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authorSebastian Dröge <slomo@circular-chaos.org>2007-08-16 09:48:27 +0000
committerSebastian Dröge <slomo@circular-chaos.org>2007-08-16 09:48:27 +0000
commite38e701293f998e0268d8c13d967266cf699a3f3 (patch)
treee45374931920205d703840d4bf1a00c9b1514e78 /gst
parentb414b128c696019f3fe019b8cc798bf762c9f3bb (diff)
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gst/filter/gstlpwsinc.*: Implement latency query and only forward those samples downstream that actually contain the ...
Original commit message from CVS: * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init), (gst_lpwsinc_init), (process_32), (process_64), (lpwsinc_build_kernel), (lpwsinc_push_residue), (lpwsinc_transform), (lpwsinc_start), (lpwsinc_query), (lpwsinc_query_type), (lpwsinc_event), (lpwsinc_set_property): * gst/filter/gstlpwsinc.h: Implement latency query and only forward those samples downstream that actually contain the data we want, i.e. drop kernel_length/2 in the beginning and append kernel_length/2 (created by convolving the filter kernel with zeroes) to the end. * tests/check/elements/lpwsinc.c: (GST_START_TEST): Adjust the unit test for this slightly changed behaviour.
Diffstat (limited to 'gst')
-rw-r--r--gst/filter/gstlpwsinc.c297
-rw-r--r--gst/filter/gstlpwsinc.h4
2 files changed, 295 insertions, 6 deletions
diff --git a/gst/filter/gstlpwsinc.c b/gst/filter/gstlpwsinc.c
index c8b7beb2..d86aace1 100644
--- a/gst/filter/gstlpwsinc.c
+++ b/gst/filter/gstlpwsinc.c
@@ -171,9 +171,13 @@ static GstFlowReturn lpwsinc_transform (GstBaseTransform * base,
static gboolean lpwsinc_get_unit_size (GstBaseTransform * base, GstCaps * caps,
guint * size);
static gboolean lpwsinc_start (GstBaseTransform * base);
+static gboolean lpwsinc_event (GstBaseTransform * base, GstEvent * event);
static gboolean lpwsinc_setup (GstAudioFilter * base,
GstRingBufferSpec * format);
+static gboolean lpwsinc_query (GstPad * pad, GstQuery * query);
+static const GstQueryType *lpwsinc_query_type (GstPad * pad);
+
/* Element class */
static void
@@ -246,6 +250,7 @@ gst_lpwsinc_class_init (GstLPWSincClass * klass)
trans_class->transform = GST_DEBUG_FUNCPTR (lpwsinc_transform);
trans_class->get_unit_size = GST_DEBUG_FUNCPTR (lpwsinc_get_unit_size);
trans_class->start = GST_DEBUG_FUNCPTR (lpwsinc_start);
+ trans_class->event = GST_DEBUG_FUNCPTR (lpwsinc_event);
filter_class->setup = GST_DEBUG_FUNCPTR (lpwsinc_setup);
}
@@ -255,11 +260,19 @@ gst_lpwsinc_init (GstLPWSinc * self, GstLPWSincClass * g_class)
self->mode = MODE_LOW_PASS;
self->window = WINDOW_HAMMING;
self->kernel_length = 101;
+ self->latency = 50;
self->frequency = 0.0;
self->kernel = NULL;
self->residue = NULL;
self->have_kernel = FALSE;
+ self->residue_length = 0;
+ self->next_ts = GST_CLOCK_TIME_NONE;
+ self->next_off = GST_BUFFER_OFFSET_NONE;
+
+ gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad, lpwsinc_query);
+ gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad,
+ lpwsinc_query_type);
}
static void
@@ -296,6 +309,10 @@ process_32 (GstLPWSinc * self, gfloat * src, gfloat * dst, guint input_samples)
self->residue[i] = self->residue[i + input_samples];
for (i = res_start; i < kernel_length * channels; i++)
self->residue[i] = src[input_samples - kernel_length * channels + i];
+
+ self->residue_length += kernel_length * channels - res_start;
+ if (self->residue_length > kernel_length * channels)
+ self->residue_length = kernel_length * channels;
}
static void
