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-rw-r--r--ext/speex/Makefile.am5
-rw-r--r--ext/speex/gstspeexdec.c582
-rw-r--r--ext/speex/gstspeexdec.h12
-rw-r--r--ext/speex/gstspeexenc.c859
-rw-r--r--ext/speex/gstspeexenc.h37
5 files changed, 328 insertions, 1167 deletions
diff --git a/ext/speex/Makefile.am b/ext/speex/Makefile.am
index f2efb3614..fe5523796 100644
--- a/ext/speex/Makefile.am
+++ b/ext/speex/Makefile.am
@@ -1,13 +1,14 @@
plugin_LTLIBRARIES = libgstspeex.la
libgstspeex_la_SOURCES = gstspeex.c gstspeexdec.c gstspeexenc.c
-libgstspeex_la_CFLAGS = \
+libgstspeex_la_CFLAGS = -DGST_USE_UNSTABLE_API \
$(GST_PLUGINS_BASE_CFLAGS) \
$(GST_BASE_CFLAGS) \
$(GST_CFLAGS) \
$(SPEEX_CFLAGS)
libgstspeex_la_LIBADD = \
- $(GST_PLUGINS_BASE_LIBS) -lgsttag-$(GST_MAJORMINOR) \
+ $(GST_PLUGINS_BASE_LIBS) \
+ -lgsttag-$(GST_MAJORMINOR) -lgstaudio-$(GST_MAJORMINOR) \
$(GST_BASE_LIBS) \
$(GST_LIBS) \
$(SPEEX_LIBS)
diff --git a/ext/speex/gstspeexdec.c b/ext/speex/gstspeexdec.c
index c5add8268..de4121436 100644
--- a/ext/speex/gstspeexdec.c
+++ b/ext/speex/gstspeexdec.c
@@ -78,53 +78,43 @@ GST_STATIC_PAD_TEMPLATE ("sink",
);
#define gst_speex_dec_parent_class parent_class
-G_DEFINE_TYPE (GstSpeexDec, gst_speex_dec, GST_TYPE_ELEMENT);
-
-static gboolean speex_dec_sink_event (GstPad * pad, GstEvent * event);
-static GstFlowReturn speex_dec_chain (GstPad * pad, GstBuffer * buf);
-static GstStateChangeReturn speex_dec_change_state (GstElement * element,
- GstStateChange transition);
-
-static gboolean speex_dec_src_event (GstPad * pad, GstEvent * event);
-static gboolean speex_dec_src_query (GstPad * pad, GstQuery * query);
-static gboolean speex_dec_sink_query (GstPad * pad, GstQuery * query);
-static const GstQueryType *speex_get_src_query_types (GstPad * pad);
-static const GstQueryType *speex_get_sink_query_types (GstPad * pad);
-static gboolean speex_dec_convert (GstPad * pad,
- GstFormat src_format, gint64 src_value,
- GstFormat * dest_format, gint64 * dest_value);
+G_DEFINE_TYPE (GstSpeexDec, gst_speex_dec, GST_TYPE_AUDIO_DECODER);
+
+static gboolean gst_speex_dec_start (GstAudioDecoder * dec);
+static gboolean gst_speex_dec_stop (GstAudioDecoder * dec);
+static gboolean gst_speex_dec_set_format (GstAudioDecoder * bdec,
+ GstCaps * caps);
+static GstFlowReturn gst_speex_dec_handle_frame (GstAudioDecoder * dec,
+ GstBuffer * buffer);
static void gst_speex_dec_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_speex_dec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
-static GstFlowReturn speex_dec_chain_parse_data (GstSpeexDec * dec,
- GstBuffer * buf, GstClockTime timestamp, GstClockTime duration);
-
-static GstFlowReturn speex_dec_chain_parse_header (GstSpeexDec * dec,
- GstBuffer * buf);
-static GstFlowReturn speex_dec_chain_parse_comments (GstSpeexDec * dec,
- GstBuffer * buf);
-
static void
gst_speex_dec_class_init (GstSpeexDecClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
+ GstAudioDecoderClass *base_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
+ base_class = (GstAudioDecoderClass *) klass;
gobject_class->set_property = gst_speex_dec_set_property;
gobject_class->get_property = gst_speex_dec_get_property;
+ base_class->start = GST_DEBUG_FUNCPTR (gst_speex_dec_start);
+ base_class->stop = GST_DEBUG_FUNCPTR (gst_speex_dec_stop);
+ base_class->set_format = GST_DEBUG_FUNCPTR (gst_speex_dec_set_format);
+ base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_speex_dec_handle_frame);
+
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_ENH,
g_param_spec_boolean ("enh", "Enh", "Enable perceptual enhancement",
DEFAULT_ENH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- gstelement_class->change_state = GST_DEBUG_FUNCPTR (speex_dec_change_state);
-
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&speex_dec_src_factory));
gst_element_class_add_pad_template (gstelement_class,
@@ -140,7 +130,6 @@ gst_speex_dec_class_init (GstSpeexDecClass * klass)
static void
gst_speex_dec_reset (GstSpeexDec * dec)
{
- gst_segment_init (&dec->segment, GST_FORMAT_UNDEFINED);
dec->packetno = 0;
dec->frame_size = 0;
dec->frame_duration = 0;
@@ -166,396 +155,38 @@ gst_speex_dec_reset (GstSpeexDec * dec)
static void
gst_speex_dec_init (GstSpeexDec * dec)
{
- dec->sinkpad =
- gst_pad_new_from_static_template (&speex_dec_sink_factory, "sink");
- gst_pad_set_chain_function (dec->sinkpad,
- GST_DEBUG_FUNCPTR (speex_dec_chain));
- gst_pad_set_event_function (dec->sinkpad,
- GST_DEBUG_FUNCPTR (speex_dec_sink_event));
- gst_pad_set_query_type_function (dec->sinkpad,
- GST_DEBUG_FUNCPTR (speex_get_sink_query_types));
- gst_pad_set_query_function (dec->sinkpad,
- GST_DEBUG_FUNCPTR (speex_dec_sink_query));
- gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
-
- dec->srcpad =
- gst_pad_new_from_static_template (&speex_dec_src_factory, "src");
- gst_pad_use_fixed_caps (dec->srcpad);
- gst_pad_set_event_function (dec->srcpad,
- GST_DEBUG_FUNCPTR (speex_dec_src_event));
- gst_pad_set_query_type_function (dec->srcpad,
- GST_DEBUG_FUNCPTR (speex_get_src_query_types));
- gst_pad_set_query_function (dec->srcpad,
- GST_DEBUG_FUNCPTR (speex_dec_src_query));
- gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
-
dec->enh = DEFAULT_ENH;
gst_speex_dec_reset (dec);
}
static gboolean
-speex_dec_sink_setcaps (GstPad * pad, GstCaps * caps)
-{
- GstSpeexDec *dec = GST_SPEEX_DEC (gst_pad_get_parent (pad));
- gboolean ret = TRUE;
- GstStructure *s;
- const GValue *streamheader;
-
- s = gst_caps_get_structure (caps, 0);
- if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
- G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
- gst_value_array_get_size (streamheader) >= 2) {
- const GValue *header, *vorbiscomment;
- GstBuffer *buf;
- GstFlowReturn res = GST_FLOW_OK;
-
- header = gst_value_array_get_value (streamheader, 0);
- if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
- buf = gst_value_get_buffer (header);
- res = speex_dec_chain_parse_header (dec, buf);
- if (res != GST_FLOW_OK)
- goto done;
- gst_buffer_replace (&dec->streamheader, buf);
- }
-
- vorbiscomment = gst_value_array_get_value (streamheader, 1);
- if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
- buf = gst_value_get_buffer (vorbiscomment);
- res = speex_dec_chain_parse_comments (dec, buf);
- if (res != GST_FLOW_OK)
- goto done;
- gst_buffer_replace (&dec->vorbiscomment, buf);
- }
- }
-
-done:
- gst_object_unref (dec);
- return ret;
-}
-
-static gboolean
-speex_dec_convert (GstPad * pad,
- GstFormat src_format, gint64 src_value,
- GstFormat * dest_format, gint64 * dest_value)
-{
- gboolean res = TRUE;
- GstSpeexDec *dec;
- guint64 scale = 1;
-
- dec = GST_SPEEX_DEC (gst_pad_get_parent (pad));
