static GstFlowReturn
gst_iir_equalizer_transform_ip (GstBaseTransform * btrans, GstBuffer * buf)
{
GstAudioFilter *filter = GST_AUDIO_FILTER (btrans);
GstIirEqualizer *equ = GST_IIR_EQUALIZER (btrans);
GstClockTime timestamp;
GstMapInfo map;
gint channels = GST_AUDIO_FILTER_CHANNELS (filter);
gboolean need_new_coefficients;
if (G_UNLIKELY (channels < 1 || equ->process == NULL))
return GST_FLOW_NOT_NEGOTIATED;
BANDS_LOCK (equ);
need_new_coefficients = equ->need_new_coefficients;
BANDS_UNLOCK (equ);
if (!need_new_coefficients && gst_base_transform_is_passthrough (btrans))
return GST_FLOW_OK;
timestamp = GST_BUFFER_TIMESTAMP (buf);
timestamp =
gst_segment_to_stream_time (&btrans->segment, GST_FORMAT_TIME, timestamp);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
GstIirEqualizerBand **filters = equ->bands;
guint f, nf = equ->freq_band_count;
gst_object_sync_values (GST_OBJECT (equ), timestamp);
/* sync values for bands too */
/* FIXME: iterating equ->bands is not thread-safe here */
for (f = 0; f < nf; f++) {
gst_object_sync_values (GST_OBJECT (filters[f]), timestamp);
}
}
BANDS_LOCK (equ);
if (need_new_coefficients) {
update_coefficients (equ);
set_passthrough (equ);
}
BANDS_UNLOCK (equ);
gst_buffer_map (buf, &map, GST_MAP_READWRITE);
equ->process (equ, map.data, map.size, channels);
gst_buffer_unmap (buf, &map);
return GST_FLOW_OK;
}
static void
gst_gdk_pixbuf_overlay_before_transform (GstBaseTransform * trans,
GstBuffer * outbuf)
{
GstClockTime stream_time;
GstGdkPixbufOverlay *overlay = GST_GDK_PIXBUF_OVERLAY (trans);
gboolean set_passthrough = FALSE;
stream_time = gst_segment_to_stream_time (&trans->segment, GST_FORMAT_TIME,
GST_BUFFER_TIMESTAMP (outbuf));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (trans), stream_time);
/* now properties have been sync'ed; maybe need to update composition */
GST_OBJECT_LOCK (overlay);
if (G_UNLIKELY (overlay->update_composition)) {
gst_gdk_pixbuf_overlay_update_composition (overlay);
overlay->update_composition = FALSE;
set_passthrough = TRUE;
}
GST_OBJECT_UNLOCK (overlay);
/* determine passthrough mode so the buffer is writable if needed
* when passed into _transform_ip */
if (G_UNLIKELY (set_passthrough))
gst_base_transform_set_passthrough (trans, overlay->comp == NULL);
}
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_amplify_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base);
guint num_samples;
GstClockTime timestamp, stream_time;
timestamp = GST_BUFFER_TIMESTAMP (buf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (G_OBJECT (filter), stream_time);
num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
if (gst_base_transform_is_passthrough (base) ||
G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
return GST_FLOW_OK;
filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
return GST_FLOW_OK;
}
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_invert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioInvert *filter = GST_AUDIO_INVERT (base);
guint num_samples;
GstClockTime timestamp, stream_time;
GstMapInfo map;
timestamp = GST_BUFFER_TIMESTAMP (buf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (filter), stream_time);
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
return GST_FLOW_OK;
gst_buffer_map (buf, &map, GST_MAP_READWRITE);
num_samples = map.size / GST_AUDIO_FILTER_BPS (filter);
