static GstCaps *
gst_speex_enc_sink_getcaps (GstPad * pad)
{
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);
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);
return caps;
}
static gboolean
gst_speex_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
{
GstSpeexEnc *enc;
enc = GST_SPEEX_ENC (benc);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_TAG:
{
if (enc->tags) {
GstTagList *list;
gst_event_parse_tag (event, &list);
gst_tag_list_insert (enc->tags, list,
gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (enc)));
} else {
g_assert_not_reached ();
}
break;
}
default:
break;
}
/* we only peeked, let base class handle it */
return GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (benc, event);
}
static GstFlowReturn
gst_speex_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
GstSpeexEnc *enc;
GstFlowReturn ret = GST_FLOW_OK;
enc = GST_SPEEX_ENC (benc);
if (!enc->header_sent) {
/* Speex streams begin with two headers; the initial header (with
most of the codec setup parameters) which is mandated by the Ogg
bitstream spec. The second header holds any comment fields.
We merely need to make the headers, then pass them to libspeex
one at a time; libspeex handles the additional Ogg bitstream
constraints */
GstBuffer *buf1, *buf2;
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_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_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_audio_encoder_set_output_format (GST_AUDIO_ENCODER (enc), caps);
gst_caps_unref (caps);
/* push out buffers */
/* store buffers for later pre_push sending */
headers = NULL;
GST_DEBUG_OBJECT (enc, "storing header buffers");
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 %" G_GSIZE_FORMAT " bytes", buf,
buf ? gst_buffer_get_size (buf) : 0);
ret = gst_speex_enc_encode (enc, buf);
return ret;
}
static void
gst_speex_enc_finalize (GObject * object)
{
GstSpeexEnc *enc;
enc = GST_SPEEX_ENC (object);
g_free (enc->last_message);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_speex_enc_start (GstAudioEncoder * benc)
{
GstSpeexEnc *enc = GST_SPEEX_ENC (benc);
GST_DEBUG_OBJECT (enc, "start");
speex_bits_init (&enc->bits);
enc->tags = gst_tag_list_new_empty ();
enc->header_sent = FALSE;
return TRUE;
}
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;
}
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 gboolean
gst_speex_enc_stop (GstAudioEncoder * benc)
{
GstSpeexEnc *enc = GST_SPEEX_ENC (benc);
GST_DEBUG_OBJECT (enc, "stop");
enc->header_sent = FALSE;
if (enc->state) {
speex_encoder_destroy (enc->state);
enc->state = NULL;
}
speex_bits_destroy (&enc->bits);
gst_tag_list_free (enc->tags);
enc->tags = NULL;
return TRUE;
}
static void
gst_speex_enc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstSpeexEnc *enc;
enc = GST_SPEEX_ENC (object);
switch (prop_id) {
case PROP_QUALITY:
g_value_set_float (value, enc->quality);
break;
case PROP_BITRATE:
g_value_set_int (value, enc->bitrate);
break;
case PROP_MODE:
g_value_set_enum (value, enc->mode);
break;
case PROP_VBR:
g_value_set_boolean (value, enc->vbr);
break;
case PROP_ABR:
g_value_set_int (value, enc->abr);
break;
case PROP_VAD:
g_value_set_boolean (value, enc->vad);
break;
case PROP_DTX:
g_value_set_boolean (value, enc->dtx);
break;
case PROP_COMPLEXITY:
g_value_set_int (value, enc->complexity);
break;
case PROP_NFRAMES:
g_value_set_int (value, enc->nframes);
break;
case PROP_LAST_MESSAGE:
g_value_set_string (value, enc->last_message);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_speex_enc_stop (GstAudioEncoder * benc)
{
GstSpeexEnc *enc = GST_SPEEX_ENC (benc);
GST_DEBUG_OBJECT (enc, "stop");
enc->header_sent = FALSE;
if (enc->state) {
speex_encoder_destroy (enc->state);
enc->state = NULL;
}
speex_bits_destroy (&enc->bits);
speex_bits_set_bit_buffer (&enc->bits, NULL, 0);
gst_tag_list_unref (enc->tags);
enc->tags = NULL;
gst_tag_setter_reset_tags (GST_TAG_SETTER (enc));
return TRUE;
}
static void
gst_speex_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstSpeexEnc *enc;
enc = GST_SPEEX_ENC (object);
switch (prop_id) {
case PROP_QUALITY:
enc->quality = g_value_get_float (value);
break;
case PROP_BITRATE:
enc->bitrate = g_value_get_int (value);
break;
case PROP_MODE:
enc->mode = g_value_get_enum (value);
break;
case PROP_VBR:
enc->vbr = g_value_get_boolean (value);
break;
case PROP_ABR:
enc->abr = g_value_get_int (value);
break;
case PROP_VAD:
enc->vad = g_value_get_boolean (value);
break;
case PROP_DTX:
enc->dtx = g_value_get_boolean (value);
break;
case PROP_COMPLEXITY:
enc->complexity = g_value_get_int (value);
break;
case PROP_NFRAMES:
enc->nframes = g_value_get_int (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_speex_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
GstSpeexEnc *enc;
enc = GST_SPEEX_ENC (benc);
enc->channels = GST_AUDIO_INFO_CHANNELS (info);
enc->rate = GST_AUDIO_INFO_RATE (info);
/* handle reconfigure */
if (enc->state) {
speex_encoder_destroy (enc->state);
enc->state = NULL;
}
if (!gst_speex_enc_setup (enc))
return FALSE;
/* 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);
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);
}
return TRUE;
}
static gboolean
gst_speex_enc_convert_src (GstPad * pad, GstFormat src_format, gint64 src_value,
GstFormat * dest_format, gint64 * dest_value)
{
gboolean res = TRUE;
GstSpeexEnc *enc;
gint64 avg;
enc = GST_SPEEX_ENC (GST_PAD_PARENT (pad));
if (enc->samples_in == 0 || enc->bytes_out == 0 || enc->rate == 0)
return FALSE;
avg = (enc->bytes_out * enc->rate) / (enc->samples_in);
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;
}
static gboolean
gst_speex_enc_sinkevent (GstPad * pad, GstEvent * event)
{
gboolean res = TRUE;
GstSpeexEnc *enc;
enc = GST_SPEEX_ENC (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
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) {
GstTagList *list;
gst_event_parse_tag (event, &list);
gst_tag_list_insert (enc->tags, list,
gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (enc)));
} 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;
}
static GstFlowReturn
gst_speex_enc_chain (GstPad * pad, GstBuffer * buf)
{
GstSpeexEnc *enc;
GstFlowReturn ret = GST_FLOW_OK;
enc = GST_SPEEX_ENC (GST_PAD_PARENT (pad));
if (!enc->setup)
goto not_setup;
if (!enc->header_sent) {
/* Speex streams begin with two headers; the initial header (with
most of the codec setup parameters) which is mandated by the Ogg
bitstream spec. The second header holds any comment fields.