-
- if (src_format == *dest_format) {
- *dest_value = src_value;
- res = TRUE;
- goto cleanup;
- }
-
- if (dec->packetno < 1) {
- res = FALSE;
- goto cleanup;
- }
-
- if (pad == dec->sinkpad &&
- (src_format == GST_FORMAT_BYTES || *dest_format == GST_FORMAT_BYTES)) {
- res = FALSE;
- goto cleanup;
- }
-
- switch (src_format) {
- case GST_FORMAT_TIME:
- switch (*dest_format) {
- case GST_FORMAT_BYTES:
- scale = 2 * dec->header->nb_channels;
- case GST_FORMAT_DEFAULT:
- *dest_value =
- gst_util_uint64_scale_int (scale * src_value, dec->header->rate,
- GST_SECOND);
- break;
- default:
- res = FALSE;
- break;
- }
- break;
- case GST_FORMAT_DEFAULT:
- switch (*dest_format) {
- case GST_FORMAT_BYTES:
- *dest_value = src_value * 2 * dec->header->nb_channels;
- break;
- case GST_FORMAT_TIME:
- *dest_value =
- gst_util_uint64_scale_int (src_value, GST_SECOND,
- dec->header->rate);
- break;
- default:
- res = FALSE;
- break;
- }
- break;
- case GST_FORMAT_BYTES:
- switch (*dest_format) {
- case GST_FORMAT_DEFAULT:
- *dest_value = src_value / (2 * dec->header->nb_channels);
- break;
- case GST_FORMAT_TIME:
- *dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND,
- dec->header->rate * 2 * dec->header->nb_channels);
- break;
- default:
- res = FALSE;
- break;
- }
- break;
- default:
- res = FALSE;
- break;
- }
-
-cleanup:
- gst_object_unref (dec);
- return res;
-}
-
-static const GstQueryType *
-speex_get_sink_query_types (GstPad * pad)
-{
- static const GstQueryType speex_dec_sink_query_types[] = {
- GST_QUERY_CONVERT,
- 0
- };
-
- return speex_dec_sink_query_types;
-}
-
-static gboolean
-speex_dec_sink_query (GstPad * pad, GstQuery * query)
-{
- GstSpeexDec *dec;
- gboolean res;
-
- dec = GST_SPEEX_DEC (gst_pad_get_parent (pad));
-
- switch (GST_QUERY_TYPE (query)) {
- case GST_QUERY_CONVERT:
- {
- GstFormat src_fmt, dest_fmt;
- gint64 src_val, dest_val;
-
- gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
- res = speex_dec_convert (pad, src_fmt, src_val, &dest_fmt, &dest_val);
- if (res) {
- gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
- }
- break;
- }
- default:
- res = gst_pad_query_default (pad, query);
- break;
- }
-
- gst_object_unref (dec);
- return res;
-}
-
-static const GstQueryType *
-speex_get_src_query_types (GstPad * pad)
-{
- static const GstQueryType speex_dec_src_query_types[] = {
- GST_QUERY_POSITION,
- GST_QUERY_DURATION,
- 0
- };
-
- return speex_dec_src_query_types;
-}
-
-static gboolean
-speex_dec_src_query (GstPad * pad, GstQuery * query)
-{
- GstSpeexDec *dec;
- gboolean res = FALSE;
-
- dec = GST_SPEEX_DEC (gst_pad_get_parent (pad));
-
- /* FIXME: why not just pass position/duration queries upstream to demuxer? */
- switch (GST_QUERY_TYPE (query)) {
- case GST_QUERY_POSITION:{
- GstSegment segment;
- GstFormat format;
- gint64 cur;
-
- gst_query_parse_position (query, &format, NULL);
-
- GST_PAD_STREAM_LOCK (dec->sinkpad);
- segment = dec->segment;
- GST_PAD_STREAM_UNLOCK (dec->sinkpad);
-
- if (segment.format != GST_FORMAT_TIME) {
- GST_DEBUG_OBJECT (dec, "segment not initialised yet");
- break;
- }
-
- if ((res = speex_dec_convert (dec->srcpad, GST_FORMAT_TIME,
- segment.position, &format, &cur))) {
- gst_query_set_position (query, format, cur);
- }
- break;
- }
- case GST_QUERY_DURATION:{
- GstFormat format;
- gint64 dur;
-
- /* get duration from demuxer */
- if (!gst_pad_query_peer_duration (dec->sinkpad, GST_FORMAT_TIME, &dur))
- break;
-
- gst_query_parse_duration (query, &format, NULL);
-
- /* and convert it into the requested format */
- if ((res = speex_dec_convert (dec->srcpad, GST_FORMAT_TIME,
- dur, &format, &dur))) {
- gst_query_set_duration (query, format, dur);
- }
- break;
- }
- default:
- res = gst_pad_query_default (pad, query);
- break;
- }
-
- gst_object_unref (dec);
- return res;
-}
-
-static gboolean
-speex_dec_src_event (GstPad * pad, GstEvent * event)
+gst_speex_dec_start (GstAudioDecoder * dec)
{
- gboolean res = FALSE;
- GstSpeexDec *dec = GST_SPEEX_DEC (gst_pad_get_parent (pad));
-
- GST_LOG_OBJECT (dec, "handling %s event", GST_EVENT_TYPE_NAME (event));
-
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_SEEK:{
- GstFormat format, tformat;
- gdouble rate;
- GstEvent *real_seek;
- GstSeekFlags flags;
- GstSeekType cur_type, stop_type;
- gint64 cur, stop;
- gint64 tcur, tstop;
-
- gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
- &stop_type, &stop);
-
- /* we have to ask our peer to seek to time here as we know
- * nothing about how to generate a granulepos from the src
- * formats or anything.
- *
- * First bring the requested format to time
- */
- tformat = GST_FORMAT_TIME;
- if (!(res = speex_dec_convert (pad, format, cur, &tformat, &tcur)))
- break;
- if (!(res = speex_dec_convert (pad, format, stop, &tformat, &tstop)))
- break;
+ GstSpeexDec *sd = GST_SPEEX_DEC (dec);
- /* then seek with time on the peer */
- real_seek = gst_event_new_seek (rate, GST_FORMAT_TIME,
- flags, cur_type, tcur, stop_type, tstop);
+ GST_DEBUG_OBJECT (dec, "start");
+ gst_speex_dec_reset (sd);
- GST_LOG_OBJECT (dec, "seek to %" GST_TIME_FORMAT, GST_TIME_ARGS (tcur));
-
- res = gst_pad_push_event (dec->sinkpad, real_seek);
- gst_event_unref (event);
- break;
- }
- default:
- res = gst_pad_event_default (pad, event);
- break;
- }
+ /* we know about concealment */
+ gst_audio_decoder_set_plc_aware (dec, TRUE);
- gst_object_unref (dec);
- return res;
+ return TRUE;
}
static gboolean
-speex_dec_sink_event (GstPad * pad, GstEvent * event)
+gst_speex_dec_stop (GstAudioDecoder * dec)
{
- GstSpeexDec *dec;
- gboolean ret = FALSE;
-
- dec = GST_SPEEX_DEC (gst_pad_get_parent (pad));
-
- GST_LOG_OBJECT (dec, "handling %s event", GST_EVENT_TYPE_NAME (event));
-
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_CAPS:
- {
- GstCaps *caps;
-
- gst_event_parse_caps (event, &caps);
- ret = speex_dec_sink_setcaps (pad, caps);
- gst_event_unref (event);
- break;
- }
- case GST_EVENT_SEGMENT:{
- GstSegment segment;
-
- gst_event_copy_segment (event, &segment);
-
- if (segment.format != GST_FORMAT_TIME)
- goto newseg_wrong_format;
-
- if (segment.rate <= 0.0)
- goto newseg_wrong_rate;
-
-#if 0
- if (update) {
- /* time progressed without data, see if we can fill the gap with
- * some concealment data */
- if (dec->segment.position < start) {
- GstClockTime duration;
-
- duration = start - dec->segment.position;
- speex_dec_chain_parse_data (dec, NULL, dec->segment.position,
- duration);
- }
- }
-#endif
-
- /* now configure the values */
- dec->segment = segment;
-
- GST_DEBUG_OBJECT (dec, "segment now: %" GST_SEGMENT_FORMAT, &segment);
- ret = gst_pad_push_event (dec->srcpad, event);
- break;
- }
- default:
- ret = gst_pad_event_default (pad, event);
- break;
- }
+ GstSpeexDec *sd = GST_SPEEX_DEC (dec);
- gst_object_unref (dec);
- return ret;
+ GST_DEBUG_OBJECT (dec, "stop");
+ gst_speex_dec_reset (sd);
- /* ERRORS */
-newseg_wrong_format:
- {
- GST_DEBUG_OBJECT (dec, "received non TIME newsegment");
- gst_object_unref (dec);
- return FALSE;
- }
-newseg_wrong_rate:
- {
- GST_DEBUG_OBJECT (dec, "negative rates not supported yet");
- gst_object_unref (dec);
- return FALSE;
- }
+ return TRUE;
}
static GstFlowReturn
-speex_dec_chain_parse_header (GstSpeexDec * dec, GstBuffer * buf)
+gst_speex_dec_parse_header (GstSpeexDec * dec, GstBuffer * buf)
{
GstCaps *caps;
char *data;
@@ -603,7 +234,7 @@ speex_dec_chain_parse_header (GstSpeexDec * dec, GstBuffer * buf)
"rate", G_TYPE_INT, dec->header->rate,
"channels", G_TYPE_INT, dec->header->nb_channels, NULL);
- if (!gst_pad_set_caps (dec->srcpad, caps))
+ if (!gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps))
goto nego_failed;
gst_caps_unref (caps);
@@ -640,7 +271,7 @@ nego_failed:
}
static GstFlowReturn
-speex_dec_chain_parse_comments (GstSpeexDec * dec, GstBuffer * buf)
+gst_speex_dec_parse_comments (GstSpeexDec * dec, GstBuffer * buf)
{
GstTagList *list;
gchar *ver, *encoder = NULL;
@@ -675,7 +306,8 @@ speex_dec_chain_parse_comments (GstSpeexDec * dec, GstBuffer * buf)
GST_INFO_OBJECT (dec, "tags: %" GST_PTR_FORMAT, list);
- gst_element_found_tags_for_pad (GST_ELEMENT (dec), dec->srcpad, list);
+ gst_element_found_tags_for_pad (GST_ELEMENT (dec),
+ GST_AUDIO_DECODER_SRC_PAD (dec), list);
g_free (encoder);
g_free (ver);
@@ -683,9 +315,47 @@ speex_dec_chain_parse_comments (GstSpeexDec * dec, GstBuffer * buf)
return GST_FLOW_OK;
}
+static gboolean
+gst_speex_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
+{
+ GstSpeexDec *dec = GST_SPEEX_DEC (bdec);
+ gboolean ret = TRUE;
+ GstStructure *s;
+ const GValue *streamheader;
+
+ s = gst_caps_get_structure (caps, 0);
+ if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
+ G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
+ gst_value_array_get_size (streamheader) >= 2) {
+ const GValue *header, *vorbiscomment;
+ GstBuffer *buf;
+ GstFlowReturn res = GST_FLOW_OK;
+
+ header = gst_value_array_get_value (streamheader, 0);
+ if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
+ buf = gst_value_get_buffer (header);
+ res = gst_speex_dec_parse_header (dec, buf);
+ if (res != GST_FLOW_OK)
+ goto done;
+ gst_buffer_replace (&dec->streamheader, buf);
+ }
+
+ vorbiscomment = gst_value_array_get_value (streamheader, 1);
+ if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
+ buf = gst_value_get_buffer (vorbiscomment);
+ res = gst_speex_dec_parse_comments (dec, buf);
+ if (res != GST_FLOW_OK)
+ goto done;
+ gst_buffer_replace (&dec->vorbiscomment, buf);
+ }
+ }
+
+done:
+ return ret;
+}
+
static GstFlowReturn
-speex_dec_chain_parse_data (GstSpeexDec * dec, GstBuffer * buf,
- GstClockTime timestamp, GstClockTime duration)
+gst_speex_dec_parse_data (GstSpeexDec * dec, GstBuffer * buf)
{
GstFlowReturn res = GST_FLOW_OK;
gint i, fpp;
@@ -696,13 +366,7 @@ speex_dec_chain_parse_data (GstSpeexDec * dec, GstBuffer * buf,
if (!dec->frame_duration)
goto not_negotiated;
- if (timestamp != -1) {
- dec->segment.position = timestamp;
- } else {
- timestamp = dec->segment.position;
- }
-
- if (buf) {
+ if (G_LIKELY (gst_buffer_get_size (buf))) {
/* send data to the bitstream */
data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
speex_bits_read_from (&dec->bits, data, size);
@@ -711,16 +375,16 @@ speex_dec_chain_parse_data (GstSpeexDec * dec, GstBuffer * buf,
fpp = dec->header->frames_per_packet;
bits = &dec->bits;
- GST_DEBUG_OBJECT (dec, "received buffer of size %u, fpp %d, %d bits", size,
- fpp, speex_bits_remaining (bits));
+ GST_DEBUG_OBJECT (dec, "received buffer of size %u, fpp %d, %d bits",
+ size, fpp, speex_bits_remaining (bits));
} else {
+ /* FIXME ? actually consider how much concealment is needed */
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "creating concealment data");
fpp = dec->header->frames_per_packet;
bits = NULL;
}
-
/* now decode each frame, catering for unknown number of them (e.g. rtp) */
for (i = 0; i < fpp; i++) {
GstBuffer *outbuf;
@@ -730,9 +394,10 @@ speex_dec_chain_parse_data (GstSpeexDec * dec, GstBuffer * buf,
GST_LOG_OBJECT (dec, "decoding frame %d/%d, %d bits remaining", i, fpp,
bits ? speex_bits_remaining (bits) : -1);
#if 0
- res = gst_pad_alloc_buffer_and_set_caps (dec->srcpad,
+ res =
+ gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec),
GST_BUFFER_OFFSET_NONE, dec->frame_size * dec->header->nb_channels * 2,
- GST_PAD_CAPS (dec->srcpad), &outbuf);
+ GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
@@ -754,39 +419,27 @@ speex_dec_chain_parse_data (GstSpeexDec * dec, GstBuffer * buf,
if (fpp == 0 && speex_bits_remaining (bits) < 8) {
/* if we did not know how many frames to expect, then we get this
at the end if there are leftover bits to pad to the next byte */
+ GST_DEBUG_OBJECT (dec, "Discarding leftover bits");
} else {
GST_WARNING_OBJECT (dec, "Unexpected end of stream found");
}
+ gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), NULL, 1);
gst_buffer_unref (outbuf);
- outbuf = NULL;
- break;
} else if (ret == -2) {
GST_WARNING_OBJECT (dec, "Decoding error: corrupted stream?");
+ gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), NULL, 1);
gst_buffer_unref (outbuf);
- outbuf = NULL;
- break;
}
if (bits && speex_bits_remaining (bits) < 0) {
GST_WARNING_OBJECT (dec, "Decoding overflow: corrupted stream?");
+ gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), NULL, 1);
gst_buffer_unref (outbuf);
- outbuf = NULL;
- break;
}
if (dec->header->nb_channels == 2)
speex_decode_stereo_int (out_data, dec->frame_size, dec->stereo);
- GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
- GST_BUFFER_DURATION (outbuf) = dec->frame_duration;
-
- dec->segment.position += dec->frame_duration;
- timestamp = dec->segment.position;
-
- GST_LOG_OBJECT (dec, "pushing buffer with ts=%" GST_TIME_FORMAT ", dur=%"
- GST_TIME_FORMAT, GST_TIME_ARGS (timestamp),
- GST_TIME_ARGS (dec->frame_duration));
-
- res = gst_pad_push (dec->srcpad, outbuf);
+ res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
@@ -826,48 +479,55 @@ memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2)
}
static GstFlowReturn
-speex_dec_chain (GstPad * pad, GstBuffer * buf)
+gst_speex_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
{
GstFlowReturn res;
GstSpeexDec *dec;
- dec = GST_SPEEX_DEC (gst_pad_get_parent (pad));
+ /* no fancy draining */
+ if (G_UNLIKELY (!buf))
+ return GST_FLOW_OK;
+
+ dec = GST_SPEEX_DEC (bdec);
