"L.A.M.E. mp3 encoder",
"Codec/Encoder/Audio",
"High-quality free MP3 encoder",
- "Erik Walthinsen <omega@cse.ogi.edu>",
+ "Erik Walthinsen <omega@cse.ogi.edu>, " "Wim Taymans <wim@fluendo.com>",
};
/* LAME can do MPEG-1, MPEG-2, and MPEG-2.5, so it has 9 possible
const GValue * value, GParamSpec * pspec);
static void gst_lame_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
-static void gst_lame_chain (GstPad * pad, GstData * _data);
+static gboolean gst_lame_sink_event (GstPad * pad, GstEvent * event);
+static GstFlowReturn gst_lame_chain (GstPad * pad, GstBuffer * buf);
static gboolean gst_lame_setup (GstLame * lame);
static GstElementStateReturn gst_lame_change_state (GstElement * element);
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+ gobject_class->set_property = gst_lame_set_property;
+ gobject_class->get_property = gst_lame_get_property;
+
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BITRATE,
g_param_spec_int ("bitrate", "Bitrate (kb/s)", "Bitrate in kbit/sec",
8, 320, 128, G_PARAM_READWRITE));
g_param_spec_enum ("preset", "Lame Preset", "Lame Preset",
GST_TYPE_LAME_PRESET, 0, G_PARAM_READWRITE));
#endif
- gobject_class->set_property = gst_lame_set_property;
- gobject_class->get_property = gst_lame_get_property;
gstelement_class->change_state = gst_lame_change_state;
}
if (!gst_lame_setup (lame)) {
GST_DEBUG_OBJECT (lame, "problem doing lame setup");
- return
+ caps =
gst_caps_copy (gst_pad_template_get_caps (gst_static_pad_template_get
(&gst_lame_src_template)));
+ } else {
+ caps = gst_caps_new_simple ("audio/mpeg",
+ "mpegversion", G_TYPE_INT, 1,
+ "layer", G_TYPE_INT, 3,
+ "rate", G_TYPE_INT, lame_get_out_samplerate (lame->lgf),
+ "channels", G_TYPE_INT, lame->num_channels, NULL);
}
- caps = gst_caps_new_simple ("audio/mpeg",
- "mpegversion", G_TYPE_INT, 1,
- "layer", G_TYPE_INT, 3,
- "rate", G_TYPE_INT, lame_get_out_samplerate (lame->lgf),
- "channels", G_TYPE_INT, lame->num_channels, NULL);
+ gst_object_unref (lame);
return caps;
}
-static GstPadLinkReturn
-gst_lame_src_link (GstPad * pad, const GstCaps * caps)
+#if 0
+static gboolean
+gst_lame_src_setcaps (GstPad * pad, GstCaps * caps)
{
GstLame *lame;
gint out_samplerate;
GstCaps *othercaps, *channelcaps;
GstPadLinkReturn result;
- lame = GST_LAME (gst_pad_get_parent (pad));
+ lame = GST_LAME (GST_PAD_PARENT (pad));
structure = gst_caps_get_structure (caps, 0);
- if (!gst_structure_get_int (structure, "rate", &out_samplerate) ||
- !gst_structure_get_int (structure, "channels", &lame->num_channels))
- g_return_val_if_reached (GST_PAD_LINK_REFUSED);
+ /* we need channels and rate */
+ if (!gst_structure_get_int (structure, "rate", &out_samplerate));
+ goto no_rate;
+ if (!gst_structure_get_int (structure, "channels", &lame->num_channels));
+ goto no_channels;
if (lame_set_out_samplerate (lame->lgf, out_samplerate) != 0)
- return GST_PAD_LINK_REFUSED;
+ goto could_not_set_samplerate;
/* we don't do channel conversion */
channelcaps = gst_caps_new_simple ("audio/x-raw-int", "channels", G_TYPE_INT,
}
return result;
+
+no_rate:
+ {
+ GST_DEBUG ("no rate specified in caps");
+ return FALSE;
+ }
+no_channels:
+ {
+ GST_DEBUG ("no channels specified in caps");
+ return FALSE;
+ }
+could_not_set_samplerate:
+ {
+ GST_DEBUG ("could not set samplerate");
+ return FALSE;
+ }
}
+#endif
-static GstPadLinkReturn
-gst_lame_sink_link (GstPad * pad, const GstCaps * caps)
+static gboolean
+gst_lame_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstLame *lame;
gint out_samplerate;
lame = GST_LAME (gst_pad_get_parent (pad));
