GST_DEBUG_CATEGORY_STATIC (speexenc_debug);
#define GST_CAT_DEFAULT speexenc_debug
+#define FORMAT_STR GST_AUDIO_NE(S16)
+
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "rate = (int) [ 6000, 48000 ], "
- "channels = (int) [ 1, 2 ], "
- "endianness = (int) BYTE_ORDER, "
- "signed = (boolean) TRUE, " "width = (int) 16, " "depth = (int) 16")
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " FORMAT_STR ", "
+ "rate = (int) [ 6000, 48000 ], " "channels = (int) [ 1, 2 ]")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
static GstFlowReturn
gst_speex_enc_pre_push (GstAudioEncoder * benc, GstBuffer ** buffer);
-static void
-gst_speex_enc_setup_interfaces (GType speexenc_type)
-{
- static const GInterfaceInfo tag_setter_info = { NULL, NULL, NULL };
-
- g_type_add_interface_static (speexenc_type, GST_TYPE_TAG_SETTER,
- &tag_setter_info);
-
- GST_DEBUG_CATEGORY_INIT (speexenc_debug, "speexenc", 0, "Speex encoder");
-}
-
-GST_BOILERPLATE_FULL (GstSpeexEnc, gst_speex_enc, GstAudioEncoder,
- GST_TYPE_AUDIO_ENCODER, gst_speex_enc_setup_interfaces);
-
-static void
-gst_speex_enc_base_init (gpointer g_class)
-{
- GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&src_factory));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&sink_factory));
- gst_element_class_set_details_simple (element_class, "Speex audio encoder",
- "Codec/Encoder/Audio",
- "Encodes audio in Speex format", "Wim Taymans <wim@fluendo.com>");
-}
+#define gst_speex_enc_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstSpeexEnc, gst_speex_enc, GST_TYPE_AUDIO_ENCODER,
+ G_IMPLEMENT_INTERFACE (GST_TYPE_TAG_SETTER, NULL);
+ G_IMPLEMENT_INTERFACE (GST_TYPE_PRESET, NULL));
static void
gst_speex_enc_class_init (GstSpeexEncClass * klass)
{
GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
GstAudioEncoderClass *base_class;
gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
base_class = (GstAudioEncoderClass *) klass;
+ gobject_class->finalize = gst_speex_enc_finalize;
gobject_class->set_property = gst_speex_enc_set_property;
gobject_class->get_property = gst_speex_enc_get_property;
"The last status message", NULL,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
- gobject_class->finalize = gst_speex_enc_finalize;
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&src_factory));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&sink_factory));
+ gst_element_class_set_details_simple (gstelement_class, "Speex audio encoder",
+ "Codec/Encoder/Audio",
+ "Encodes audio in Speex format", "Wim Taymans <wim@fluendo.com>");
+
+ GST_DEBUG_CATEGORY_INIT (speexenc_debug, "speexenc", 0, "Speex encoder");
}
static void
}
static void
-gst_speex_enc_init (GstSpeexEnc * enc, GstSpeexEncClass * klass)
+gst_speex_enc_init (GstSpeexEnc * enc)
{
GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
GST_DEBUG_OBJECT (enc, "start");
speex_bits_init (&enc->bits);
- enc->tags = gst_tag_list_new ();
+ enc->tags = gst_tag_list_new_empty ();
enc->header_sent = FALSE;
return TRUE;
g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL);
enc->headers = NULL;
+ gst_tag_setter_reset_tags (GST_TAG_SETTER (enc));
+
return TRUE;
}
gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (enc)));
if (merged_tags == NULL)
- merged_tags = gst_tag_list_new ();
+ merged_tags = gst_tag_list_new_empty ();
GST_DEBUG_OBJECT (enc, "merged tags = %" GST_PTR_FORMAT, merged_tags);
comments = gst_tag_list_to_vorbiscomment_buffer (merged_tags, NULL,
static GstFlowReturn
gst_speex_enc_push_buffer (GstSpeexEnc * enc, GstBuffer * buffer)
{
- guint size;