@@ -333,6 +350,10 @@ process_64 (GstLPWSinc * self, gdouble * src, gdouble * dst,
self->residue[i] = self->residue[i + input_samples];
for (i = res_start; i < kernel_length * channels; i++)
self->residue[i] = src[input_samples - kernel_length * channels + i];
+
+ self->residue_length += kernel_length * channels - res_start;
+ if (self->residue_length > kernel_length * channels)
+ self->residue_length = kernel_length * channels;
}
static void
@@ -400,12 +421,88 @@ lpwsinc_build_kernel (GstLPWSinc * self)
}
/* set up the residue memory space */
- if (self->residue)
- g_free (self->residue);
- self->residue =
- g_new0 (gdouble, len * GST_AUDIO_FILTER (self)->format.channels);
+ if (!self->residue)
+ self->residue =
+ g_new0 (gdouble, len * GST_AUDIO_FILTER (self)->format.channels);
self->have_kernel = TRUE;
+ self->residue_length = 0;
+}
+
+static void
+lpwsinc_push_residue (GstLPWSinc * self)
+{
+ GstBuffer *outbuf;
+ GstFlowReturn res;
+ gint rate = GST_AUDIO_FILTER (self)->format.rate;
+ gint channels = GST_AUDIO_FILTER (self)->format.channels;
+ gint outsize, outsamples;
+ gint diffsize, diffsamples;
+ guint8 *in, *out;
+
+ /* Calculate the number of samples and their memory size that
+ * should be pushed from the residue */
+ outsamples = MIN (self->latency, self->residue_length / channels);
+ outsize = outsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
+ if (outsize == 0)
+ return;
+
+ /* Process the difference between latency and residue_length samples
+ * to start at the actual data instead of starting at the zeros before
+ * when we only got one buffer smaller than latency */
+ diffsamples = self->latency - self->residue_length / channels;
+ diffsize =
+ diffsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
+ if (diffsize > 0) {
+ in = g_new0 (guint8, diffsize);
+ out = g_new0 (guint8, diffsize);
+ self->process (self, in, out, diffsamples * channels);
+ g_free (in);
+ g_free (out);
+ }
+
+ res = gst_pad_alloc_buffer (GST_BASE_TRANSFORM (self)->srcpad,
+ GST_BUFFER_OFFSET_NONE, outsize,
+ GST_PAD_CAPS (GST_BASE_TRANSFORM (self)->srcpad), &outbuf);
+
+ if (G_UNLIKELY (res != GST_FLOW_OK)) {
+ GST_WARNING_OBJECT (self, "failed allocating buffer of %d bytes", outsize);
+ return;
+ }
+
+ /* Convolve the residue with zeros to get the actual remaining data */
+ in = g_new0 (guint8, outsize);
+ self->process (self, in, GST_BUFFER_DATA (outbuf), outsamples * channels);
+ g_free (in);
+
+ /* Set timestamp, offset, etc from the values we
+ * saved when processing the regular buffers */
+ if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
+ GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
+ else
+ GST_BUFFER_TIMESTAMP (outbuf) = 0;
+ GST_BUFFER_DURATION (outbuf) =
+ gst_util_uint64_scale (outsamples, GST_SECOND, rate);
+ self->next_ts += gst_util_uint64_scale (outsamples, GST_SECOND, rate);
+
+ if (self->next_off != GST_BUFFER_OFFSET_NONE) {
+ GST_BUFFER_OFFSET (outbuf) = self->next_off;
+ GST_BUFFER_OFFSET_END (outbuf) = self->next_off + outsamples;
+ }
+
+ GST_DEBUG_OBJECT (self, "Pushing residue buffer of size %d with timestamp: %"
+ GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld,"
+ " offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
+ GST_BUFFER_OFFSET_END (outbuf), outsamples);
+
+ res = gst_pad_push (GST_BASE_TRANSFORM (self)->srcpad, outbuf);
+
+ if (G_UNLIKELY (res != GST_FLOW_OK)) {
+ GST_WARNING_OBJECT (self, "failed to push residue");
+ }
+
}
/* GstAudioFilter vmethod implementations */
@@ -456,8 +553,12 @@ lpwsinc_transform (GstBaseTransform * base, GstBuffer * inbuf,