/* If we have the streamheader and vorbiscomment from the caps already
* ignore them here */
if (dec->streamheader && dec->vorbiscomment) {
if (memcmp_buffers (dec->streamheader, buf)) {
+ GST_DEBUG_OBJECT (dec, "found streamheader");
+ gst_audio_decoder_finish_frame (bdec, NULL, 1);
res = GST_FLOW_OK;
} else if (memcmp_buffers (dec->vorbiscomment, buf)) {
+ GST_DEBUG_OBJECT (dec, "found vorbiscomments");
+ gst_audio_decoder_finish_frame (bdec, NULL, 1);
res = GST_FLOW_OK;
} else {
- res =
- speex_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
- GST_BUFFER_DURATION (buf));
+ res = gst_speex_dec_parse_data (dec, buf);
}
} else {
/* Otherwise fall back to packet counting and assume that the
* first two packets are the headers. */
switch (dec->packetno) {
case 0:
- res = speex_dec_chain_parse_header (dec, buf);
+ GST_DEBUG_OBJECT (dec, "counted streamheader");
+ res = gst_speex_dec_parse_header (dec, buf);
+ gst_audio_decoder_finish_frame (bdec, NULL, 1);
break;
case 1:
- res = speex_dec_chain_parse_comments (dec, buf);
+ GST_DEBUG_OBJECT (dec, "counted vorbiscomments");
+ res = gst_speex_dec_parse_comments (dec, buf);
+ gst_audio_decoder_finish_frame (bdec, NULL, 1);
break;
default:
- res =
- speex_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
- GST_BUFFER_DURATION (buf));
+ {
+ res = gst_speex_dec_parse_data (dec, buf);
break;
+ }
}
}
dec->packetno++;
- gst_buffer_unref (buf);
- gst_object_unref (dec);
-
return res;
}
@@ -906,37 +566,3 @@ gst_speex_dec_set_property (GObject * object, guint prop_id,
break;
}
}
-
-
-static GstStateChangeReturn
-speex_dec_change_state (GstElement * element, GstStateChange transition)
-{
- GstStateChangeReturn ret;
- GstSpeexDec *dec = GST_SPEEX_DEC (element);
-
- switch (transition) {
- case GST_STATE_CHANGE_NULL_TO_READY:
- case GST_STATE_CHANGE_READY_TO_PAUSED:
- case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
- default:
- break;
- }
-
- ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
- if (ret != GST_STATE_CHANGE_SUCCESS)
- return ret;
-
- switch (transition) {
- case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
- break;
- case GST_STATE_CHANGE_PAUSED_TO_READY:
- gst_speex_dec_reset (dec);
- break;
- case GST_STATE_CHANGE_READY_TO_NULL:
- break;
- default:
- break;
- }
-
- return ret;
-}
diff --git a/ext/speex/gstspeexdec.h b/ext/speex/gstspeexdec.h
index 660d8053d..8187af87c 100644
--- a/ext/speex/gstspeexdec.h
+++ b/ext/speex/gstspeexdec.h
@@ -22,6 +22,8 @@
#define __GST_SPEEX_DEC_H__
#include <gst/gst.h>
+#include <gst/audio/gstaudiodecoder.h>
+
#include <speex/speex.h>
#include <speex/speex_callbacks.h>
#include <speex/speex_header.h>
@@ -44,11 +46,7 @@ typedef struct _GstSpeexDec GstSpeexDec;
typedef struct _GstSpeexDecClass GstSpeexDecClass;
struct _GstSpeexDec {
- GstElement element;
-
- /* pads */
- GstPad *sinkpad;
- GstPad *srcpad;
+ GstAudioDecoder element;
void *state;
SpeexStereoState *stereo;
@@ -67,14 +65,12 @@ struct _GstSpeexDec {
GstClockTime frame_duration;
guint64 packetno;
- GstSegment segment; /* STREAM LOCK */
-
GstBuffer *streamheader;
GstBuffer *vorbiscomment;
};
struct _GstSpeexDecClass {
- GstElementClass parent_class;
+ GstAudioDecoderClass parent_class;
};
GType gst_speex_dec_get_type (void);
diff --git a/ext/speex/gstspeexenc.c b/ext/speex/gstspeexenc.c
index e202d9224..e4eda25cf 100644
--- a/ext/speex/gstspeexenc.c
+++ b/ext/speex/gstspeexenc.c
@@ -113,43 +113,28 @@ gst_speex_enc_mode_get_type (void)
return speex_enc_mode_type;
}
-#if 0
-static const GstFormat *
-gst_speex_enc_get_formats (GstPad * pad)
-{
- static const GstFormat src_formats[] = {
- GST_FORMAT_BYTES,
- GST_FORMAT_TIME,
- 0
- };
- static const GstFormat sink_formats[] = {
- GST_FORMAT_BYTES,
- GST_FORMAT_DEFAULT,
- GST_FORMAT_TIME,
- 0
- };
-
- return (GST_PAD_IS_SRC (pad) ? src_formats : sink_formats);
-}
-#endif
-
static void gst_speex_enc_finalize (GObject * object);
-static gboolean gst_speex_enc_sink_event (GstPad * pad, GstEvent * event);
-static GstFlowReturn gst_speex_enc_chain (GstPad * pad, GstBuffer * buf);
static gboolean gst_speex_enc_setup (GstSpeexEnc * enc);
static void gst_speex_enc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_speex_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
-static GstStateChangeReturn gst_speex_enc_change_state (GstElement * element,
- GstStateChange transition);
-static GstFlowReturn gst_speex_enc_encode (GstSpeexEnc * enc, gboolean flush);
+static GstFlowReturn gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf);
+
+static gboolean gst_speex_enc_start (GstAudioEncoder * enc);
+static gboolean gst_speex_enc_stop (GstAudioEncoder * enc);
+static gboolean gst_speex_enc_set_format (GstAudioEncoder * enc,
+ GstAudioInfo * info);
+static GstFlowReturn gst_speex_enc_handle_frame (GstAudioEncoder * enc,
+ GstBuffer * in_buf);
+static gboolean gst_speex_enc_sink_event (GstAudioEncoder * enc,
+ GstEvent * event);
#define gst_speex_enc_parent_class parent_class
-G_DEFINE_TYPE_WITH_CODE (GstSpeexEnc, gst_speex_enc, GST_TYPE_ELEMENT,
+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));
@@ -158,62 +143,67 @@ 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->start = GST_DEBUG_FUNCPTR (gst_speex_enc_start);
+ 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);
+
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_QUALITY,
g_param_spec_float ("quality", "Quality", "Encoding quality",
0.0, 10.0, DEFAULT_QUALITY,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BITRATE,
g_param_spec_int ("bitrate", "Encoding Bit-rate",
"Specify an encoding bit-rate (in bps). (0 = automatic)",
0, G_MAXINT, DEFAULT_BITRATE,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode", "The encoding mode",
GST_TYPE_SPEEX_ENC_MODE, GST_SPEEX_ENC_MODE_AUTO,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_VBR,
g_param_spec_boolean ("vbr", "VBR",
"Enable variable bit-rate", DEFAULT_VBR,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_ABR,
g_param_spec_int ("abr", "ABR",
"Enable average bit-rate (0 = disabled)",
0, G_MAXINT, DEFAULT_ABR,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_VAD,
g_param_spec_boolean ("vad", "VAD",
"Enable voice activity detection", DEFAULT_VAD,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_DTX,
g_param_spec_boolean ("dtx", "DTX",
"Enable discontinuous transmission", DEFAULT_DTX,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_COMPLEXITY,
g_param_spec_int ("complexity", "Complexity",
"Set encoding complexity",
0, G_MAXINT, DEFAULT_COMPLEXITY,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_NFRAMES,
g_param_spec_int ("nframes", "NFrames",
"Number of frames per buffer",
0, G_MAXINT, DEFAULT_NFRAMES,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LAST_MESSAGE,