+ GST_DEBUG_OBJECT (enc, "pushing output buffer of size %u",
+ gst_buffer_get_size (buffer));
- size = GST_BUFFER_SIZE (buffer);
- GST_DEBUG_OBJECT (enc, "pushing output buffer of size %u", size);
-
- gst_buffer_set_caps (buffer, GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)));
return gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (enc), buffer);
}
gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf)
{
gint frame_size = enc->frame_size;
- gint bytes = frame_size * 2 * enc->channels, samples, size;
+ gint bytes = frame_size * 2 * enc->channels, samples;
gint outsize, written, dtx_ret = 0;
- guint8 *data;
+ guint8 *data, *data0 = NULL, *bdata, *outdata;
+ gsize bsize, size;
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
if (G_LIKELY (buf)) {
- data = GST_BUFFER_DATA (buf);
- size = GST_BUFFER_SIZE (buf);
+ bdata = gst_buffer_map (buf, &bsize, NULL, GST_MAP_READ);
- if (G_UNLIKELY (size % bytes)) {
+ if (G_UNLIKELY (bsize % bytes)) {
GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
- size = ((size / bytes) + 1) * bytes;
- data = g_malloc0 (size);
- memcpy (data, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
+
+ size = ((bsize / bytes) + 1) * bytes;
+ data0 = data = g_malloc0 (size);
+ memcpy (data, bdata, bsize);
+ gst_buffer_unmap (buf, bdata, bsize);
+ bdata = NULL;
+ } else {
+ data = bdata;
+ size = bsize;
}
} else {
GST_DEBUG_OBJECT (enc, "nothing to drain");
speex_bits_insert_terminator (&enc->bits);
outsize = speex_bits_nbytes (&enc->bits);
+ if (bdata)
+ gst_buffer_unmap (buf, bdata, bsize);
+
+#if 0
ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc),
GST_BUFFER_OFFSET_NONE, outsize,
GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf);
if ((GST_FLOW_OK != ret))
goto done;
+#endif
+ outbuf = gst_buffer_new_allocate (NULL, outsize, 0);
+ outdata = gst_buffer_map (outbuf, NULL, NULL, GST_MAP_WRITE);
- written = speex_bits_write (&enc->bits,
- (gchar *) GST_BUFFER_DATA (outbuf), outsize);
+ written = speex_bits_write (&enc->bits, (gchar *) outdata, outsize);
if (G_UNLIKELY (written < outsize)) {
GST_ERROR_OBJECT (enc, "short write: %d < %d bytes", written, outsize);
- GST_BUFFER_SIZE (outbuf) = written;
} else if (G_UNLIKELY (written > outsize)) {
GST_ERROR_OBJECT (enc, "overrun: %d > %d bytes", written, outsize);
+ written = outsize;
}
+ gst_buffer_unmap (outbuf, outdata, written);
if (!dtx_ret)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
outbuf, samples);
done:
+ g_free (data0);
return ret;
}
va_start (va, buf);
/* put buffers in a fixed list */
while (buf) {
- g_assert (gst_buffer_is_metadata_writable (buf));
+ g_assert (gst_buffer_is_writable (buf));
/* mark buffer */
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
/* create header buffer */
data = (guint8 *) speex_header_to_packet (&enc->header, &data_len);
- buf1 = gst_buffer_new ();
- GST_BUFFER_DATA (buf1) = GST_BUFFER_MALLOCDATA (buf1) = data;
- GST_BUFFER_SIZE (buf1) = data_len;
+ buf1 = gst_buffer_new_wrapped (data, data_len);
GST_BUFFER_OFFSET_END (buf1) = 0;
GST_BUFFER_OFFSET (buf1) = 0;
/* negotiate with these caps */
GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps);
- gst_buffer_set_caps (buf1, caps);
- gst_buffer_set_caps (buf2, caps);
gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps);
gst_caps_unref (caps);
}
GST_DEBUG_OBJECT (enc, "received buffer %p of %u bytes", buf,
- buf ? GST_BUFFER_SIZE (buf) : 0);
+ buf ? gst_buffer_get_size (buf) : 0);
ret = gst_speex_enc_encode (enc, buf);
projects / platform / upstream / gstreamer.git / blobdiff