{
GstLPWSinc *self = GST_LPWSINC (base);
GstClockTime timestamp;
+ gint channels = GST_AUDIO_FILTER (self)->format.channels;
+ gint rate = GST_AUDIO_FILTER (self)->format.rate;
gint input_samples =
GST_BUFFER_SIZE (outbuf) / (GST_AUDIO_FILTER (self)->format.width / 8);
+ gint output_samples = input_samples;
+ gint diff;
/* don't process data in passthrough-mode */
if (gst_base_transform_is_passthrough (base))
@@ -471,9 +572,100 @@ lpwsinc_transform (GstBaseTransform * base, GstBuffer * inbuf,
if (!self->have_kernel)
lpwsinc_build_kernel (self);
+ /* Reset the residue if already existing on discont buffers */
+ if (GST_BUFFER_IS_DISCONT (inbuf)) {
+ if (channels && self->residue)
+ memset (self->residue, 0, channels *
+ self->kernel_length * sizeof (gdouble));
+ self->residue_length = 0;
+ self->next_ts = GST_CLOCK_TIME_NONE;
+ self->next_off = GST_BUFFER_OFFSET_NONE;
+ }
+
+ /* Calculate the number of samples we can push out now without outputting
+ * kernel_length/2 zeros in the beginning */
+ diff = (self->kernel_length / 2) * channels - self->residue_length;
+ if (diff > 0)
+ output_samples -= diff;
+
self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf),
input_samples);
+ if (output_samples <= 0) {
+ /* Drop buffer and save original timestamp/offset for later use */
+ if (!GST_CLOCK_TIME_IS_VALID (self->next_ts)
+ && GST_BUFFER_TIMESTAMP_IS_VALID (outbuf))
+ self->next_ts = GST_BUFFER_TIMESTAMP (outbuf);
+ if (self->next_off == GST_BUFFER_OFFSET_NONE
+ && GST_BUFFER_OFFSET_IS_VALID (outbuf))
+ self->next_off = GST_BUFFER_OFFSET (outbuf);
+ return GST_BASE_TRANSFORM_FLOW_DROPPED;
+ } else if (output_samples < input_samples) {
+ /* First (probably partial) buffer after starting from
+ * a clean residue. Use stored timestamp/offset here */
+ if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
+ GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
+
+ if (self->next_off != GST_BUFFER_OFFSET_NONE) {
+ GST_BUFFER_OFFSET (outbuf) = self->next_off;
+ if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf))
+ GST_BUFFER_OFFSET_END (outbuf) =
+ self->next_off + output_samples / channels;
+ } else {
+ /* We dropped no buffer, offset is valid, offset_end must be adjusted by diff */
+ if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf))
+ GST_BUFFER_OFFSET_END (outbuf) -= diff / channels;
+ }
+
+ if (GST_BUFFER_DURATION_IS_VALID (outbuf))
+ GST_BUFFER_DURATION (outbuf) -=
+ gst_util_uint64_scale (diff, GST_SECOND, channels * rate);
+
+ GST_BUFFER_DATA (outbuf) +=
+ diff * (GST_AUDIO_FILTER (self)->format.width / 8);
+ GST_BUFFER_SIZE (outbuf) -=
+ diff * (GST_AUDIO_FILTER (self)->format.width / 8);
+ } else {
+ GstClockTime ts_latency =
+ gst_util_uint64_scale (self->latency, GST_SECOND, rate);
+
+ /* Normal buffer, adjust timestamp/offset/etc by latency */
+ if (GST_BUFFER_TIMESTAMP (outbuf) < ts_latency) {
+ GST_WARNING_OBJECT (self, "GST_BUFFER_TIMESTAMP (outbuf) < latency");
+ GST_BUFFER_TIMESTAMP (outbuf) = 0;
+ } else {
+ GST_BUFFER_TIMESTAMP (outbuf) -= ts_latency;
+ }
+
+ if (GST_BUFFER_OFFSET_IS_VALID (outbuf)) {
+ if (GST_BUFFER_OFFSET (outbuf) > self->latency) {
+ GST_BUFFER_OFFSET (outbuf) -= self->latency;
+ } else {
+ GST_WARNING_OBJECT (self, "GST_BUFFER_OFFSET (outbuf) < latency");
+ GST_BUFFER_OFFSET (outbuf) = 0;
+ }
+ }
+
+ if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf)) {
+ if (GST_BUFFER_OFFSET_END (outbuf) > self->latency) {
+ GST_BUFFER_OFFSET_END (outbuf) -= self->latency;
+ } else {
+ GST_WARNING_OBJECT (self, "GST_BUFFER_OFFSET_END (outbuf) < latency");
