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audiofxbasefirfilter: Rewrite timestamp tracking

Browse source 
It's much simpler now and doesn't introduce accumulating rounding
errors.
This commit is contained in:
Sebastian Dröge 2009-11-25 18:12:05 +01:00
parent c57be62881
commit abb437454e
2 changed files with 63 additions and 41 deletions
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99
gst/audiofx/audiofxbasefirfilter.c
View file

@ -126,8 +126,9 @@ gst_audio_fx_base_fir_filter_init (GstAudioFXBaseFIRFilter * self,
self->kernel = NULL;
self->buffer = NULL;
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
self->start_ts = GST_CLOCK_TIME_NONE;
self->start_off = GST_BUFFER_OFFSET_NONE;
self->nsamples = 0;
gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad,
gst_audio_fx_base_fir_filter_query);
@ -237,20 +238,23 @@ gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
/* 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;
if (GST_CLOCK_TIME_IS_VALID (self->start_ts))
GST_BUFFER_TIMESTAMP (outbuf) = self->start_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);
GST_BUFFER_TIMESTAMP (outbuf) +=
gst_util_uint64_scale_round (self->nsamples, 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;
self->next_off = GST_BUFFER_OFFSET_END (outbuf);
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_round (outsamples, GST_SECOND, rate);
if (self->start_off != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) = self->start_off + self->nsamples;
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET (outbuf) + outsamples;
}
self->nsamples += outsamples;
GST_DEBUG_OBJECT (self, "Pushing residue buffer of size %d with timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
G_GUINT64_FORMAT ", offset_end: %" G_GUINT64_FORMAT ", nsamples: %d",
@ -282,8 +286,9 @@ gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
g_free (self->buffer);
self->buffer = NULL;
self->buffer_fill = 0;
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
self->start_ts = GST_CLOCK_TIME_NONE;
self->start_off = GST_BUFFER_OFFSET_NONE;
self->nsamples = 0;
}
if (format->width == 32)
@ -303,7 +308,7 @@ gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
GstClockTime timestamp;
GstClockTime timestamp, expected_timestamp;
gint channels = GST_AUDIO_FILTER (self)->format.channels;
gint rate = GST_AUDIO_FILTER (self)->format.rate;
gint input_samples =
@ -312,7 +317,8 @@ gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
gint diff = 0;
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
if (!GST_CLOCK_TIME_IS_VALID (timestamp)
&& !GST_CLOCK_TIME_IS_VALID (self->start_ts)) {
GST_ERROR_OBJECT (self, "Invalid timestamp");
return GST_FLOW_ERROR;
}
@ -325,20 +331,29 @@ gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
if (!self->buffer)
self->buffer = g_new0 (gdouble, self->kernel_length * channels);
if (GST_CLOCK_TIME_IS_VALID (self->start_ts))
expected_timestamp =
self->start_ts + gst_util_uint64_scale_round (self->nsamples,
GST_SECOND, rate);
else
expected_timestamp = GST_CLOCK_TIME_NONE;
/* Reset the residue if already existing on discont buffers */
if (GST_BUFFER_IS_DISCONT (inbuf) || (GST_CLOCK_TIME_IS_VALID (self->next_ts)
&& timestamp - gst_util_uint64_scale (MIN (self->latency,
if (GST_BUFFER_IS_DISCONT (inbuf)
|| (GST_CLOCK_TIME_IS_VALID (expected_timestamp)
&& timestamp - gst_util_uint64_scale_round (MIN (self->latency,
self->buffer_fill / channels), GST_SECOND,
rate) - self->next_ts > 5 * GST_MSECOND)) {
rate) - expected_timestamp > 5 * GST_MSECOND)) {
GST_DEBUG_OBJECT (self, "Discontinuity detected - flushing");
if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
if (GST_CLOCK_TIME_IS_VALID (expected_timestamp))
