mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-02-17 03:35:21 +00:00
Code Issues 1 Projects Releases Wiki Activity

ext/soundtouch/gstpitch.*: Implement LATENCY query and notify about latency changes.

Browse source 
Original commit message from CVS:
* ext/soundtouch/gstpitch.cc:
* ext/soundtouch/gstpitch.hh:
Implement LATENCY query and notify about latency changes.
Unfortunately we don't have a fixed latency but it changes
a bit with each buffer so we only send an LATENCY event with
the maximum latency if it changes.
Always calculate the timestamp, duration, etc from the sample
rate instead of using a pre-calculated duration for one sample
to prevent large rounding errors.
This commit is contained in:
Sebastian Dröge 2008-01-27 07:37:40 +00:00
parent 5b791c2ce5
commit 26b4a028d9
3 changed files with 106 additions and 15 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

13
ChangeLog
View file

@ -1,3 +1,16 @@
2008-01-27 Sebastian Dröge <slomo@circular-chaos.org>
* ext/soundtouch/gstpitch.cc:
* ext/soundtouch/gstpitch.hh:
Implement LATENCY query and notify about latency changes.
Unfortunately we don't have a fixed latency but it changes
a bit with each buffer so we only send an LATENCY event with
the maximum latency if it changes.
Always calculate the timestamp, duration, etc from the sample
rate instead of using a pre-calculated duration for one sample
to prevent large rounding errors.
2008-01-27 Sebastian Dröge <slomo@circular-chaos.org>
Based on a patch by:

105
ext/soundtouch/gstpitch.cc
View file

@ -194,6 +194,7 @@ gst_pitch_init (GstPitch * pitch, GstPitchClass * pitch_class)
pitch->priv->st->setPitch (pitch->pitch);
pitch->priv->stream_time_ratio = 1.0;
pitch->min_latency = pitch->max_latency = 0;
}
@ -312,7 +313,6 @@ gst_pitch_sink_setcaps (GstPad * pad, GstCaps * caps)
/* calculate sample size */
pitch->sample_size = (sizeof (gfloat) * channels);
pitch->sample_duration = gst_util_uint64_scale_int (GST_SECOND, 1, rate);
GST_OBJECT_UNLOCK (pitch);
@ -371,7 +371,8 @@ gst_pitch_prepare_buffer (GstPitch * pitch)
return NULL;
}
GST_BUFFER_DURATION (buffer) = samples * pitch->sample_duration;
GST_BUFFER_DURATION (buffer) =
gst_util_uint64_scale (samples, GST_SECOND, pitch->samplerate);
/* temporary store samples here, to avoid having to recalculate this */
GST_BUFFER_OFFSET (buffer) = (gint64) samples;
@ -466,18 +467,17 @@ gst_pitch_convert (GstPitch * pitch,
GstFormat * dst_format, gint64 * dst_value)
{
gboolean res = TRUE;
GstClockTime sample_duration;
guint sample_size;
gint samplerate;
g_return_val_if_fail (dst_format && dst_value, FALSE);
GST_OBJECT_LOCK (pitch);
sample_duration = pitch->sample_duration;
sample_size = pitch->sample_size;
samplerate = pitch->samplerate;
GST_OBJECT_UNLOCK (pitch);
if (sample_size == 0 || sample_duration == 0 ||
sample_duration == GST_CLOCK_TIME_NONE) {
if (sample_size == 0 || samplerate == 0) {
return FALSE;
}
@ -490,11 +490,12 @@ gst_pitch_convert (GstPitch * pitch,
case GST_FORMAT_BYTES:
switch (*dst_format) {
case GST_FORMAT_TIME:
*dst_value = src_value / sample_size;
*dst_value *= sample_duration;
*dst_value =
gst_util_uint64_scale_int (src_value, GST_SECOND,
sample_size * samplerate);
break;
case GST_FORMAT_DEFAULT:
*dst_value = src_value / sample_size;
*dst_value = gst_util_uint64_scale_int (src_value, 1, sample_size);
break;
default:
res = FALSE;
@ -504,11 +505,13 @@ gst_pitch_convert (GstPitch * pitch,
case GST_FORMAT_TIME:
switch (*dst_format) {
case GST_FORMAT_BYTES:
*dst_value = src_value / sample_duration;
*dst_value *= sample_size;
*dst_value =
gst_util_uint64_scale_int (src_value, samplerate * sample_size,
GST_SECOND);
break;
case GST_FORMAT_DEFAULT:
*dst_value = src_value / sample_duration;
*dst_value =
gst_util_uint64_scale_int (src_value, samplerate, GST_SECOND);
break;
default:
res = FALSE;
@ -518,10 +521,11 @@ gst_pitch_convert (GstPitch * pitch,
case GST_FORMAT_DEFAULT:
switch (*dst_format) {
case GST_FORMAT_BYTES:
*dst_value = src_value * sample_size;
*dst_value = gst_util_uint64_scale_int (src_value, sample_size, 1);
break;
case GST_FORMAT_TIME:
*dst_value = src_value * sample_duration;
*dst_value =
gst_util_uint64_scale_int (src_value, GST_SECOND, samplerate);
break;
default:
res = FALSE;
@ -543,6 +547,7 @@ gst_pitch_get_query_types (GstPad * pad)
GST_QUERY_POSITION,
GST_QUERY_DURATION,
GST_QUERY_CONVERT,
GST_QUERY_LATENCY,
GST_QUERY_NONE
};
@ -630,6 +635,46 @@ gst_pitch_src_query (GstPad * pad, GstQuery * query)
}
break;
}
case GST_QUERY_LATENCY:
{
GstClockTime min, max;
gboolean live;
GstPad *peer;
if ((peer = gst_pad_get_peer (pitch->sinkpad))) {
if ((res = gst_pad_query (peer, query))) {
gst_query_parse_latency (query, &live, &min, &max);
GST_DEBUG ("Peer latency: min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
/* add our own latency */
GST_DEBUG ("Our latency: min %" GST_TIME_FORMAT
", max %" GST_TIME_FORMAT,
GST_TIME_ARGS (pitch->min_latency),
GST_TIME_ARGS (pitch->max_latency));
min += pitch->min_latency;
if (max != GST_CLOCK_TIME_NONE)
max += pitch->max_latency;
else
max = pitch->max_latency;
GST_DEBUG ("Calculated total latency : min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
g_print ("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;
@ -728,10 +773,12 @@ gst_pitch_sink_event (GstPad * pad, GstEvent * event)
case GST_EVENT_FLUSH_STOP:
gst_pitch_flush_buffer (pitch, FALSE);
pitch->priv->st->clear ();
pitch->min_latency = pitch->max_latency = 0;
break;
case GST_EVENT_EOS:
gst_pitch_flush_buffer (pitch, TRUE);
pitch->priv->st->clear ();
pitch->min_latency = pitch->max_latency = 0;
break;
case GST_EVENT_NEWSEGMENT:
if (!gst_pitch_process_segment (pitch, &event)) {
@ -742,6 +789,7 @@ gst_pitch_sink_event (GstPad * pad, GstEvent * event)
event = NULL;
}
pitch->priv->st->clear ();
pitch->min_latency = pitch->max_latency = 0;
break;
default:
break;
@ -755,17 +803,41 @@ gst_pitch_sink_event (GstPad * pad, GstEvent * event)
return res;
}
static void
gst_pitch_update_latency (GstPitch * pitch, GstClockTime timestamp)
{
GstClockTimeDiff current_latency, min_latency, max_latency;
current_latency =
timestamp / pitch->priv->stream_time_ratio - pitch->next_buffer_time;
min_latency = MIN (pitch->min_latency, current_latency);
max_latency = MAX (pitch->max_latency, current_latency);
if (pitch->min_latency != min_latency || pitch->max_latency != max_latency) {
pitch->min_latency = min_latency;
pitch->max_latency = max_latency;
gst_pad_push_event (pitch->sinkpad, gst_event_new_latency (max_latency));
gst_element_post_message (GST_ELEMENT (pitch),
gst_message_new_latency (GST_OBJECT (pitch)));
}
}
static GstFlowReturn
gst_pitch_chain (GstPad * pad, GstBuffer * buffer)
{
GstPitch *pitch;
GstPitchPrivate *priv;
GstClockTime timestamp;
pitch = GST_PITCH (GST_PAD_PARENT (pad));
priv = GST_PITCH_GET_PRIVATE (pitch);
gst_object_sync_values (G_OBJECT (pitch), pitch->next_buffer_time);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
/* push the received samples on the soundtouch buffer */
GST_LOG_OBJECT (pitch, "incoming buffer (%d samples)",
(gint) (GST_BUFFER_SIZE (buffer) / pitch->sample_size));
@ -793,6 +865,10 @@ gst_pitch_chain (GstPad * pad, GstBuffer * buffer)
GST_BUFFER_SIZE (buffer) / pitch->sample_size);
gst_buffer_unref (buffer);
/* Calculate latency */
gst_pitch_update_latency (pitch, timestamp);
/* and try to extract some samples from the soundtouch buffer */
if (!priv->st->isEmpty ()) {
GstBuffer *out_buffer;
@ -818,6 +894,7 @@ gst_pitch_change_state (GstElement * element, GstStateChange transition)
pitch->next_buffer_time = 0;
pitch->next_buffer_offset = 0;
pitch->priv->st->clear ();
pitch->min_latency = pitch->max_latency = 0;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;

3
ext/soundtouch/gstpitch.hh
View file

@ -67,12 +67,13 @@ struct _GstPitch
gint samplerate; /* samplerate */
gint channels; /* number of audio channels */
gsize sample_size; /* number of bytes for a single sample */
GstClockTime sample_duration; /* time for 1 sample */
/* stream tracking */
GstClockTime next_buffer_time;
gint64 next_buffer_offset;
GstClockTimeDiff min_latency, max_latency;
GstPitchPrivate *priv;
};