structure = gst_caps_get_structure (caps, 0);
- if (!gst_structure_get_int (structure, "rate", &lame->samplerate) ||
- !gst_structure_get_int (structure, "channels", &lame->num_channels))
- g_return_val_if_reached (GST_PAD_LINK_REFUSED);
+ if (!gst_structure_get_int (structure, "rate", &lame->samplerate))
+ goto no_rate;
+ if (!gst_structure_get_int (structure, "channels", &lame->num_channels))
+ goto no_channels;
+ /* let lame choose a default samplerate */
lame_set_out_samplerate (lame->lgf, 0);
- if (!gst_lame_setup (lame)) {
- GST_ELEMENT_ERROR (lame, CORE, NEGOTIATION, (NULL),
- ("could not initialize encoder (wrong parameters?)"));
- return GST_PAD_LINK_REFUSED;
- }
+ if (!gst_lame_setup (lame))
+ goto setup_failed;
out_samplerate = lame_get_out_samplerate (lame->lgf);
othercaps =
"channels", G_TYPE_INT, lame->num_channels,
"rate", G_TYPE_INT, out_samplerate, NULL);
- return gst_pad_try_set_caps (lame->srcpad, othercaps);
+ /* and use these caps */
+ gst_pad_set_caps (lame->srcpad, othercaps);
+ gst_caps_unref (othercaps);
+
+ return TRUE;
+
+no_rate:
+ {
+ GST_ELEMENT_ERROR (lame, CORE, NEGOTIATION, (NULL),
+ ("no rate specified in input"));
+ return FALSE;
+ }
+no_channels:
+ {
+ GST_ELEMENT_ERROR (lame, CORE, NEGOTIATION, (NULL),
+ ("no channels specified in input"));
+ return FALSE;
+ }
+setup_failed:
+ {
+ GST_ELEMENT_ERROR (lame, CORE, NEGOTIATION, (NULL),
+ ("could not initialize encoder (wrong parameters?)"));
+ return FALSE;
+ }
}
static void
lame->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_lame_sink_template), "sink");
- gst_element_add_pad (GST_ELEMENT (lame), lame->sinkpad);
+ gst_pad_set_event_function (lame->sinkpad, gst_lame_sink_event);
gst_pad_set_chain_function (lame->sinkpad, gst_lame_chain);
- gst_pad_set_link_function (lame->sinkpad, gst_lame_sink_link);
+ gst_pad_set_setcaps_function (lame->sinkpad, gst_lame_sink_setcaps);
+ gst_element_add_pad (GST_ELEMENT (lame), lame->sinkpad);
lame->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_lame_src_template), "src");
- gst_element_add_pad (GST_ELEMENT (lame), lame->srcpad);
- gst_pad_set_link_function (lame->srcpad, gst_lame_src_link);
gst_pad_set_getcaps_function (lame->srcpad, gst_lame_src_getcaps);
- GST_FLAG_SET (lame, GST_ELEMENT_EVENT_AWARE);
+ gst_element_add_pad (GST_ELEMENT (lame), lame->srcpad);
GST_DEBUG ("setting up lame encoder");
lame->lgf = lame_init ();
id3tag_init (lame->lgf);
- lame->newmediacount = 0;
GST_DEBUG_OBJECT (lame, "done initializing");
}
}
}
-static void
-gst_lame_chain (GstPad * pad, GstData * _data)
+static gboolean
+gst_lame_sink_event (GstPad * pad, GstEvent * event)
+{
+ GstLame *lame;
+
+ lame = GST_LAME (gst_pad_get_parent (pad));
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_EOS:
+ GST_DEBUG_OBJECT (lame, "handling EOS event");
+ /* FIXME, push last data packet */
+
+ gst_pad_push_event (lame->srcpad, event);
+ break;
+ case GST_EVENT_FLUSH_START:
+ GST_DEBUG_OBJECT (lame, "handling FLUSH start event");
+ /* forward event */
+ gst_pad_push_event (lame->srcpad, event);
+
+ /* make streaming stop */
+ GST_STREAM_LOCK (pad);
+ GST_STREAM_UNLOCK (pad);
+ break;
+ case GST_EVENT_FLUSH_STOP:
+ {
+ guchar *mp3_data = NULL;
+ gint mp3_buffer_size, mp3_size = 0;
+
+ GST_DEBUG_OBJECT (lame, "handling FLUSH stop event");
+
+ /* clear buffers */
+ GST_STREAM_LOCK (pad);
+ mp3_buffer_size = 7200;
+ mp3_data = g_malloc (mp3_buffer_size);
+ mp3_size = lame_encode_flush (lame->lgf, mp3_data, mp3_buffer_size);
+
+ gst_pad_push_event (lame->srcpad, event);