gst_audio_fx_base_fir_filter_push_residue (self);
self->buffer_fill = 0;
self->next_ts = timestamp;
self->next_off = GST_BUFFER_OFFSET (inbuf);
} else if (!GST_CLOCK_TIME_IS_VALID (self->next_ts)) {
self->next_ts = timestamp;
self->next_off = GST_BUFFER_OFFSET (inbuf);
expected_timestamp = self->start_ts = timestamp;
self->start_off = GST_BUFFER_OFFSET (inbuf);
self->nsamples = 0;
} else if (!GST_CLOCK_TIME_IS_VALID (self->start_ts)) {
expected_timestamp = self->start_ts = timestamp;
self->start_off = GST_BUFFER_OFFSET (inbuf);
}
/* Calculate the number of samples we can push out now without outputting
@ -354,14 +369,17 @@ gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
return GST_BASE_TRANSFORM_FLOW_DROPPED;
}
GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
GST_BUFFER_TIMESTAMP (outbuf) = expected_timestamp;
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale (output_samples / channels, GST_SECOND, rate);
GST_BUFFER_OFFSET (outbuf) = self->next_off;
if (GST_BUFFER_OFFSET_IS_VALID (outbuf))
GST_BUFFER_OFFSET_END (outbuf) = self->next_off + output_samples / channels;
else
gst_util_uint64_scale_round (output_samples / channels, GST_SECOND, rate);
if (self->start_off != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) = self->start_off + self->nsamples;
GST_BUFFER_OFFSET_END (outbuf) =
self->start_off + output_samples / channels;
} else {
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
}
if (output_samples < input_samples) {
GST_BUFFER_DATA (outbuf) +=
@ -370,8 +388,7 @@ gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
diff * (GST_AUDIO_FILTER (self)->format.width / 8);
}
self->next_ts += GST_BUFFER_DURATION (outbuf);
self->next_off = GST_BUFFER_OFFSET_END (outbuf);
self->nsamples += output_samples / channels;
GST_DEBUG_OBJECT (self, "Pushing buffer of size %d with timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
@ -389,8 +406,9 @@ gst_audio_fx_base_fir_filter_start (GstBaseTransform * base)
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
self->buffer_fill = 0;
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
self->start_ts = GST_CLOCK_TIME_NONE;
self->start_off = GST_BUFFER_OFFSET_NONE;
self->nsamples = 0;
return TRUE;
}
@ -433,7 +451,8 @@ gst_audio_fx_base_fir_filter_query (GstPad * pad, GstQuery * query)
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
/* add our own latency */
latency = gst_util_uint64_scale (self->latency, GST_SECOND, rate);
latency =
gst_util_uint64_scale_round (self->latency, GST_SECOND, rate);
GST_DEBUG_OBJECT (self, "Our latency: %"
GST_TIME_FORMAT, GST_TIME_ARGS (latency));
@ -479,8 +498,9 @@ gst_audio_fx_base_fir_filter_event (GstBaseTransform * base, GstEvent * event)
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
gst_audio_fx_base_fir_filter_push_residue (self);
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
self->start_ts = GST_CLOCK_TIME_NONE;
self->start_off = GST_BUFFER_OFFSET_NONE;
self->nsamples = 0;
break;
default:
break;
@ -499,8 +519,9 @@ gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter * self,
GST_BASE_TRANSFORM_LOCK (self);
if (self->buffer) {
gst_audio_fx_base_fir_filter_push_residue (self);
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
self->start_ts = GST_CLOCK_TIME_NONE;
self->start_off = GST_BUFFER_OFFSET_NONE;
self->nsamples = 0;
self->buffer_fill = 0;
}

5
gst/audiofx/audiofxbasefirfilter.h
View file

@ -65,8 +65,9 @@ struct _GstAudioFXBaseFIRFilter {
guint64 latency;
GstClockTime next_ts;
guint64 next_off;
GstClockTime start_ts; /* start timestamp after a discont */
guint64 start_off; /* start offset after a discont */
guint64 nsamples; /* number of samples since last discont */
};
struct _GstAudioFXBaseFIRFilterClass {