g_param_spec_string ("last-message", "last-message",
"The last status message", NULL,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
- gstelement_class->change_state =
- GST_DEBUG_FUNCPTR (gst_speex_enc_change_state);
-
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_add_pad_template (gstelement_class,
@@ -233,166 +223,49 @@ gst_speex_enc_finalize (GObject * object)
enc = GST_SPEEX_ENC (object);
g_free (enc->last_message);
- g_object_unref (enc->adapter);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
-static gboolean
-gst_speex_enc_sink_setcaps (GstPad * pad, GstCaps * caps)
-{
- GstSpeexEnc *enc;
- GstStructure *structure;
-
- enc = GST_SPEEX_ENC (GST_PAD_PARENT (pad));
- enc->setup = FALSE;
-
- structure = gst_caps_get_structure (caps, 0);
- gst_structure_get_int (structure, "channels", &enc->channels);
- gst_structure_get_int (structure, "rate", &enc->rate);
-
- gst_speex_enc_setup (enc);
-
- return enc->setup;
-}
-
-
-static GstCaps *
-gst_speex_enc_sink_getcaps (GstPad * pad, GstCaps * filter)
+static void
+gst_speex_enc_init (GstSpeexEnc * enc)
{
- GstCaps *caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
- GstCaps *peercaps = NULL;
- GstSpeexEnc *enc = GST_SPEEX_ENC (gst_pad_get_parent_element (pad));
-
- peercaps = gst_pad_peer_get_caps (enc->srcpad, filter);
-
- if (peercaps) {
- if (!gst_caps_is_empty (peercaps) && !gst_caps_is_any (peercaps)) {
- GstStructure *ps = gst_caps_get_structure (peercaps, 0);
- GstStructure *s = gst_caps_get_structure (caps, 0);
- gint rate, channels;
-
- if (gst_structure_get_int (ps, "rate", &rate)) {
- gst_structure_fixate_field_nearest_int (s, "rate", rate);
- }
-
- if (gst_structure_get_int (ps, "channels", &channels)) {
- gst_structure_fixate_field_nearest_int (s, "channels", channels);
- }
- }
- gst_caps_unref (peercaps);
- }
-
- gst_object_unref (enc);
+ GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
- return caps;
+ /* arrange granulepos marking (and required perfect ts) */
+ gst_audio_encoder_set_mark_granule (benc, TRUE);
+ gst_audio_encoder_set_perfect_timestamp (benc, TRUE);
}
-
static gboolean
-gst_speex_enc_convert_src (GstPad * pad, GstFormat src_format, gint64 src_value,
- GstFormat dest_format, gint64 * dest_value)
+gst_speex_enc_start (GstAudioEncoder * benc)
{
- gboolean res = TRUE;
- GstSpeexEnc *enc;
- gint64 avg;
-
- enc = GST_SPEEX_ENC (GST_PAD_PARENT (pad));
+ GstSpeexEnc *enc = GST_SPEEX_ENC (benc);
- if (enc->samples_in == 0 || enc->bytes_out == 0 || enc->rate == 0)
- return FALSE;
-
- avg = (enc->bytes_out * enc->rate) / (enc->samples_in);
+ GST_DEBUG_OBJECT (enc, "start");
+ speex_bits_init (&enc->bits);
+ enc->tags = gst_tag_list_new ();
+ enc->header_sent = FALSE;
- switch (src_format) {
- case GST_FORMAT_BYTES:
- switch (dest_format) {
- case GST_FORMAT_TIME:
- *dest_value = src_value * GST_SECOND / avg;
- break;
- default:
- res = FALSE;
- }
- break;
- case GST_FORMAT_TIME:
- switch (dest_format) {
- case GST_FORMAT_BYTES:
- *dest_value = src_value * avg / GST_SECOND;
- break;
- default:
- res = FALSE;
- }
- break;
- default:
- res = FALSE;
- }
- return res;
+ return TRUE;
}
static gboolean
-gst_speex_enc_convert_sink (GstPad * pad, GstFormat src_format,
- gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
+gst_speex_enc_stop (GstAudioEncoder * benc)
{
- gboolean res = TRUE;
- guint scale = 1;
- gint bytes_per_sample;
- GstSpeexEnc *enc;
+ GstSpeexEnc *enc = GST_SPEEX_ENC (benc);
- enc = GST_SPEEX_ENC (GST_PAD_PARENT (pad));
-
- bytes_per_sample = enc->channels * 2;
-
- switch (src_format) {
- case GST_FORMAT_BYTES:
- switch (*dest_format) {
- case GST_FORMAT_DEFAULT:
- if (bytes_per_sample == 0)
- return FALSE;
- *dest_value = src_value / bytes_per_sample;
- break;
- case GST_FORMAT_TIME:
- {
- gint byterate = bytes_per_sample * enc->rate;
-
- if (byterate == 0)
- return FALSE;
- *dest_value = src_value * GST_SECOND / byterate;
- break;
- }
- default:
- res = FALSE;
- }
- break;
- case GST_FORMAT_DEFAULT:
- switch (*dest_format) {
- case GST_FORMAT_BYTES:
- *dest_value = src_value * bytes_per_sample;
- break;
- case GST_FORMAT_TIME:
- if (enc->rate == 0)
- return FALSE;
- *dest_value = src_value * GST_SECOND / enc->rate;
- break;
- default:
- res = FALSE;
- }
- break;
- case GST_FORMAT_TIME:
- switch (*dest_format) {
- case GST_FORMAT_BYTES:
- scale = bytes_per_sample;
- /* fallthrough */
- case GST_FORMAT_DEFAULT:
- *dest_value = src_value * scale * enc->rate / GST_SECOND;
- break;
- default:
- res = FALSE;
- }
- break;
- default:
- res = FALSE;
+ GST_DEBUG_OBJECT (enc, "stop");
+ enc->header_sent = FALSE;
+ if (enc->state) {
+ speex_encoder_destroy (enc->state);
+ enc->state = NULL;
}
- return res;
+ speex_bits_destroy (&enc->bits);
+ gst_tag_list_free (enc->tags);
+ enc->tags = NULL;
+
+ return TRUE;
}
static gint64
@@ -405,185 +278,45 @@ gst_speex_enc_get_latency (GstSpeexEnc * enc)
return 34 * GST_MSECOND;
}
-static const GstQueryType *
-gst_speex_enc_get_query_types (GstPad * pad)
-{
- static const GstQueryType gst_speex_enc_src_query_types[] = {
- GST_QUERY_POSITION,
- GST_QUERY_DURATION,
- GST_QUERY_CONVERT,
- GST_QUERY_LATENCY,
- 0
- };
-
- return gst_speex_enc_src_query_types;
-}
-
static gboolean
-gst_speex_enc_src_query (GstPad * pad, GstQuery * query)
+gst_speex_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
- gboolean res = TRUE;
GstSpeexEnc *enc;
- enc = GST_SPEEX_ENC (gst_pad_get_parent (pad));
-
- switch (GST_QUERY_TYPE (query)) {
- case GST_QUERY_POSITION:
- {
- GstFormat req_fmt;
- gint64 pos, val;
-
- gst_query_parse_position (query, &req_fmt, NULL);
- if ((res = gst_pad_query_peer_position (enc->sinkpad, req_fmt, &val))) {
- gst_query_set_position (query, req_fmt, val);
- break;
- }
-
- res = gst_pad_query_peer_position (enc->sinkpad, GST_FORMAT_TIME, &pos);
- if (!res)
- break;
-
- if ((res =
- gst_pad_query_peer_convert (enc->sinkpad, GST_FORMAT_TIME, pos,
- req_fmt, &val))) {
- gst_query_set_position (query, req_fmt, val);
- }
- break;
- }
- case GST_QUERY_DURATION:
- {
- GstFormat req_fmt;
- gint64 dur, val;
-
- gst_query_parse_duration (query, &req_fmt, NULL);
- if ((res = gst_pad_query_peer_duration (enc->sinkpad, req_fmt, &val))) {
- gst_query_set_duration (query, req_fmt, val);
- break;
- }
+ enc = GST_SPEEX_ENC (benc);
- res = gst_pad_query_peer_duration (enc->sinkpad, GST_FORMAT_TIME, &dur);
- if (!res)
- break;
+ enc->channels = GST_AUDIO_INFO_CHANNELS (info);
+ enc->rate = GST_AUDIO_INFO_RATE (info);
- if ((res =
- gst_pad_query_peer_convert (enc->sinkpad, GST_FORMAT_TIME, dur,
- req_fmt, &val))) {
- gst_query_set_duration (query, req_fmt, val);
- }
- break;
- }