+ GST_BUFFER_OFFSET_END (outbuf) = 0;
+ }
+ }
+ }
+
+ GST_DEBUG_OBJECT (self, "Pushing buffer of size %d with timestamp: %"
+ GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld,"
+ " offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
+ GST_BUFFER_OFFSET_END (outbuf), output_samples / channels);
+
+ self->next_ts = GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf);
+ self->next_off = GST_BUFFER_OFFSET_END (outbuf);
+
return GST_FLOW_OK;
}
@@ -488,9 +680,93 @@ lpwsinc_start (GstBaseTransform * base)
memset (self->residue, 0, channels *
self->kernel_length * sizeof (gdouble));
+ self->residue_length = 0;
+ self->next_ts = GST_CLOCK_TIME_NONE;
+ self->next_off = GST_BUFFER_OFFSET_NONE;
+
return TRUE;
}
+static gboolean
+lpwsinc_query (GstPad * pad, GstQuery * query)
+{
+ GstLPWSinc *self = GST_LPWSINC (gst_pad_get_parent (pad));
+ gboolean res = TRUE;
+
+ switch (GST_QUERY_TYPE (query)) {
+ case GST_QUERY_LATENCY:
+ {
+ GstClockTime min, max;
+ gboolean live;
+ guint64 latency;
+ GstPad *peer;
+ gint rate = GST_AUDIO_FILTER (self)->format.rate;
+
+ if ((peer = gst_pad_get_peer (GST_BASE_TRANSFORM (self)->sinkpad))) {
+ if ((res = gst_pad_query (peer, query))) {
+ gst_query_parse_latency (query, &live, &min, &max);
+
+ GST_DEBUG_OBJECT (self, "Peer latency: min %"
+ GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (min), GST_TIME_ARGS (max));
+
+ /* add our own latency */
+ latency =
+ (rate != 0) ? gst_util_uint64_scale (self->latency, GST_SECOND,
+ rate) : 0;
+
+ GST_DEBUG_OBJECT (self, "Our latency: %"
+ GST_TIME_FORMAT, GST_TIME_ARGS (latency));
+
+ min += latency;
+ if (max != GST_CLOCK_TIME_NONE)
+ max += latency;
+
+ GST_DEBUG_OBJECT (self, "Calculated total latency : min %"
+ GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (min), GST_TIME_ARGS (max));
+
+ gst_query_set_latency (query, live, min, max);
+ }
+ gst_object_unref (peer);
+ }
+ break;
+ }
+ default:
+ res = gst_pad_query_default (pad, query);
+ break;
+ }
+ gst_object_unref (self);
+ return res;
+}
+
+static const GstQueryType *
+lpwsinc_query_type (GstPad * pad)
+{
+ static const GstQueryType types[] = {
+ GST_QUERY_LATENCY,
+ 0
+ };
+
+ return types;
+}
+
+static gboolean
+lpwsinc_event (GstBaseTransform * base, GstEvent * event)
+{
+ GstLPWSinc *self = GST_LPWSINC (base);
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_EOS:
+ lpwsinc_push_residue (self);
+ break;
+ default:
+ break;
+ }
+
+ return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);
+}
+
static void
lpwsinc_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
@@ -507,8 +783,17 @@ lpwsinc_set_property (GObject * object, guint prop_id, const GValue * value,
val = g_value_get_int (value);
if (val % 2 == 0)
val++;
- self->kernel_length = val;
- lpwsinc_build_kernel (self);
+
+ if (val != self->kernel_length) {
+ if (self->residue) {
+ lpwsinc_push_residue (self);
+ g_free (self->residue);
+ self->residue = NULL;
+ }
+ self->kernel_length = val;
+ self->latency = val / 2;
+ lpwsinc_build_kernel (self);
+ }
GST_BASE_TRANSFORM_UNLOCK (self);
break;
}
diff --git a/gst/filter/gstlpwsinc.h b/gst/filter/gstlpwsinc.h
index 8a8cd475..3e09bd52 100644
--- a/gst/filter/gstlpwsinc.h
+++ b/gst/filter/gstlpwsinc.h
@@ -71,6 +71,10 @@ struct _GstLPWSinc {
gdouble *residue; /* buffer for left-over samples from previous buffer */
gdouble *kernel; /* filter kernel */
gboolean have_kernel;
+ gint residue_length;
+ guint64 latency;
+ GstClockTime next_ts;
+ guint64 next_off;
};
struct _GstLPWSincClass {