*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
/**
* <refsect2>
* <title>Example pipelines</title>
* |[
- * gst-launch audiotestsrc num-buffers=100 ! speexenc ! oggmux ! filesink location=beep.ogg
+ * gst-launch-1.0 audiotestsrc num-buffers=100 ! speexenc ! oggmux ! filesink location=beep.ogg
* ]| Encode an Ogg/Speex file.
* </refsect2>
*/
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, "
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " FORMAT_STR ", "
+ "layout = (string) interleaved, "
+ "rate = (int) [ 6000, 48000 ], "
+ "channels = (int) 1; "
+ "audio/x-raw, "
+ "format = (string) " FORMAT_STR ", "
+ "layout = (string) interleaved, "
"rate = (int) [ 6000, 48000 ], "
- "channels = (int) [ 1, 2 ], "
- "endianness = (int) BYTE_ORDER, "
- "signed = (boolean) TRUE, " "width = (int) 16, " "depth = (int) 16")
+ "channels = (int) 2, " "channel-mask = (bitmask) 0x3")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GstBuffer * in_buf);
static gboolean gst_speex_enc_sink_event (GstAudioEncoder * enc,
GstEvent * event);
-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;
base_class->stop = GST_DEBUG_FUNCPTR (gst_speex_enc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_speex_enc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_speex_enc_handle_frame);
- base_class->event = GST_DEBUG_FUNCPTR (gst_speex_enc_sink_event);
- base_class->pre_push = GST_DEBUG_FUNCPTR (gst_speex_enc_pre_push);
+ base_class->sink_event = GST_DEBUG_FUNCPTR (gst_speex_enc_sink_event);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_QUALITY,
g_param_spec_float ("quality", "Quality", "Encoding quality",
"The last status message", NULL,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
- gobject_class->finalize = gst_speex_enc_finalize;
+ gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
+ gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);
+ gst_element_class_set_static_metadata (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);
/* arrange granulepos marking (and required perfect ts) */
gst_audio_encoder_set_mark_granule (benc, TRUE);
gst_audio_encoder_set_perfect_timestamp (benc, TRUE);
+ GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));
}
static gboolean
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;
+ enc->encoded_samples = 0;
return TRUE;
}
enc->state = NULL;
}
speex_bits_destroy (&enc->bits);
- gst_tag_list_free (enc->tags);
+ speex_bits_set_bit_buffer (&enc->bits, NULL, 0);
+ gst_tag_list_unref (enc->tags);
enc->tags = NULL;
- 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,
0, "Encoded with GStreamer Speexenc");
- gst_tag_list_free (merged_tags);
+ gst_tag_list_unref (merged_tags);
GST_BUFFER_OFFSET (comments) = 0;
GST_BUFFER_OFFSET_END (comments) = 0;
return TRUE;
}
-/* push out the buffer */
-static GstFlowReturn
-gst_speex_enc_push_buffer (GstSpeexEnc * enc, GstBuffer * buffer)
-{
- guint size;
-
- 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);
-}
-
static gboolean
gst_speex_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
{
}
break;
}
+ case GST_EVENT_SEGMENT:
+ enc->encoded_samples = 0;
+ break;
default:
break;
}
/* we only peeked, let base class handle it */
- return FALSE;
+ return GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (benc, event);
}
static GstFlowReturn
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;
+ GstMapInfo map;
+ guint8 *data, *data0 = NULL, *bdata;
+ gsize bsize, size;
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
+ GstSegment *segment;
+ GstClockTime duration;
if (G_LIKELY (buf)) {
- data = GST_BUFFER_DATA (buf);
- size = GST_BUFFER_SIZE (buf);
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ bdata = map.data;
+ bsize = map.size;
- 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));
+
+ /* If encoding part of a frame, and we have no set stop time on
+ * the output segment, we update the segment stop time to reflect
+ * the last sample. This will let oggmux set the last page's
+ * granpos to tell a decoder the dummy samples should be clipped.