filter->process (filter, map.data, num_samples);
gst_buffer_unmap (buf, &map);
return GST_FLOW_OK;
}
static GstFlowReturn
gst_iir_equalizer_transform_ip (GstBaseTransform * btrans, GstBuffer * buf)
{
GstAudioFilter *filter = GST_AUDIO_FILTER (btrans);
GstIirEqualizer *equ = GST_IIR_EQUALIZER (btrans);
GstClockTime timestamp;
if (G_UNLIKELY (filter->format.channels < 1 || equ->process == NULL))
return GST_FLOW_NOT_NEGOTIATED;
if (equ->need_new_coefficients) {
update_coefficients (equ);
set_passthrough (equ);
}
if (gst_base_transform_is_passthrough (btrans))
return GST_FLOW_OK;
timestamp = GST_BUFFER_TIMESTAMP (buf);
timestamp =
gst_segment_to_stream_time (&btrans->segment, GST_FORMAT_TIME, timestamp);
if (GST_CLOCK_TIME_IS_VALID (timestamp))
gst_object_sync_values (G_OBJECT (equ), timestamp);
equ->process (equ, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
filter->format.channels);
return GST_FLOW_OK;
}
static GstFlowReturn
gst_ladspa_sink_type_render (GstBaseSink * base, GstBuffer * buf)
{
GstLADSPASink *ladspa = GST_LADSPA_SINK (base);
GstMapInfo info;
if (ladspa->num_buffers_left == 0)
goto eos;
if (ladspa->num_buffers_left != -1)
ladspa->num_buffers_left--;
gst_object_sync_values (GST_OBJECT (ladspa), GST_BUFFER_TIMESTAMP (buf));
gst_buffer_map (buf, &info, GST_MAP_READ);
gst_ladspa_transform (&ladspa->ladspa, NULL,
info.size / sizeof (LADSPA_Data) / ladspa->ladspa.klass->count.audio.in,
info.data);
gst_buffer_unmap (buf, &info);
if (ladspa->num_buffers_left == 0)
goto eos;
return GST_FLOW_OK;
/* ERRORS */
eos:
{
GST_DEBUG_OBJECT (ladspa, "we are EOS");
return GST_FLOW_EOS;
}
}
static GstFlowReturn
gst_quarktv_transform_frame (GstVideoFilter * vfilter, GstVideoFrame * in_frame,
GstVideoFrame * out_frame)
{
GstQuarkTV *filter = GST_QUARKTV (vfilter);
gint area;
guint32 *src, *dest;
GstClockTime timestamp;
GstBuffer **planetable;
gint planes, current_plane;
timestamp = GST_BUFFER_TIMESTAMP (in_frame->buffer);
timestamp =
gst_segment_to_stream_time (&GST_BASE_TRANSFORM (vfilter)->segment,
GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (timestamp))
gst_object_sync_values (GST_OBJECT (filter), timestamp);
if (G_UNLIKELY (filter->planetable == NULL))
return GST_FLOW_FLUSHING;
src = GST_VIDEO_FRAME_PLANE_DATA (in_frame, 0);
dest = GST_VIDEO_FRAME_PLANE_DATA (out_frame, 0);
GST_OBJECT_LOCK (filter);
area = filter->area;
planetable = filter->planetable;
planes = filter->planes;
current_plane = filter->current_plane;
if (planetable[current_plane])
gst_buffer_unref (planetable[current_plane]);
planetable[current_plane] = gst_buffer_ref (in_frame->buffer);
/* For each pixel */
while (--area) {
GstBuffer *rand;
/* pick a random buffer */
rand = planetable[(current_plane + (fastrand () >> 24)) % planes];
/* Copy the pixel from the random buffer to dest, FIXME, slow */
if (rand)
gst_buffer_extract (rand, area * 4, &dest[area], 4);
else
dest[area] = src[area];
}
filter->current_plane--;
if (filter->current_plane < 0)
filter->current_plane = planes - 1;
GST_OBJECT_UNLOCK (filter);
return GST_FLOW_OK;
}
static GstFlowReturn
gst_revtv_transform (GstBaseTransform * trans, GstBuffer * in, GstBuffer * out)
{
GstRevTV *filter = GST_REVTV (trans);
guint32 *src, *dest;
gint width, height;
guint32 *nsrc;
gint y, x, R, G, B, yval;
GstFlowReturn ret = GST_FLOW_OK;
gint linespace, vscale;