- case GST_QUERY_CONVERT:
- {
- GstFormat src_fmt, dest_fmt;
- gint64 src_val, dest_val;
-
- gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
- if (!(res = gst_speex_enc_convert_src (pad, src_fmt, src_val, dest_fmt,
- &dest_val)))
- goto error;
- gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
- break;
- }
- case GST_QUERY_LATENCY:
- {
- gboolean live;
- GstClockTime min_latency, max_latency;
- gint64 latency;
-
- if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
- gst_query_parse_latency (query, &live, &min_latency, &max_latency);
- GST_LOG_OBJECT (pad, "Upstream latency: %" GST_PTR_FORMAT, query);
-
- latency = gst_speex_enc_get_latency (enc);
-
- /* add our latency */
- min_latency += latency;
- if (max_latency != -1)
- max_latency += latency;
-
- gst_query_set_latency (query, live, min_latency, max_latency);
- GST_LOG_OBJECT (pad, "Adjusted latency: %" GST_PTR_FORMAT, query);
- }
- break;
- }
- default:
- res = gst_pad_peer_query (enc->sinkpad, query);
- break;
+ /* handle reconfigure */
+ if (enc->state) {
+ speex_encoder_destroy (enc->state);
+ enc->state = NULL;
}
-error:
-
- gst_object_unref (enc);
-
- return res;
-}
+ if (!gst_speex_enc_setup (enc))
+ return FALSE;
-static gboolean
-gst_speex_enc_sink_query (GstPad * pad, GstQuery * query)
-{
- gboolean res = TRUE;
+ /* feedback to base class */
+ gst_audio_encoder_set_latency (benc,
+ gst_speex_enc_get_latency (enc), gst_speex_enc_get_latency (enc));
+ gst_audio_encoder_set_lookahead (benc, enc->lookahead);
- switch (GST_QUERY_TYPE (query)) {
- case GST_QUERY_CONVERT:
- {
- GstFormat src_fmt, dest_fmt;
- gint64 src_val, dest_val;
-
- gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
- if (!(res =
- gst_speex_enc_convert_sink (pad, src_fmt, src_val, &dest_fmt,
- &dest_val)))
- goto error;
- gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
- break;
- }
- default:
- res = gst_pad_query_default (pad, query);
- break;
+ if (enc->nframes == 0) {
+ /* as many frames as available input allows */
+ gst_audio_encoder_set_frame_samples_min (benc, enc->frame_size);
+ gst_audio_encoder_set_frame_samples_max (benc, enc->frame_size);
+ gst_audio_encoder_set_frame_max (benc, 0);
+ } else {
+ /* exactly as many frames as configured */
+ gst_audio_encoder_set_frame_samples_min (benc,
+ enc->frame_size * enc->nframes);
+ gst_audio_encoder_set_frame_samples_max (benc,
+ enc->frame_size * enc->nframes);
+ gst_audio_encoder_set_frame_max (benc, 1);
}
-error:
- return res;
-}
-
-static void
-gst_speex_enc_init (GstSpeexEnc * enc)
-{
- enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
- gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
- gst_pad_set_event_function (enc->sinkpad,
- GST_DEBUG_FUNCPTR (gst_speex_enc_sink_event));
- gst_pad_set_chain_function (enc->sinkpad,
- GST_DEBUG_FUNCPTR (gst_speex_enc_chain));
- gst_pad_set_getcaps_function (enc->sinkpad,
- GST_DEBUG_FUNCPTR (gst_speex_enc_sink_getcaps));
- gst_pad_set_query_function (enc->sinkpad,
- GST_DEBUG_FUNCPTR (gst_speex_enc_sink_query));
-
- enc->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
- gst_pad_set_query_function (enc->srcpad,
- GST_DEBUG_FUNCPTR (gst_speex_enc_src_query));
- gst_pad_set_query_type_function (enc->srcpad,
- GST_DEBUG_FUNCPTR (gst_speex_enc_get_query_types));
- gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
-
- enc->channels = -1;
- enc->rate = -1;
-
- enc->quality = DEFAULT_QUALITY;
- enc->bitrate = DEFAULT_BITRATE;
- enc->mode = DEFAULT_MODE;
- enc->vbr = DEFAULT_VBR;
- enc->abr = DEFAULT_ABR;
- enc->vad = DEFAULT_VAD;
- enc->dtx = DEFAULT_DTX;
- enc->complexity = DEFAULT_COMPLEXITY;
- enc->nframes = DEFAULT_NFRAMES;
-
- enc->setup = FALSE;
- enc->header_sent = FALSE;
-
- enc->adapter = gst_adapter_new ();
+ return TRUE;
}
static GstBuffer *
@@ -610,7 +343,7 @@ gst_speex_enc_create_metadata_buffer (GstSpeexEnc * enc)
0, "Encoded with GStreamer Speexenc");
gst_tag_list_free (merged_tags);
- GST_BUFFER_OFFSET (comments) = enc->bytes_out;
+ GST_BUFFER_OFFSET (comments) = 0;
GST_BUFFER_OFFSET_END (comments) = 0;
return comments;
@@ -628,8 +361,6 @@ gst_speex_enc_set_last_msg (GstSpeexEnc * enc, const gchar * msg)
static gboolean
gst_speex_enc_setup (GstSpeexEnc * enc)
{
- enc->setup = FALSE;
-
switch (enc->mode) {
case GST_SPEEX_ENC_MODE_UWB:
GST_LOG_OBJECT (enc, "configuring for requested UWB mode");
@@ -746,105 +477,27 @@ gst_speex_enc_setup (GstSpeexEnc * enc)
GST_LOG_OBJECT (enc, "we have frame size %d, lookahead %d", enc->frame_size,
enc->lookahead);
- enc->setup = TRUE;
-
return TRUE;
}
-/* prepare a buffer for transmission */
-static GstBuffer *
-gst_speex_enc_buffer_from_data (GstSpeexEnc * enc, guchar * data,
- gint data_len, guint64 granulepos)
-{
- GstBuffer *outbuf;
-
- outbuf = gst_buffer_new_and_alloc (data_len);
- gst_buffer_fill (outbuf, 0, data, data_len);
- GST_BUFFER_OFFSET (outbuf) = enc->bytes_out;
- GST_BUFFER_OFFSET_END (outbuf) = granulepos;
-
- GST_LOG_OBJECT (enc, "encoded buffer of %d bytes", data_len);
- return outbuf;
-}
-
-
-/* push out the buffer and do internal bookkeeping */
+/* push out the buffer */
static GstFlowReturn
gst_speex_enc_push_buffer (GstSpeexEnc * enc, GstBuffer * buffer)
{
- guint size;
-
- size = gst_buffer_get_size (buffer);
- enc->bytes_out += size;
-
- GST_DEBUG_OBJECT (enc, "pushing output buffer of size %u", size);
-
- return gst_pad_push (enc->srcpad, buffer);
-}
-
-static GstCaps *
-gst_speex_enc_set_header_on_caps (GstCaps * caps, GstBuffer * buf1,
- GstBuffer * buf2)
-{
- GstStructure *structure = NULL;
- GstBuffer *buf;
- GValue array = { 0 };
- GValue value = { 0 };
-
- caps = gst_caps_make_writable (caps);
- structure = gst_caps_get_structure (caps, 0);
+ GST_DEBUG_OBJECT (enc, "pushing output buffer of size %u",
+ gst_buffer_get_size (buffer));
- g_assert (gst_buffer_is_writable (buf1));
- g_assert (gst_buffer_is_writable (buf2));
-
- /* mark buffers */
- GST_BUFFER_FLAG_SET (buf1, GST_BUFFER_FLAG_IN_CAPS);
- GST_BUFFER_FLAG_SET (buf2, GST_BUFFER_FLAG_IN_CAPS);
-
- /* put buffers in a fixed list */
- g_value_init (&array, GST_TYPE_ARRAY);
- g_value_init (&value, GST_TYPE_BUFFER);
- buf = gst_buffer_copy (buf1);
- gst_value_set_buffer (&value, buf);
- gst_buffer_unref (buf);
- gst_value_array_append_value (&array, &value);
- g_value_unset (&value);
- g_value_init (&value, GST_TYPE_BUFFER);
- buf = gst_buffer_copy (buf2);
- gst_value_set_buffer (&value, buf);
- gst_buffer_unref (buf);
- gst_value_array_append_value (&array, &value);
- gst_structure_set_value (structure, "streamheader", &array);
- g_value_unset (&value);
- g_value_unset (&array);
-
- return caps;
+ return gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (enc), buffer);