+ GST_STREAM_UNLOCK (pad);
+ break;
+ }
+ case GST_EVENT_TAG:
+ GST_DEBUG_OBJECT (lame, "handling TAG event");
+ if (lame->tags) {
+ GstTagList *taglist;
+
+ gst_event_parse_tag (event, &taglist),
+ gst_tag_list_insert (lame->tags, taglist,
+ gst_tag_setter_get_merge_mode (GST_TAG_SETTER (lame)));
+ } else {
+ g_assert_not_reached ();
+ }
+ gst_pad_push_event (lame->srcpad, event);
+ break;
+ default:
+ gst_pad_push_event (lame->srcpad, event);
+ break;
+ }
+ return TRUE;
+}
+
+static GstFlowReturn
+gst_lame_chain (GstPad * pad, GstBuffer * buf)
{
- GstBuffer *buf = GST_BUFFER (_data);
GstLame *lame;
GstBuffer *outbuf;
- gchar *mp3_data = NULL;
+ guchar *mp3_data = NULL;
gint mp3_buffer_size, mp3_size = 0;
- gboolean eos = FALSE;
+ gint64 duration;
+ GstFlowReturn result;
lame = GST_LAME (gst_pad_get_parent (pad));
GST_LOG_OBJECT (lame, "entered chain");
- if (GST_IS_EVENT (buf)) {
- switch (GST_EVENT_TYPE (buf)) {
- case GST_EVENT_EOS:
- GST_DEBUG_OBJECT (lame, "handling EOS event");
- eos = TRUE;
- case GST_EVENT_FLUSH:
- GST_DEBUG_OBJECT (lame, "handling FLUSH event");
- mp3_buffer_size = 7200;
- mp3_data = g_malloc (mp3_buffer_size);
-
- mp3_size = lame_encode_flush (lame->lgf, mp3_data, mp3_buffer_size);
- gst_event_unref (GST_EVENT (buf));
- break;
- case GST_EVENT_TAG:
- GST_DEBUG_OBJECT (lame, "handling TAG event");
- if (lame->tags) {
- gst_tag_list_insert (lame->tags,
- gst_event_tag_get_list (GST_EVENT (buf)),
- gst_tag_setter_get_merge_mode (GST_TAG_SETTER (lame)));
- } else {
- g_assert_not_reached ();
- }
-
- gst_pad_event_default (pad, GST_EVENT (buf));
- break;
- case GST_EVENT_DISCONTINUOUS:
- if (GST_EVENT_DISCONT_NEW_MEDIA (GST_EVENT (buf))) {
- /* do not re-initialise if it is first new media discont */
- if (lame->newmediacount++ > 0) {
- lame_close (lame->lgf);
- lame->lgf = lame_init ();
- lame->initialized = FALSE;
- lame->last_ts = GST_CLOCK_TIME_NONE;
-
- gst_lame_setup (lame);
- }
- }
- gst_pad_event_default (pad, GST_EVENT (buf));
-
- break;
- default:
- gst_pad_event_default (pad, GST_EVENT (buf));
- break;
- }
+ if (!lame->initialized)
+ goto not_initialized;
+
+ /* allocate space for output */
+ mp3_buffer_size =
+ ((GST_BUFFER_SIZE (buf) / (2 + lame->num_channels)) * 1.25) + 7200;
+ mp3_data = g_malloc (mp3_buffer_size);
+
+ /* lame seems to be too stupid to get mono interleaved going */
+ if (lame->num_channels == 1) {
+ mp3_size = lame_encode_buffer (lame->lgf,
+ (short int *) (GST_BUFFER_DATA (buf)),
+ (short int *) (GST_BUFFER_DATA (buf)),
+ GST_BUFFER_SIZE (buf) / 2, mp3_data, mp3_buffer_size);
} else {
- gint64 duration;
+ mp3_size = lame_encode_buffer_interleaved (lame->lgf,
+ (short int *) (GST_BUFFER_DATA (buf)),
+ GST_BUFFER_SIZE (buf) / 2 / lame->num_channels,
+ mp3_data, mp3_buffer_size);
+ }
- if (!lame->initialized) {
- gst_buffer_unref (buf);
- GST_ELEMENT_ERROR (lame, CORE, NEGOTIATION, (NULL),
- ("encoder not initialized (input is not audio?)"));
- return;
- }
+ GST_LOG_OBJECT (lame, "encoded %d bytes of audio to %d bytes of mp3",
+ GST_BUFFER_SIZE (buf), mp3_size);
- /* allocate space for output */
- mp3_buffer_size =
- ((GST_BUFFER_SIZE (buf) / (2 + lame->num_channels)) * 1.25) + 7200;
- mp3_data = g_malloc (mp3_buffer_size);
-
- /* lame seems to be too stupid to get mono interleaved going */
- if (lame->num_channels == 1) {
- mp3_size = lame_encode_buffer (lame->lgf,
- (short int *) (GST_BUFFER_DATA (buf)),
- (short int *) (GST_BUFFER_DATA (buf)),
- GST_BUFFER_SIZE (buf) / 2, mp3_data, mp3_buffer_size);
- } else {