GstClockTime timestamp, stream_time;
timestamp = GST_BUFFER_TIMESTAMP (in);
stream_time =
gst_segment_to_stream_time (&trans->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (G_OBJECT (filter), stream_time);
src = (guint32 *) GST_BUFFER_DATA (in);
dest = (guint32 *) GST_BUFFER_DATA (out);
GST_OBJECT_LOCK (filter);
width = filter->width;
height = filter->height;
/* Clear everything to black */
memset (dest, 0, width * height * sizeof (guint32));
linespace = filter->linespace;
vscale = filter->vscale;
/* draw the offset lines */
for (y = 0; y < height; y += linespace) {
for (x = 0; x <= width; x++) {
nsrc = src + (y * width) + x;
/* Calc Y Value for curpix */
R = ((*nsrc) & 0xff0000) >> (16 - 1);
G = ((*nsrc) & 0xff00) >> (8 - 2);
B = (*nsrc) & 0xff;
yval = y - ((short) (R + G + B) / vscale);
if (yval > 0) {
dest[x + (yval * width)] = THE_COLOR;
}
}
}
GST_OBJECT_UNLOCK (filter);
return ret;
}
static GstFlowReturn
gst_quarktv_transform (GstBaseTransform * trans, GstBuffer * in,
GstBuffer * out)
{
GstQuarkTV *filter = GST_QUARKTV (trans);
gint area;
guint32 *src, *dest;
GstFlowReturn ret = GST_FLOW_OK;
GstClockTime timestamp;
GstBuffer **planetable;
gint planes, current_plane;
timestamp = GST_BUFFER_TIMESTAMP (in);
timestamp =
gst_segment_to_stream_time (&trans->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (timestamp))
gst_object_sync_values (G_OBJECT (filter), timestamp);
if (G_UNLIKELY (filter->planetable == NULL))
return GST_FLOW_WRONG_STATE;
GST_OBJECT_LOCK (filter);
area = filter->area;
src = (guint32 *) GST_BUFFER_DATA (in);
dest = (guint32 *) GST_BUFFER_DATA (out);
planetable = filter->planetable;
planes = filter->planes;
current_plane = filter->current_plane;
if (planetable[current_plane])
gst_buffer_unref (planetable[current_plane]);
planetable[current_plane] = gst_buffer_ref (in);
/* For each pixel */
while (--area) {
GstBuffer *rand;
/* pick a random buffer */
rand = planetable[(current_plane + (fastrand () >> 24)) % planes];
/* Copy the pixel from the random buffer to dest */
dest[area] =
(rand ? ((guint32 *) GST_BUFFER_DATA (rand))[area] : src[area]);
}
filter->current_plane--;
if (filter->current_plane < 0)
filter->current_plane = planes - 1;
GST_OBJECT_UNLOCK (filter);
return ret;
}
static void
gst_gdk_pixbuf_overlay_before_transform (GstBaseTransform * trans,
GstBuffer * outbuf)
{
GstClockTime stream_time;
stream_time = gst_segment_to_stream_time (&trans->segment, GST_FORMAT_TIME,
GST_BUFFER_TIMESTAMP (outbuf));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (trans), stream_time);
}
/* start and stop are not symmetric -- start will open the device, but not start
* capture. it's setcaps that will start capture, which is called via basesrc's
* negotiate method. stop will both stop capture and close the device.
*/
static gboolean
gst_v4l2src_start (GstBaseSrc * src)
{
GstV4l2Src *v4l2src = GST_V4L2SRC (src);
v4l2src->offset = 0;
/* activate settings for first frame */
v4l2src->ctrl_time = 0;
gst_object_sync_values (GST_OBJECT (src), v4l2src->ctrl_time);
return TRUE;
}
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_echo_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioEcho *self = GST_AUDIO_ECHO (base);
guint num_samples;
GstClockTime timestamp, stream_time;
GstMapInfo map;
g_mutex_lock (&self->lock);
