/* get notified of caps and plug in the correct process function */
static gboolean
gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
GstRingBufferSpec * format)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
gboolean ret = TRUE;
if (self->residue) {
gst_audio_fx_base_fir_filter_push_residue (self);
g_free (self->residue);
self->residue = NULL;
self->residue_length = 0;
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
}
if (format->width == 32)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_32;
else if (format->width == 64)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_64;
else
ret = FALSE;
return TRUE;
}
/* get notified of caps and plug in the correct process function */
static gboolean
gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
const GstAudioInfo * info)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
g_mutex_lock (&self->lock);
if (self->buffer) {
gst_audio_fx_base_fir_filter_push_residue (self);
g_free (self->buffer);
self->buffer = NULL;
self->buffer_fill = 0;
self->buffer_length = 0;
self->start_ts = GST_CLOCK_TIME_NONE;
self->start_off = GST_BUFFER_OFFSET_NONE;
self->nsamples_out = 0;
self->nsamples_in = 0;
}
gst_audio_fx_base_fir_filter_select_process_function (self,
GST_AUDIO_INFO_FORMAT (info), GST_AUDIO_INFO_CHANNELS (info));
g_mutex_unlock (&self->lock);
return (self->process != NULL);
}
static void
gst_audio_wsincband_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (object);
g_return_if_fail (GST_IS_AUDIO_WSINC_BAND (self));
switch (prop_id) {
case PROP_LENGTH:{
gint val;
g_mutex_lock (self->lock);
val = g_value_get_int (value);
if (val % 2 == 0)
val++;
if (val != self->kernel_length) {
gst_audio_fx_base_fir_filter_push_residue (GST_AUDIO_FX_BASE_FIR_FILTER
(self));
self->kernel_length = val;
gst_audio_wsincband_build_kernel (self);
}
g_mutex_unlock (self->lock);
break;
}
case PROP_LOWER_FREQUENCY:
g_mutex_lock (self->lock);
self->lower_frequency = g_value_get_float (value);
gst_audio_wsincband_build_kernel (self);
g_mutex_unlock (self->lock);
break;
case PROP_UPPER_FREQUENCY:
g_mutex_lock (self->lock);
self->upper_frequency = g_value_get_float (value);
gst_audio_wsincband_build_kernel (self);
g_mutex_unlock (self->lock);
break;
case PROP_MODE:
g_mutex_lock (self->lock);
self->mode = g_value_get_enum (value);
gst_audio_wsincband_build_kernel (self);
g_mutex_unlock (self->lock);
break;
case PROP_WINDOW:
g_mutex_lock (self->lock);
self->window = g_value_get_enum (value);
gst_audio_wsincband_build_kernel (self);
g_mutex_unlock (self->lock);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
void
gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter * self,
gdouble * kernel, guint kernel_length, guint64 latency)
{
gboolean latency_changed;
g_return_if_fail (kernel != NULL);
g_return_if_fail (self != NULL);
GST_BASE_TRANSFORM_LOCK (self);
latency_changed = (self->latency != latency
|| (!self->low_latency && self->kernel_length < FFT_THRESHOLD
&& kernel_length >= FFT_THRESHOLD)
|| (!self->low_latency && self->kernel_length >= FFT_THRESHOLD
&& kernel_length < FFT_THRESHOLD));
/* FIXME: If the latency changes, the buffer size changes too and we
* have to drain in any case until this is fixed in the future */
if (self->buffer && (!self->drain_on_changes || latency_changed)) {
gst_audio_fx_base_fir_filter_push_residue (self);
self->start_ts = GST_CLOCK_TIME_NONE;
self->start_off = GST_BUFFER_OFFSET_NONE;
self->nsamples_out = 0;
self->nsamples_in = 0;
self->buffer_fill = 0;
}
g_free (self->kernel);
if (!self->drain_on_changes || latency_changed) {
g_free (self->buffer);
self->buffer = NULL;
self->buffer_fill = 0;
self->buffer_length = 0;
}
self->kernel = kernel;
self->kernel_length = kernel_length;
gst_audio_fx_base_fir_filter_calculate_frequency_response (self);
gst_audio_fx_base_fir_filter_select_process_function (self,
GST_AUDIO_FILTER_CAST (self)->format.width,
GST_AUDIO_FILTER_CAST (self)->format.channels);
if (latency_changed) {
self->latency = latency;
gst_element_post_message (GST_ELEMENT (self),
gst_message_new_latency (GST_OBJECT (self)));
}
GST_BASE_TRANSFORM_UNLOCK (self);
}
static gboolean
gst_audio_fx_base_fir_filter_event (GstBaseTransform * base, GstEvent * event)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