- mp3_size = lame_encode_buffer_interleaved (lame->lgf,
- (short int *) (GST_BUFFER_DATA (buf)),
- GST_BUFFER_SIZE (buf) / 2 / lame->num_channels,
- mp3_data, mp3_buffer_size);
- }
+ duration = (GST_SECOND * GST_BUFFER_SIZE (buf) /
+ (2 * lame->samplerate * lame->num_channels));
- GST_LOG_OBJECT (lame, "encoded %d bytes of audio to %d bytes of mp3",
- GST_BUFFER_SIZE (buf), mp3_size);
-
- duration = (GST_SECOND * GST_BUFFER_SIZE (buf) /
- (2 * lame->samplerate * lame->num_channels));
-
- if (GST_BUFFER_DURATION (buf) != GST_CLOCK_TIME_NONE &&
- GST_BUFFER_DURATION (buf) != duration)
- GST_DEBUG_OBJECT (lame, "incoming buffer had incorrect duration "
- GST_TIME_FORMAT "outgoing buffer will have correct duration "
- GST_TIME_FORMAT,
- GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_TIME_ARGS (duration));
-
- if (lame->last_ts == GST_CLOCK_TIME_NONE) {
- lame->last_ts = GST_BUFFER_TIMESTAMP (buf);
- lame->last_offs = GST_BUFFER_OFFSET (buf);
- lame->last_duration = duration;
- } else {
- lame->last_duration += duration;
- }
+ if (GST_BUFFER_DURATION (buf) != GST_CLOCK_TIME_NONE &&
+ GST_BUFFER_DURATION (buf) != duration)
+ GST_DEBUG_OBJECT (lame, "incoming buffer had incorrect duration %"
+ GST_TIME_FORMAT "outgoing buffer will have correct duration %"
+ GST_TIME_FORMAT,
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_TIME_ARGS (duration));
- gst_buffer_unref (buf);
+ if (lame->last_ts == GST_CLOCK_TIME_NONE) {
+ lame->last_ts = GST_BUFFER_TIMESTAMP (buf);
+ lame->last_offs = GST_BUFFER_OFFSET (buf);
+ lame->last_duration = duration;
+ } else {
+ lame->last_duration += duration;
}
+ gst_buffer_unref (buf);
+
if (mp3_size > 0) {
outbuf = gst_buffer_new ();
GST_BUFFER_DATA (outbuf) = mp3_data;
GST_BUFFER_OFFSET (outbuf) = lame->last_offs;
GST_BUFFER_DURATION (outbuf) = lame->last_duration;
- gst_pad_push (lame->srcpad, GST_DATA (outbuf));
+ result = gst_pad_push (lame->srcpad, outbuf);
lame->last_ts = GST_CLOCK_TIME_NONE;
} else {
g_free (mp3_data);
+ result = GST_FLOW_OK;
}
+ gst_object_unref (lame);
- if (eos) {
- gst_pad_push (lame->srcpad, GST_DATA (gst_event_new (GST_EVENT_EOS)));
- gst_element_set_eos (GST_ELEMENT (lame));
+ return result;
+
+ /* ERRORS */
+not_initialized:
+ {
+ gst_buffer_unref (buf);
+ GST_ELEMENT_ERROR (lame, CORE, NEGOTIATION, (NULL),
+ ("encoder not initialized (input is not audio?)"));
+ gst_object_unref (lame);
+ return GST_FLOW_ERROR;
}
}
gst_lame_change_state (GstElement * element)
{
GstLame *lame;
-
- g_return_val_if_fail (GST_IS_LAME (element), GST_STATE_FAILURE);
+ gint transition;
+ GstElementStateReturn result;
lame = GST_LAME (element);
- GST_DEBUG ("state pending %d", GST_STATE_PENDING (element));
+ transition = GST_STATE_TRANSITION (lame);
- switch (GST_STATE_TRANSITION (element)) {
+ switch (transition) {
case GST_STATE_READY_TO_PAUSED:
lame->last_ts = GST_CLOCK_TIME_NONE;
break;
- case GST_STATE_READY_TO_NULL:
+ default:
+ break;
+ }
+
+ /* if we haven't failed already, give the parent class a chance to ;-) */
+ result = GST_ELEMENT_CLASS (parent_class)->change_state (element);
+
+ switch (transition) {
+ case GST_STATE_PAUSED_TO_READY:
if (lame->initialized) {
lame_close (lame->lgf);
lame->lgf = lame_init ();
break;
}
- /* if we haven't failed already, give the parent class a chance to ;-) */
- if (GST_ELEMENT_CLASS (parent_class)->change_state)
- return GST_ELEMENT_CLASS (parent_class)->change_state (element);
-
- return GST_STATE_SUCCESS;
+ return result;
}
static gboolean
projects / platform / upstream / gst-plugins-good.git / commitdiff