timestamp = GST_BUFFER_TIMESTAMP (buf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (self), stream_time);
if (self->buffer == NULL) {
guint bpf, rate;
bpf = GST_AUDIO_FILTER_BPF (self);
rate = GST_AUDIO_FILTER_RATE (self);
self->delay_frames =
MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1);
self->buffer_size_frames =
MAX (gst_util_uint64_scale (self->max_delay, rate, GST_SECOND), 1);
self->buffer_size = self->buffer_size_frames * bpf;
self->buffer = g_try_malloc0 (self->buffer_size);
self->buffer_pos = 0;
if (self->buffer == NULL) {
g_mutex_unlock (&self->lock);
GST_ERROR_OBJECT (self, "Failed to allocate %u bytes", self->buffer_size);
return GST_FLOW_ERROR;
}
}
gst_buffer_map (buf, &map, GST_MAP_READWRITE);
num_samples = map.size / GST_AUDIO_FILTER_BPS (self);
self->process (self, map.data, num_samples);
gst_buffer_unmap (buf, &map);
g_mutex_unlock (&self->lock);
return GST_FLOW_OK;
}
static GstFlowReturn
gst_webvtt_enc_chain (GstPad * pad, GstBuffer * buf)
{
GstWebvttEnc *webvttenc;
GstBuffer *new_buffer;
gchar *timing;
GstFlowReturn ret;
webvttenc = GST_WEBVTT_ENC (gst_pad_get_parent_element (pad));
if (!webvttenc->pushed_header) {
const char *header = "WEBVTT\n\n";
new_buffer = gst_buffer_new_and_alloc (strlen (header));
memcpy (GST_BUFFER_DATA (new_buffer), header, strlen (header));
GST_BUFFER_TIMESTAMP (new_buffer) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (new_buffer) = GST_CLOCK_TIME_NONE;
ret = gst_pad_push (webvttenc->srcpad, new_buffer);
if (ret != GST_FLOW_OK) {
goto out;
}
webvttenc->pushed_header = TRUE;
}
gst_object_sync_values (G_OBJECT (webvttenc), GST_BUFFER_TIMESTAMP (buf));
timing = gst_webvtt_enc_timeconvertion (webvttenc, buf);
new_buffer =
gst_buffer_new_and_alloc (strlen (timing) + GST_BUFFER_SIZE (buf) + 1);
memcpy (GST_BUFFER_DATA (new_buffer), timing, strlen (timing));
memcpy (GST_BUFFER_DATA (new_buffer) + strlen (timing), GST_BUFFER_DATA (buf),
GST_BUFFER_SIZE (buf));
memcpy (GST_BUFFER_DATA (new_buffer) + GST_BUFFER_SIZE (new_buffer) - 1,
"\n", 1);
g_free (timing);
GST_BUFFER_TIMESTAMP (new_buffer) = GST_BUFFER_TIMESTAMP (buf);
GST_BUFFER_DURATION (new_buffer) = GST_BUFFER_DURATION (buf);
ret = gst_pad_push (webvttenc->srcpad, new_buffer);
out:
gst_buffer_unref (buf);
gst_object_unref (webvttenc);
return ret;
}
static GstFlowReturn
gst_gaussianblur_transform_frame (GstVideoFilter * vfilter,
GstVideoFrame * in_frame, GstVideoFrame * out_frame)
{
GstGaussianBlur *filter = GST_GAUSSIANBLUR (vfilter);
GstClockTime timestamp;
gint64 stream_time;
gfloat sigma;
guint8 *src, *dest;
/* GstController: update the properties */
timestamp = GST_BUFFER_TIMESTAMP (in_frame->buffer);
stream_time =
gst_segment_to_stream_time (&GST_BASE_TRANSFORM (filter)->segment,
GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (filter), stream_time);
GST_OBJECT_LOCK (filter);
sigma = filter->sigma;
GST_OBJECT_UNLOCK (filter);
if (filter->cur_sigma != sigma) {
g_free (filter->kernel);
filter->kernel = NULL;
g_free (filter->kernel_sum);
filter->kernel_sum = NULL;
filter->cur_sigma = sigma;
}
if (filter->kernel == NULL &&
!make_gaussian_kernel (filter, filter->cur_sigma)) {
GST_ELEMENT_ERROR (filter, RESOURCE, NO_SPACE_LEFT, ("Out of memory"),
("Failed to allocation gaussian kernel"));
return GST_FLOW_ERROR;
}
/*
* Perform gaussian smoothing on the image using the input standard
* deviation.