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;
break;
default:
break;
}
return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);
}
/* get notified of caps and plug in the correct process function */
static gboolean
gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
GstRingBufferSpec * format)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
if (self->buffer) {
gst_audio_fx_base_fir_filter_push_residue (self);
g_free (self->buffer);
self->buffer = NULL;
self->buffer_fill = 0;
self->buffer_length = 0;
self->start_ts = GST_CLOCK_TIME_NONE;
self->start_off = GST_BUFFER_OFFSET_NONE;
self->nsamples_out = 0;
self->nsamples_in = 0;
}
gst_audio_fx_base_fir_filter_select_process_function (self, format->width,
format->channels);
return (self->process != NULL);
}
void
gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter * self,
gdouble * kernel, guint kernel_length, guint64 latency)
{
g_return_if_fail (kernel != NULL);
g_return_if_fail (self != NULL);
GST_BASE_TRANSFORM_LOCK (self);
if (self->residue) {
gst_audio_fx_base_fir_filter_push_residue (self);
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
self->residue_length = 0;
}
g_free (self->kernel);
g_free (self->residue);
self->kernel = kernel;
self->kernel_length = kernel_length;
if (GST_AUDIO_FILTER (self)->format.channels) {
self->residue =
g_new0 (gdouble,
kernel_length * GST_AUDIO_FILTER (self)->format.channels);
self->residue_length = 0;
}
if (self->latency != latency) {
self->latency = latency;
gst_element_post_message (GST_ELEMENT (self),
gst_message_new_latency (GST_OBJECT (self)));
}
GST_BASE_TRANSFORM_UNLOCK (self);
}
static GstFlowReturn
gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
GstClockTime timestamp, expected_timestamp;
gint channels = GST_AUDIO_FILTER_CHANNELS (self);
gint rate = GST_AUDIO_FILTER_RATE (self);
gint bps = GST_AUDIO_FILTER_BPS (self);
GstMapInfo inmap, outmap;
guint input_samples;
guint output_samples;
guint generated_samples;
guint64 output_offset;
gint64 diff = 0;
GstClockTime stream_time;
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
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;
}
g_mutex_lock (&self->lock);
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);
g_return_val_if_fail (self->kernel != NULL, GST_FLOW_ERROR);
g_return_val_if_fail (channels != 0, GST_FLOW_ERROR);
if (GST_CLOCK_TIME_IS_VALID (self->start_ts))
expected_timestamp =
self->start_ts + gst_util_uint64_scale_int (self->nsamples_in,
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 (expected_timestamp)
&& (ABS (GST_CLOCK_DIFF (timestamp,
expected_timestamp) > 5 * GST_MSECOND)))) {
GST_DEBUG_OBJECT (self, "Discontinuity detected - flushing");
if (GST_CLOCK_TIME_IS_VALID (expected_timestamp))
gst_audio_fx_base_fir_filter_push_residue (self);
self->buffer_fill = 0;
g_free (self->buffer);
self->buffer = NULL;
self->start_ts = timestamp;
self->start_off = GST_BUFFER_OFFSET (inbuf);
self->nsamples_out = 0;
self->nsamples_in = 0;
} else if (!GST_CLOCK_TIME_IS_VALID (self->start_ts)) {
self->start_ts = timestamp;
self->start_off = GST_BUFFER_OFFSET (inbuf);
}
gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
input_samples = (inmap.size / bps) / channels;
output_samples = (outmap.size / bps) / channels;
self->nsamples_in += input_samples;
generated_samples =
self->process (self, inmap.data, outmap.data, input_samples);
gst_buffer_unmap (inbuf, &inmap);
gst_buffer_unmap (outbuf, &outmap);
g_assert (generated_samples <= output_samples);
self->nsamples_out += generated_samples;
if (generated_samples == 0)
goto no_samples;
/* Calculate the number of samples we can push out now without outputting
* latency zeros in the beginning */
diff = ((gint64) self->nsamples_out) - ((gint64) self->latency);
if (diff < 0)
goto no_samples;
if (diff < generated_samples) {
gint64 tmp = diff;
diff = generated_samples - diff;
generated_samples = tmp;
} else {
diff = 0;
}
gst_buffer_resize (outbuf, diff * bps * channels,
generated_samples * bps * channels);
output_offset = self->nsamples_out - self->latency - generated_samples;
GST_BUFFER_TIMESTAMP (outbuf) =