+ */
+ segment = &GST_AUDIO_ENCODER_OUTPUT_SEGMENT (enc);
+ GST_DEBUG_OBJECT (enc, "existing output segment %" GST_SEGMENT_FORMAT,
+ segment);
+ if (!GST_CLOCK_TIME_IS_VALID (segment->stop)) {
+ int input_samples = bsize / (enc->channels * 2);
+ GST_DEBUG_OBJECT (enc,
+ "No stop time and partial frame, updating segment");
+ duration =
+ gst_util_uint64_scale (enc->encoded_samples + input_samples,
+ GST_SECOND, enc->rate);
+ segment->stop = segment->start + duration;
+ GST_DEBUG_OBJECT (enc, "new output segment %" GST_SEGMENT_FORMAT,
+ segment);
+ gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
+ gst_event_new_segment (segment));
+ }
+
+ size = ((bsize / bytes) + 1) * bytes;
+ data0 = data = g_malloc0 (size);
+ memcpy (data, bdata, bsize);
+ gst_buffer_unmap (buf, &map);
+ 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, &map);
+
+#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, NULL);
+ gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
- written = speex_bits_write (&enc->bits,
- (gchar *) GST_BUFFER_DATA (outbuf), outsize);
+ written = speex_bits_write (&enc->bits, (gchar *) map.data, 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, &map);
+ gst_buffer_resize (outbuf, 0, written);
if (!dtx_ret)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc),
outbuf, samples);
+ enc->encoded_samples += frame_size;
done:
+ g_free (data0);
return ret;
}
/*
* (really really) FIXME: move into core (dixit tpm)
*/
-/**
+/*
* _gst_caps_set_buffer_array:
- * @caps: a #GstCaps
+ * @caps: (transfer full): a #GstCaps
* @field: field in caps to set
* @buf: header buffers
*
* Adds given buffers to an array of buffers set as the given @field
* on the given @caps. List of buffer arguments must be NULL-terminated.
*
- * Returns: input caps with a streamheader field added, or NULL if some error
+ * Returns: (transfer full): input caps with a streamheader field added, or NULL
+ * if some error occurred
*/
static GstCaps *
_gst_caps_set_buffer_array (GstCaps * caps, const gchar * field,
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);
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
g_value_init (&value, GST_TYPE_BUFFER);
buf = gst_buffer_copy (buf);
- GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
gst_value_set_buffer (&value, buf);
gst_buffer_unref (buf);
gst_value_array_append_value (&array, &value);
buf = va_arg (va, GstBuffer *);
}
+ va_end (va);
gst_structure_set_value (structure, field, &array);
g_value_unset (&array);
GstCaps *caps;
guchar *data;
gint data_len;
+ GList *headers;
/* 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_audio_encoder_set_output_format (GST_AUDIO_ENCODER (enc), caps);
gst_caps_unref (caps);
/* push out buffers */
/* store buffers for later pre_push sending */
- g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL);
- enc->headers = NULL;
+ headers = NULL;
GST_DEBUG_OBJECT (enc, "storing header buffers");
- enc->headers = g_slist_prepend (enc->headers, buf2);
- enc->headers = g_slist_prepend (enc->headers, buf1);
+ headers = g_list_prepend (headers, buf2);
+ headers = g_list_prepend (headers, buf1);
+ gst_audio_encoder_set_headers (benc, headers);
enc->header_sent = TRUE;
}
- GST_DEBUG_OBJECT (enc, "received buffer %p of %u bytes", buf,
- buf ? GST_BUFFER_SIZE (buf) : 0);
+ GST_DEBUG_OBJECT (enc, "received buffer %p of %" G_GSIZE_FORMAT " bytes", buf,
+ buf ? gst_buffer_get_size (buf) : 0);
ret = gst_speex_enc_encode (enc, buf);
return ret;
}
-static GstFlowReturn
-gst_speex_enc_pre_push (GstAudioEncoder * benc, GstBuffer ** buffer)
-{
- GstSpeexEnc *enc;
- GstFlowReturn ret = GST_FLOW_OK;
-
- enc = GST_SPEEX_ENC (benc);
-
- /* FIXME 0.11 ? get rid of this special ogg stuff and have it
- * put and use 'codec data' in caps like anything else,
- * with all the usual out-of-band advantage etc */
- if (G_UNLIKELY (enc->headers)) {
- GSList *header = enc->headers;
-
- /* try to push all of these, if we lose one, might as well lose all */
- while (header) {
- if (ret == GST_FLOW_OK)
- ret = gst_speex_enc_push_buffer (enc, header->data);
- else
- gst_speex_enc_push_buffer (enc, header->data);
- header = g_slist_next (header);
- }
-
- g_slist_free (enc->headers);
- enc->headers = NULL;
- }
-
- return ret;
-}
-
static void
gst_speex_enc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
projects / platform / upstream / gst-plugins-good.git / blobdiff