}
-
static gboolean
-gst_speex_enc_sink_event (GstPad * pad, GstEvent * event)
+gst_speex_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
{
- gboolean res = TRUE;
GstSpeexEnc *enc;
- enc = GST_SPEEX_ENC (gst_pad_get_parent (pad));
+ enc = GST_SPEEX_ENC (benc);
switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_CAPS:
- {
- GstCaps *caps;
-
- gst_event_parse_caps (event, &caps);
- res = gst_speex_enc_sink_setcaps (pad, caps);
- gst_event_unref (event);
- break;
- }
- case GST_EVENT_EOS:
- if (enc->setup)
- gst_speex_enc_encode (enc, TRUE);
- res = gst_pad_event_default (pad, event);
- break;
case GST_EVENT_TAG:
{
if (enc->tags) {
@@ -856,116 +509,165 @@ gst_speex_enc_sink_event (GstPad * pad, GstEvent * event)
} else {
g_assert_not_reached ();
}
- res = gst_pad_event_default (pad, event);
break;
}
default:
- res = gst_pad_event_default (pad, event);
break;
}
- gst_object_unref (enc);
-
- return res;
+ /* we only peeked, let base class handle it */
+ return FALSE;
}
static GstFlowReturn
-gst_speex_enc_encode (GstSpeexEnc * enc, gboolean flush)
+gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf)
{
gint frame_size = enc->frame_size;
- gint bytes = frame_size * 2 * enc->channels;
+ gint bytes = frame_size * 2 * enc->channels, samples;
+ gint outsize, written, dtx_ret = 0;
+ guint8 *data, *bdata, *outdata;
+ gsize bsize, size;
+ GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
- if (flush && gst_adapter_available (enc->adapter) % bytes != 0) {
- guint diff = gst_adapter_available (enc->adapter) % bytes;
- GstBuffer *buf = gst_buffer_new_and_alloc (diff);
- gst_buffer_memset (buf, 0, 0, diff);
- gst_adapter_push (enc->adapter, buf);
- }
+ if (G_LIKELY (buf)) {
+ bdata = gst_buffer_map (buf, &bsize, NULL, GST_MAP_READ);
+
+ if (G_UNLIKELY (bsize % bytes)) {
+ GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
- while (gst_adapter_available (enc->adapter) >= bytes) {
- gint16 *data;
- gint outsize, written, dtx_ret;
- GstBuffer *outbuf;
- gchar *outdata;
+ size = ((bsize / bytes) + 1) * bytes;
+ 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");
+ goto done;
+ }
- data = (gint16 *) gst_adapter_take (enc->adapter, bytes);
+ samples = size / (2 * enc->channels);
+ speex_bits_reset (&enc->bits);
- enc->samples_in += frame_size;
+ /* FIXME what about dropped samples if DTS enabled ?? */
+ while (size) {
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)", frame_size, bytes);
if (enc->channels == 2) {
- speex_encode_stereo_int (data, frame_size, &enc->bits);
+ speex_encode_stereo_int ((gint16 *) data, frame_size, &enc->bits);
}
- dtx_ret = speex_encode_int (enc->state, data, &enc->bits);
-
- g_free (data);
+ dtx_ret += speex_encode_int (enc->state, (gint16 *) data, &enc->bits);
- enc->frameno++;
- enc->frameno_out++;
+ data += bytes;
+ size -= bytes;
+ }
- if ((enc->frameno % enc->nframes) != 0)
- continue;
+ speex_bits_insert_terminator (&enc->bits);
+ outsize = speex_bits_nbytes (&enc->bits);
- 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 (enc->srcpad,
- GST_BUFFER_OFFSET_NONE, outsize, GST_PAD_CAPS (enc->srcpad), &outbuf);
- if ((GST_FLOW_OK != ret))
- goto done;
+ 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);
+ outbuf = gst_buffer_new_allocate (NULL, outsize, 0);
+ outdata = gst_buffer_map (outbuf, NULL, NULL, GST_MAP_WRITE);
- outdata = gst_buffer_map (outbuf, NULL, NULL, GST_MAP_WRITE);
- written = speex_bits_write (&enc->bits, outdata, 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_unmap (outbuf, outdata, written);
+ if (G_UNLIKELY (written < outsize)) {
+ GST_ERROR_OBJECT (enc, "short write: %d < %d bytes", written, outsize);
+ } 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);
- speex_bits_reset (&enc->bits);
+ ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc),
+ outbuf, samples);
- if (!dtx_ret)
- GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
+done:
+ return ret;
+}
- GST_BUFFER_TIMESTAMP (outbuf) = enc->start_ts +
- gst_util_uint64_scale_int ((enc->frameno_out -
- enc->nframes) * frame_size - enc->lookahead, GST_SECOND, enc->rate);
- GST_BUFFER_DURATION (outbuf) =
- gst_util_uint64_scale_int (frame_size * enc->nframes, GST_SECOND,
- enc->rate);
- /* set gp time and granulepos; see gst-plugins-base/ext/ogg/README */
- GST_BUFFER_OFFSET_END (outbuf) = enc->granulepos_offset +
- ((enc->frameno_out) * frame_size - enc->lookahead);
- GST_BUFFER_OFFSET (outbuf) =
- gst_util_uint64_scale_int (GST_BUFFER_OFFSET_END (outbuf), GST_SECOND,
- enc->rate);
+/*
+ * (really really) FIXME: move into core (dixit tpm)
+ */
+/**
+ * _gst_caps_set_buffer_array:
+ * @caps: 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
+ */
+static GstCaps *
+_gst_caps_set_buffer_array (GstCaps * caps, const gchar * field,
+ GstBuffer * buf, ...)
+{
+ GstStructure *structure = NULL;
+ va_list va;
+ GValue array = { 0 };
+ GValue value = { 0 };
- ret = gst_speex_enc_push_buffer (enc, outbuf);
+ g_return_val_if_fail (caps != NULL, NULL);
+ g_return_val_if_fail (gst_caps_is_fixed (caps), NULL);
+ g_return_val_if_fail (field != NULL, NULL);
- if ((GST_FLOW_OK != ret) && (GST_FLOW_NOT_LINKED != ret))
- goto done;
+ caps = gst_caps_make_writable (caps);
+ structure = gst_caps_get_structure (caps, 0);
+
+ g_value_init (&array, GST_TYPE_ARRAY);
+
+ va_start (va, buf);
+ /* put buffers in a fixed list */
+ while (buf) {
+ g_assert (gst_buffer_is_writable (buf));
+
+ /* mark buffer */
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
+
+ g_value_init (&value, GST_TYPE_BUFFER);
+ buf = gst_buffer_copy (buf);
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
+ gst_value_set_buffer (&value, buf);
+ gst_buffer_unref (buf);
+ gst_value_array_append_value (&array, &value);
+ g_value_unset (&value);
+
+ buf = va_arg (va, GstBuffer *);
}
-done:
+ gst_structure_set_value (structure, field, &array);
+ g_value_unset (&array);
- return ret;
+ return caps;
}
static GstFlowReturn
-gst_speex_enc_chain (GstPad * pad, GstBuffer * buf)
+gst_speex_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
GstSpeexEnc *enc;
GstFlowReturn ret = GST_FLOW_OK;
- enc = GST_SPEEX_ENC (GST_PAD_PARENT (pad));
-
- if (!enc->setup)
- goto not_setup;
+ enc = GST_SPEEX_ENC (benc);
if (!enc->header_sent) {
/* Speex streams begin with two headers; the initial header (with
@@ -981,24 +683,20 @@ gst_speex_enc_chain (GstPad * pad, GstBuffer * buf)
/* create header buffer */
data = (guint8 *) speex_header_to_packet (&enc->header, &data_len);
- buf1 = gst_speex_enc_buffer_from_data (enc, data, data_len, 0);
- free (data);
+ buf1 = gst_buffer_new_wrapped (data, data_len);
+ GST_BUFFER_OFFSET_END (buf1) = 0;
+ GST_BUFFER_OFFSET (buf1) = 0;