We merely need to make the headers, then pass them to libspeex
one at a time; libspeex handles the additional Ogg bitstream
constraints */
GstBuffer *buf1, *buf2;
GstCaps *caps;
guchar *data;
gint data_len;
/* 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);
/* create comment buffer */
buf2 = gst_speex_enc_create_metadata_buffer (enc);
/* mark and put on caps */
caps = gst_pad_get_caps (enc->srcpad);
caps = gst_speex_enc_set_header_on_caps (caps, buf1, buf2);
gst_caps_set_simple (caps,
"rate", G_TYPE_INT, enc->rate,
"channels", G_TYPE_INT, enc->channels, 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_buffer_set_caps (buf1, caps);
gst_buffer_set_caps (buf2, caps);
gst_caps_unref (caps);
/* push out buffers */
ret = gst_speex_enc_push_buffer (enc, buf1);
if (ret != GST_FLOW_OK) {
gst_buffer_unref (buf2);
goto done;
}
ret = gst_speex_enc_push_buffer (enc, buf2);
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 (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_SIZE (buf)) {
gst_buffer_unref (buf);
return GST_FLOW_OK;
}
buf = gst_buffer_make_metadata_writable (buf);
GST_BUFFER_DATA (buf) += diff_bytes;
GST_BUFFER_SIZE (buf) -= diff_bytes;
GST_BUFFER_TIMESTAMP (buf) += diff;
if (GST_BUFFER_DURATION_IS_VALID (buf))
GST_BUFFER_DURATION (buf) -= diff;
}
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_SIZE (buf));
/* push buffer to adapter */
gst_adapter_push (enc->adapter, buf);
buf = NULL;
ret = gst_speex_enc_encode (enc, FALSE);
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;
}
}
static gboolean
gst_speex_enc_src_query (GstPad * pad, GstQuery * query)
{
gboolean res = TRUE;
GstSpeexEnc *enc;
enc = GST_SPEEX_ENC (gst_pad_get_parent (pad));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
GstFormat fmt, 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;
}
fmt = GST_FORMAT_TIME;
if (!(res = gst_pad_query_peer_position (enc->sinkpad, &fmt, &pos)))
break;
if ((res =
gst_pad_query_peer_convert (enc->sinkpad, fmt, pos, &req_fmt,
&val)))
gst_query_set_position (query, req_fmt, val);
break;
}
case GST_QUERY_DURATION:
{
GstFormat fmt, 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;
}
fmt = GST_FORMAT_TIME;
if (!(res = gst_pad_query_peer_duration (enc->sinkpad, &fmt, &dur)))
break;
if ((res =
gst_pad_query_peer_convert (enc->sinkpad, fmt, 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;
}
error:
gst_object_unref (enc);
return res;
}
static gboolean
gst_speex_enc_convert_sink (GstPad * pad, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
{
gboolean res = TRUE;
guint scale = 1;
gint bytes_per_sample;
GstSpeexEnc *enc;
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;
}
return res;
}
static GstFlowReturn
gst_speex_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
GstSpeexEnc *enc;
GstFlowReturn ret = GST_FLOW_OK;
enc = GST_SPEEX_ENC (benc);
if (!enc->header_sent) {
/* Speex streams begin with two headers; the initial header (with
most of the codec setup parameters) which is mandated by the Ogg
bitstream spec. The second header holds any comment fields.
We merely need to make the headers, then pass them to libspeex
one at a time; libspeex handles the additional Ogg bitstream
constraints */
GstBuffer *buf1, *buf2;
GstCaps *caps;
guchar *data;
gint data_len;
/* 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;
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_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_buffer_set_caps (buf1, caps);
gst_buffer_set_caps (buf2, caps);
gst_caps_unref (caps);
/* push out buffers */
ret = gst_speex_enc_push_buffer (enc, buf1);
if (ret != GST_FLOW_OK) {
gst_buffer_unref (buf2);
goto done;
}
ret = gst_speex_enc_push_buffer (enc, buf2);
if (ret != GST_FLOW_OK)
goto done;
enc->header_sent = TRUE;
}
GST_DEBUG_OBJECT (enc, "received buffer %p of %u bytes", buf,
buf ? GST_BUFFER_SIZE (buf) : 0);
ret = gst_speex_enc_encode (enc, buf);
done:
return ret;
}