*/
src = GST_VIDEO_FRAME_COMP_DATA (in_frame, 0);
dest = GST_VIDEO_FRAME_COMP_DATA (out_frame, 0);
gst_video_frame_copy (out_frame, in_frame);
gaussian_smooth (filter, src, dest);
return GST_FLOW_OK;
}
static GstFlowReturn
gst_srt_enc_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstSrtEnc *srtenc = GST_SRT_ENC (parent);
GstClockTime ts, dur = GST_SECOND;
GstBuffer *new_buffer;
GstMapInfo map_info;
GString *s;
gsize buf_size;
gst_object_sync_values (GST_OBJECT (srtenc), GST_BUFFER_PTS (buf));
ts = GST_BUFFER_PTS (buf) + srtenc->timestamp;
if (GST_BUFFER_DURATION_IS_VALID (buf))
dur = GST_BUFFER_DURATION (buf) + srtenc->duration;
else if (srtenc->duration > 0)
dur = srtenc->duration;
else
dur = GST_SECOND;
buf_size = gst_buffer_get_size (buf);
s = g_string_sized_new (10 + 50 + buf_size + 2 + 1);
/* stanza count */
g_string_append_printf (s, "%d\n", srtenc->counter++);
/* start_time --> end_time */
gst_srt_enc_append_timestamp_to_string (ts, s);
g_string_append_printf (s, " --> ");
gst_srt_enc_append_timestamp_to_string (ts + dur, s);
g_string_append_c (s, '\n');
/* text */
if (gst_buffer_map (buf, &map_info, GST_MAP_READ)) {
g_string_append_len (s, (const gchar *) map_info.data, map_info.size);
gst_buffer_unmap (buf, &map_info);
}
g_string_append (s, "\n\n");
buf_size = s->len;
new_buffer = gst_buffer_new_wrapped (g_string_free (s, FALSE), buf_size);
GST_BUFFER_TIMESTAMP (new_buffer) = GST_BUFFER_TIMESTAMP (buf);
GST_BUFFER_DURATION (new_buffer) = GST_BUFFER_DURATION (buf);
gst_buffer_unref (buf);
return gst_pad_push (srtenc->srcpad, new_buffer);
}
/* this function does the actual processing
*/
static GstFlowReturn
gst_freeverb_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstFreeverb *filter = GST_FREEVERB (base);
guint num_samples;
GstClockTime timestamp;
GstMapInfo inmap, outmap;
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
timestamp =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
num_samples = outmap.size / (2 * GST_AUDIO_INFO_BPS (&filter->info));
GST_DEBUG_OBJECT (filter, "processing %u samples at %" GST_TIME_FORMAT,
num_samples, GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (timestamp))
gst_object_sync_values (GST_OBJECT (filter), timestamp);
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DISCONT))) {
filter->drained = FALSE;
}
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP))) {
if (filter->drained) {
memset (outmap.data, 0, outmap.size);
}
} else {
filter->drained = FALSE;
}
if (!filter->drained) {
filter->drained =
filter->process (filter, inmap.data, outmap.data, num_samples);
}
if (filter->drained) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
}
gst_buffer_unmap (inbuf, &inmap);
gst_buffer_unmap (outbuf, &outmap);
return GST_FLOW_OK;
}
static void
gst_gamma_before_transform (GstBaseTransform * base, GstBuffer * outbuf)
{
GstGamma *gamma = GST_GAMMA (base);
GstClockTime timestamp, stream_time;
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (gamma, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (gamma), stream_time);
}
static void
gst_geometric_transform_before_transform (GstBaseTransform * trans,
GstBuffer * outbuf)
{
GstGeometricTransform *gt = GST_GEOMETRIC_TRANSFORM_CAST (trans);
GstClockTime timestamp, stream_time;
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
stream_time =
gst_segment_to_stream_time (&trans->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (gt, "sync to %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (G_OBJECT (gt), stream_time);
}
/* this function does the actual processing
*/
static GstFlowReturn
gst_plugin_template_transform_ip (GstBaseTransform * base, GstBuffer * outbuf)
{
GstPluginTemplate *filter = GST_PLUGIN_TEMPLATE (base);
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (outbuf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (outbuf));
if (filter->silent == FALSE)
g_print ("I'm plugged, therefore I'm in.\n");
/* FIXME: do something interesting here. This simply copies the source
* to the destination. */
return GST_FLOW_OK;
}
static void
gst_video_flip_before_transform (GstBaseTransform * trans, GstBuffer * in)
{
GstVideoFlip *videoflip = GST_VIDEO_FLIP (trans);
GstClockTime timestamp, stream_time;
timestamp = GST_BUFFER_TIMESTAMP (in);
stream_time =
gst_segment_to_stream_time (&trans->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (videoflip, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (G_OBJECT (videoflip), stream_time);
}
static void
gst_video_balance_before_transform (GstBaseTransform * base, GstBuffer * buf)
{
GstVideoBalance *balance = GST_VIDEO_BALANCE (base);
GstClockTime timestamp, stream_time;