self->start_ts + gst_util_uint64_scale_int (output_offset, GST_SECOND,
rate);
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (output_samples, GST_SECOND, rate);
if (self->start_off != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) = self->start_off + output_offset;
GST_BUFFER_OFFSET_END (outbuf) =
GST_BUFFER_OFFSET (outbuf) + generated_samples;
} else {
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
}
g_mutex_unlock (&self->lock);
GST_DEBUG_OBJECT (self,
"Pushing buffer of size %" G_GSIZE_FORMAT " with timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
G_GUINT64_FORMAT ", offset_end: %" G_GUINT64_FORMAT ", nsamples_out: %d",
gst_buffer_get_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), generated_samples);
return GST_FLOW_OK;
no_samples:
{
g_mutex_unlock (&self->lock);
return GST_BASE_TRANSFORM_FLOW_DROPPED;
}
}
void
gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter * self,
gdouble * kernel, guint kernel_length, guint64 latency,
const GstAudioInfo * info)
{
gboolean latency_changed;
GstAudioFormat format;
gint channels;
g_return_if_fail (kernel != NULL);
g_return_if_fail (self != NULL);
g_mutex_lock (&self->lock);
latency_changed = (self->latency != latency
|| (!self->low_latency && self->kernel_length < FFT_THRESHOLD
&& kernel_length >= FFT_THRESHOLD)
|| (!self->low_latency && self->kernel_length >= FFT_THRESHOLD
&& kernel_length < FFT_THRESHOLD));
/* FIXME: If the latency changes, the buffer size changes too and we
* have to drain in any case until this is fixed in the future */
if (self->buffer && (!self->drain_on_changes || latency_changed)) {
gst_audio_fx_base_fir_filter_push_residue (self);
self->start_ts = GST_CLOCK_TIME_NONE;
self->start_off = GST_BUFFER_OFFSET_NONE;
self->nsamples_out = 0;
self->nsamples_in = 0;
self->buffer_fill = 0;
}
g_free (self->kernel);
if (!self->drain_on_changes || latency_changed) {
g_free (self->buffer);
self->buffer = NULL;
self->buffer_fill = 0;
self->buffer_length = 0;
}
self->kernel = kernel;
self->kernel_length = kernel_length;
if (info) {
format = GST_AUDIO_INFO_FORMAT (info);
channels = GST_AUDIO_INFO_CHANNELS (info);
} else {
format = GST_AUDIO_FILTER_FORMAT (self);
channels = GST_AUDIO_FILTER_CHANNELS (self);
}
gst_audio_fx_base_fir_filter_calculate_frequency_response (self);
gst_audio_fx_base_fir_filter_select_process_function (self, format, channels);
if (latency_changed) {
self->latency = latency;
gst_element_post_message (GST_ELEMENT (self),
gst_message_new_latency (GST_OBJECT (self)));
}
g_mutex_unlock (&self->lock);
}
static GstFlowReturn
gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (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 = 0;
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
GST_ERROR_OBJECT (self, "Invalid timestamp");
return GST_FLOW_ERROR;
}
gst_object_sync_values (G_OBJECT (self), timestamp);
g_return_val_if_fail (self->kernel != NULL, GST_FLOW_ERROR);
g_return_val_if_fail (channels != 0, GST_FLOW_ERROR);
if (!self->residue)
self->residue = g_new0 (gdouble, self->kernel_length * channels);
/* 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,
self->residue_length / channels), GST_SECOND,
rate) - self->next_ts > 5 * GST_MSECOND)) {
GST_DEBUG_OBJECT (self, "Discontinuity detected - flushing");
if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
gst_audio_fx_base_fir_filter_push_residue (self);
self->residue_length = 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);
}
/* Calculate the number of samples we can push out now without outputting
* latency zeros in the beginning */
diff = self->latency * 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) {
return GST_BASE_TRANSFORM_FLOW_DROPPED;
}
GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
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_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
if (output_samples < input_samples) {
GST_BUFFER_DATA (outbuf) +=
diff * (GST_AUDIO_FILTER (self)->format.width / 8);
GST_BUFFER_SIZE (outbuf) -=
diff * (GST_AUDIO_FILTER (self)->format.width / 8);
}
self->next_ts += GST_BUFFER_DURATION (outbuf);
self->next_off = GST_BUFFER_OFFSET_END (outbuf);
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);
return GST_FLOW_OK;
}