/* create comment buffer */
buf2 = gst_speex_enc_create_metadata_buffer (enc);
/* mark and put on caps */
- caps = gst_pad_get_caps (enc->srcpad, NULL);
- caps = gst_speex_enc_set_header_on_caps (caps, buf1, buf2);
-
- gst_caps_set_simple (caps,
- "rate", G_TYPE_INT, enc->rate,
+ caps = gst_caps_new_simple ("audio/x-speex", "rate", G_TYPE_INT, enc->rate,
"channels", G_TYPE_INT, enc->channels, NULL);
+ caps = _gst_caps_set_buffer_array (caps, "streamheader", buf1, buf2, NULL);
/* negotiate with these caps */
GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps);
- gst_pad_set_caps (enc->srcpad, caps);
- gst_caps_unref (caps);
/* push out buffers */
ret = gst_speex_enc_push_buffer (enc, buf1);
@@ -1013,102 +711,16 @@ gst_speex_enc_chain (GstPad * pad, GstBuffer * buf)
if (ret != GST_FLOW_OK)
goto done;
- speex_bits_reset (&enc->bits);
-
enc->header_sent = TRUE;
}
- /* Save the timestamp of the first buffer. This will be later
- * used as offset for all following buffers */
- if (enc->start_ts == GST_CLOCK_TIME_NONE) {
- if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
- enc->start_ts = GST_BUFFER_TIMESTAMP (buf);
- enc->granulepos_offset = gst_util_uint64_scale
- (GST_BUFFER_TIMESTAMP (buf), enc->rate, GST_SECOND);
- } else {
- enc->start_ts = 0;
- enc->granulepos_offset = 0;
- }
- }
-
- /* Check if we have a continous stream, if not drop some samples or the buffer or
- * insert some silence samples */
- if (enc->next_ts != GST_CLOCK_TIME_NONE &&
- GST_BUFFER_TIMESTAMP_IS_VALID (buf) &&
- GST_BUFFER_TIMESTAMP (buf) < enc->next_ts) {
- guint64 diff = enc->next_ts - GST_BUFFER_TIMESTAMP (buf);
- guint64 diff_bytes;
-
- GST_WARNING_OBJECT (enc, "Buffer is older than previous "
- "timestamp + duration (%" GST_TIME_FORMAT "< %" GST_TIME_FORMAT
- "), cannot handle. Clipping buffer.",
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
- GST_TIME_ARGS (enc->next_ts));
-
- diff_bytes = GST_CLOCK_TIME_TO_FRAMES (diff, enc->rate) * enc->channels * 2;
- if (diff_bytes >= gst_buffer_get_size (buf)) {
- gst_buffer_unref (buf);
- return GST_FLOW_OK;
- }
- buf = gst_buffer_make_writable (buf);
- gst_buffer_resize (buf, diff_bytes, -1);
-
- GST_BUFFER_TIMESTAMP (buf) += diff;
- if (GST_BUFFER_DURATION_IS_VALID (buf))
- GST_BUFFER_DURATION (buf) -= diff;
- }
+ GST_DEBUG_OBJECT (enc, "received buffer %p of %u bytes", buf,
+ buf ? gst_buffer_get_size (buf) : 0);
- if (enc->next_ts != GST_CLOCK_TIME_NONE
- && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
- guint64 max_diff =
- gst_util_uint64_scale (enc->frame_size, GST_SECOND, enc->rate);
-
- if (GST_BUFFER_TIMESTAMP (buf) != enc->next_ts &&
- GST_BUFFER_TIMESTAMP (buf) - enc->next_ts > max_diff) {
- GST_WARNING_OBJECT (enc,
- "Discontinuity detected: %" G_GUINT64_FORMAT " > %" G_GUINT64_FORMAT,
- GST_BUFFER_TIMESTAMP (buf) - enc->next_ts, max_diff);
-
- gst_speex_enc_encode (enc, TRUE);
-
- enc->frameno_out = 0;
- enc->start_ts = GST_BUFFER_TIMESTAMP (buf);
- enc->granulepos_offset = gst_util_uint64_scale
- (GST_BUFFER_TIMESTAMP (buf), enc->rate, GST_SECOND);
- }
- }
-
- if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)
- && GST_BUFFER_DURATION_IS_VALID (buf))
- enc->next_ts = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
- else
- enc->next_ts = GST_CLOCK_TIME_NONE;
-
- GST_DEBUG_OBJECT (enc, "received buffer of %u bytes",
- gst_buffer_get_size (buf));
-
- /* push buffer to adapter */
- gst_adapter_push (enc->adapter, buf);
- buf = NULL;
-
- ret = gst_speex_enc_encode (enc, FALSE);
+ ret = gst_speex_enc_encode (enc, buf);
done:
-
- if (buf)
- gst_buffer_unref (buf);
-
return ret;
-
- /* ERRORS */
-not_setup:
- {
- GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
- ("encoder not initialized (input is not audio?)"));
- ret = GST_FLOW_NOT_NEGOTIATED;
- goto done;
- }
-
}
@@ -1198,54 +810,3 @@ gst_speex_enc_set_property (GObject * object, guint prop_id,
break;
}
}
-
-static GstStateChangeReturn
-gst_speex_enc_change_state (GstElement * element, GstStateChange transition)
-{
- GstSpeexEnc *enc = GST_SPEEX_ENC (element);
- GstStateChangeReturn res;
-
- switch (transition) {
- case GST_STATE_CHANGE_NULL_TO_READY:
- enc->tags = gst_tag_list_new ();
- break;
- case GST_STATE_CHANGE_READY_TO_PAUSED:
- speex_bits_init (&enc->bits);
- enc->frameno = 0;
- enc->frameno_out = 0;
- enc->samples_in = 0;
- enc->start_ts = GST_CLOCK_TIME_NONE;
- enc->next_ts = GST_CLOCK_TIME_NONE;
- enc->granulepos_offset = 0;
- break;
- case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
- /* fall through */
- default:
- break;
- }
-
- res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
- if (res == GST_STATE_CHANGE_FAILURE)
- return res;
-
- switch (transition) {
- case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
- break;
- case GST_STATE_CHANGE_PAUSED_TO_READY:
- enc->setup = FALSE;
- enc->header_sent = FALSE;
- if (enc->state) {
- speex_encoder_destroy (enc->state);
- enc->state = NULL;
- }
- speex_bits_destroy (&enc->bits);
- break;
- case GST_STATE_CHANGE_READY_TO_NULL:
- gst_tag_list_free (enc->tags);
- enc->tags = NULL;
- default:
- break;
- }
-
- return res;
-}
diff --git a/ext/speex/gstspeexenc.h b/ext/speex/gstspeexenc.h
index 6de9ebd45..17cb1e2df 100644
--- a/ext/speex/gstspeexenc.h
+++ b/ext/speex/gstspeexenc.h
@@ -23,7 +23,7 @@
#include <gst/gst.h>
-#include <gst/base/gstadapter.h>
+#include <gst/audio/gstaudioencoder.h>
#include <speex/speex.h>
#include <speex/speex_header.h>
@@ -53,14 +53,7 @@ typedef struct _GstSpeexEnc GstSpeexEnc;
typedef struct _GstSpeexEncClass GstSpeexEncClass;
struct _GstSpeexEnc {
- GstElement element;
-
- /* pads */
- GstPad *sinkpad,
- *srcpad;
-
- gint packet_count;
- gint n_packets;
+ GstAudioEncoder element;
SpeexBits bits;
SpeexHeader header;
@@ -70,9 +63,9 @@ struct _GstSpeexEnc {
const SpeexMode *speex_mode;
#endif
void *state;
- GstSpeexMode mode;
- GstAdapter *adapter;
+ /* properties */
+ GstSpeexMode mode;
gfloat quality;
gint bitrate;
gboolean vbr;
@@ -81,40 +74,24 @@ struct _GstSpeexEnc {
gboolean dtx;
gint complexity;
gint nframes;
-
- gint lookahead;
+ gchar *last_message;
gint channels;
gint rate;
- gboolean setup;
gboolean header_sent;
- guint64 samples_in;
- guint64 bytes_out;
-
GstTagList *tags;
- gchar *last_message;
-
gint frame_size;
- guint64 frameno;
- guint64 frameno_out;
+ gint lookahead;
guint8 *comments;
gint comment_len;
-
- /* Timestamp and granulepos tracking */
- GstClockTime start_ts;
- GstClockTime next_ts;
- guint64 granulepos_offset;
};
struct _GstSpeexEncClass {
- GstElementClass parent_class;
-
- /* signals */
- void (*frame_encoded) (GstElement *element);
+ GstAudioEncoderClass parent_class;
};
GType gst_speex_enc_get_type (void);