timestamp = GST_BUFFER_TIMESTAMP (buf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (balance, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (balance), stream_time);
}
static GstFlowReturn
gst_stereo_transform_ip (GstBaseTransform * base, GstBuffer * outbuf)
{
GstStereo *stereo = GST_STEREO (base);
gint samples;
gint i;
gdouble avg, ldiff, rdiff, tmp;
gdouble mul = stereo->stereo;
gint16 *data;
GstMapInfo info;
if (!gst_buffer_map (outbuf, &info, GST_MAP_READWRITE))
return GST_FLOW_ERROR;
data = (gint16 *) info.data;
samples = info.size / 2;
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (outbuf)))
gst_object_sync_values (GST_OBJECT (stereo), GST_BUFFER_TIMESTAMP (outbuf));
if (stereo->active) {
for (i = 0; i < samples / 2; i += 2) {
avg = (data[i] + data[i + 1]) / 2;
ldiff = data[i] - avg;
rdiff = data[i + 1] - avg;
tmp = avg + ldiff * mul;
if (tmp < -32768)
tmp = -32768;
if (tmp > 32767)
tmp = 32767;
data[i] = tmp;
tmp = avg + rdiff * mul;
if (tmp < -32768)
tmp = -32768;
if (tmp > 32767)
tmp = 32767;
data[i + 1] = tmp;
}
}
gst_buffer_unmap (outbuf, &info);
return GST_FLOW_OK;
}
/* start and stop are not symmetric -- start will open the device, but not start
* capture. it's setcaps that will start capture, which is called via basesrc's
* negotiate method. stop will both stop capture and close the device.
*/
static gboolean
gst_v4l2src_start (GstBaseSrc * src)
{
GstV4l2Src *v4l2src = GST_V4L2SRC (src);
v4l2src->offset = 0;
v4l2src->renegotiation_adjust = 0;
/* activate settings for first frame */
v4l2src->ctrl_time = 0;
gst_object_sync_values (GST_OBJECT (src), v4l2src->ctrl_time);
v4l2src->has_bad_timestamp = FALSE;
v4l2src->last_timestamp = 0;
return TRUE;
}
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_echo_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioEcho *self = GST_AUDIO_ECHO (base);
guint num_samples;
GstClockTime timestamp, stream_time;
timestamp = GST_BUFFER_TIMESTAMP (buf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (G_OBJECT (self), stream_time);
num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (self)->format.width / 8);
if (self->buffer == NULL) {
guint width, rate, channels;
width = GST_AUDIO_FILTER (self)->format.width / 8;
rate = GST_AUDIO_FILTER (self)->format.rate;
channels = GST_AUDIO_FILTER (self)->format.channels;
self->delay_frames =
MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1);
self->buffer_size_frames =
MAX (gst_util_uint64_scale (self->max_delay, rate, GST_SECOND), 1);
self->buffer_size = self->buffer_size_frames * width * channels;
self->buffer = g_try_malloc0 (self->buffer_size);
self->buffer_pos = 0;
if (self->buffer == NULL) {
GST_ERROR_OBJECT (self, "Failed to allocate %u bytes", self->buffer_size);
return GST_FLOW_ERROR;
}
}
self->process (self, GST_BUFFER_DATA (buf), num_samples);
return GST_FLOW_OK;
}
static void
gst_smpte_alpha_before_transform (GstBaseTransform * trans, GstBuffer * buf)
{
GstSMPTEAlpha *smpte = GST_SMPTE_ALPHA (trans);
GstClockTime timestamp, stream_time;
/* first sync the controller to the current stream_time of the buffer */
timestamp = GST_BUFFER_TIMESTAMP (buf);
stream_time =
gst_segment_to_stream_time (&trans->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (smpte, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (smpte), stream_time);
}
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_amplify_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base);
guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
if (gst_base_transform_is_passthrough (base) ||
G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
return GST_FLOW_OK;
filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
return GST_FLOW_OK;
}
static GstFlowReturn
gauss_blur_process_frame (GstBaseTransform * btrans,
GstBuffer * in_buf, GstBuffer * out_buf)
{
GaussBlur *gb = GAUSS_BLUR (btrans);
GstClockTime timestamp;
gint64 stream_time;
gfloat sigma;
/* GstController: update the properties */
timestamp = GST_BUFFER_TIMESTAMP (in_buf);
stream_time =
gst_segment_to_stream_time (&btrans->segment, GST_FORMAT_TIME, timestamp);
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (G_OBJECT (gb), stream_time);
GST_OBJECT_LOCK (gb);
sigma = gb->sigma;
GST_OBJECT_UNLOCK (gb);
if (gb->cur_sigma != sigma) {
g_free (gb->kernel);
gb->kernel = NULL;
g_free (gb->kernel_sum);
gb->kernel_sum = NULL;
gb->cur_sigma = sigma;
}
if (gb->kernel == NULL && !make_gaussian_kernel (gb, gb->cur_sigma)) {
GST_ELEMENT_ERROR (btrans, RESOURCE, NO_SPACE_LEFT, ("Out of memory"),
("Failed to allocation gaussian kernel"));
return GST_FLOW_ERROR;
}
/*
* Perform gaussian smoothing on the image using the input standard
* deviation.
