate_sink_caps (void)
{
static volatile gsize sinkcaps = 0;
GstCaps *caps = gst_caps_new_empty ();
GstStructure *s, *t;
gint i, c;
static const int rates[] = {
8000, 11025, 12000, 16000, 22050, 24000,
32000, 44100, 48000, 64000, 88200, 96000
};
GValue rates_arr = { 0, };
GValue tmp_v = { 0, };
if (g_once_init_enter (&sinkcaps)) {
GstCaps *tmp = gst_caps_new_empty ();
GstStructure *s, *t;
gint i, c;
static const int rates[] = {
8000, 11025, 12000, 16000, 22050, 24000,
32000, 44100, 48000, 64000, 88200, 96000
};
GValue rates_arr = { 0, };
GValue tmp_v = { 0, };
g_value_init (&rates_arr, GST_TYPE_LIST);
g_value_init (&tmp_v, G_TYPE_INT);
for (i = 0; i < G_N_ELEMENTS (rates); i++) {
g_value_set_int (&tmp_v, rates[i]);
gst_value_list_append_value (&rates_arr, &tmp_v);
}
g_value_unset (&tmp_v);
s = gst_structure_new ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
"layout", G_TYPE_STRING, "interleaved", NULL);
gst_structure_set_value (s, "rate", &rates_arr);
for (i = 1; i <= 6; i++) {
guint64 channel_mask = 0;
t = gst_structure_copy (s);
gst_structure_set (t, "channels", G_TYPE_INT, i, NULL);
if (i > 1) {
for (c = 0; c < i; c++)
channel_mask |=
G_GUINT64_CONSTANT (1) << aac_channel_positions[i - 1][c];
gst_structure_set (t, "channel-mask", GST_TYPE_BITMASK, channel_mask,
NULL);
}
gst_caps_append_structure (tmp, t);
}
gst_structure_free (s);
g_value_unset (&rates_arr);
GST_DEBUG_OBJECT (enc, "Generated sinkcaps: %" GST_PTR_FORMAT, tmp);
g_once_init_leave (&sinkcaps, (gsize) tmp);
g_value_init (&rates_arr, GST_TYPE_LIST);
g_value_init (&tmp_v, G_TYPE_INT);
for (i = 0; i < G_N_ELEMENTS (rates); i++) {
g_value_set_int (&tmp_v, rates[i]);
gst_value_list_append_value (&rates_arr, &tmp_v);
}
g_value_unset (&tmp_v);
return gst_audio_encoder_proxy_getcaps (enc, (GstCaps *) sinkcaps, filter);
s = gst_structure_new ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
"layout", G_TYPE_STRING, "interleaved", NULL);
gst_structure_set_value (s, "rate", &rates_arr);
t = gst_structure_copy (s);
gst_structure_set (t, "channels", G_TYPE_INT, 1, NULL);
gst_caps_append_structure (caps, t);
for (i = 2; i <= 6; i++) {
guint64 channel_mask = 0;
t = gst_structure_copy (s);
gst_structure_set (t, "channels", G_TYPE_INT, i, NULL);
for (c = 0; c < i; c++)
channel_mask |= G_GUINT64_CONSTANT (1) << aac_channel_positions[i - 1][c];
gst_structure_set (t, "channel-mask", GST_TYPE_BITMASK, channel_mask, NULL);
gst_caps_append_structure (caps, t);
}
gst_structure_free (s);
g_value_unset (&rates_arr);
GST_DEBUG ("Generated sinkcaps: %" GST_PTR_FORMAT, caps);
return caps;
}
static gboolean