*/
memcpy (GST_BUFFER_DATA (out_buf), GST_BUFFER_DATA (in_buf),
gb->height * gb->stride);
gaussian_smooth (gb, GST_BUFFER_DATA (in_buf), GST_BUFFER_DATA (out_buf));
return GST_FLOW_OK;
}
/* this function does the actual processing
*/
static GstFlowReturn
gst_audio_panorama_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstAudioPanorama *filter = GST_AUDIO_PANORAMA (base);
guint num_samples = GST_BUFFER_SIZE (outbuf) / (2 * filter->width);
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (outbuf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (outbuf));
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP))) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
memset (GST_BUFFER_DATA (outbuf), 0, GST_BUFFER_SIZE (outbuf));
return GST_FLOW_OK;
}
filter->process (filter, GST_BUFFER_DATA (inbuf),
GST_BUFFER_DATA (outbuf), num_samples);
return GST_FLOW_OK;
}
static GstFlowReturn
gst_stereo_transform_ip (GstBaseTransform * base, GstBuffer * outbuf)
{
GstStereo *stereo = GST_STEREO (base);
gint16 *data = (gint16 *) GST_BUFFER_DATA (outbuf);
gint samples = GST_BUFFER_SIZE (outbuf) / 2;
gint i;
gdouble avg, ldiff, rdiff, tmp;
gdouble mul = stereo->stereo;
if (!gst_buffer_is_writable (outbuf))
return GST_FLOW_OK;
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (outbuf)))
gst_object_sync_values (GST_OBJECT (stereo), GST_BUFFER_TIMESTAMP (outbuf));
if (stereo->active) {
for (i = 0; i < samples / 2; i += 2) {
avg = (data[i] + data[i + 1]) / 2;
ldiff = data[i] - avg;
rdiff = data[i + 1] - avg;
tmp = avg + ldiff * mul;
if (tmp < -32768)
tmp = -32768;
if (tmp > 32767)
tmp = 32767;
data[i] = tmp;
tmp = avg + rdiff * mul;
if (tmp < -32768)
tmp = -32768;
if (tmp > 32767)
tmp = 32767;
data[i + 1] = tmp;
}
}
return GST_FLOW_OK;
}
/* this function does the actual processing
*/
static GstFlowReturn
gst_freeverb_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstFreeverb *filter = GST_FREEVERB (base);
guint num_samples = GST_BUFFER_SIZE (outbuf) / (2 * filter->width);
GstClockTime timestamp;
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
timestamp =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "processing %u samples at %" GST_TIME_FORMAT,
num_samples, GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (timestamp))
gst_object_sync_values (G_OBJECT (filter), timestamp);
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DISCONT))) {
filter->drained = FALSE;
}
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP))) {
if (filter->drained) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
memset (GST_BUFFER_DATA (outbuf), 0, GST_BUFFER_SIZE (outbuf));
return GST_FLOW_OK;
}
} else {
filter->drained = FALSE;
}
filter->drained = filter->process (filter, GST_BUFFER_DATA (inbuf),
GST_BUFFER_DATA (outbuf), num_samples);
if (filter->drained) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
}
return GST_FLOW_OK;
}
