static void
gst_spectrum_alloc_channel_data (GstSpectrum * spectrum)
{
gint i;
GstSpectrumChannel *cd;
guint bands = spectrum->bands;
guint nfft = 2 * bands - 2;
g_assert (spectrum->channel_data == NULL);
spectrum->num_channels = (spectrum->multi_channel) ?
GST_AUDIO_FILTER_CHANNELS (spectrum) : 1;
GST_DEBUG_OBJECT (spectrum, "allocating data for %d channels",
spectrum->num_channels);
spectrum->channel_data = g_new (GstSpectrumChannel, spectrum->num_channels);
for (i = 0; i < spectrum->num_channels; i++) {
cd = &spectrum->channel_data[i];
cd->fft_ctx = gst_fft_f32_new (nfft, FALSE);
cd->input = g_new0 (gfloat, nfft);
cd->input_tmp = g_new0 (gfloat, nfft);
cd->freqdata = g_new0 (GstFFTF32Complex, bands);
cd->spect_magnitude = g_new0 (gfloat, bands);
cd->spect_phase = g_new0 (gfloat, bands);
}
}
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_audio_fx_base_fir_filter_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (object);
switch (prop_id) {
case PROP_LOW_LATENCY:{
gboolean low_latency;
if (GST_STATE (self) >= GST_STATE_PAUSED) {
g_warning ("Changing the \"low-latency\" property "
"is only allowed in states < PAUSED");
return;
}
g_mutex_lock (&self->lock);
low_latency = g_value_get_boolean (value);
if (self->low_latency != low_latency) {
self->low_latency = low_latency;
gst_audio_fx_base_fir_filter_calculate_frequency_response (self);
gst_audio_fx_base_fir_filter_select_process_function (self,
GST_AUDIO_FILTER_FORMAT (self), GST_AUDIO_FILTER_CHANNELS (self));
}
g_mutex_unlock (&self->lock);
break;
}
case PROP_DRAIN_ON_CHANGES:{
g_mutex_lock (&self->lock);
self->drain_on_changes = g_value_get_boolean (value);
g_mutex_unlock (&self->lock);
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
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_push_residue (GstAudioFXBaseFIRFilter * self)
{
GstBuffer *outbuf;
GstFlowReturn res;
gint rate = GST_AUDIO_FILTER_RATE (self);
gint channels = GST_AUDIO_FILTER_CHANNELS (self);
gint bps = GST_AUDIO_FILTER_BPS (self);
gint outsize, outsamples;
GstMapInfo map;
guint8 *in, *out;
if (channels == 0 || rate == 0 || self->nsamples_in == 0) {
self->buffer_fill = 0;
g_free (self->buffer);
self->buffer = NULL;
return;
}
/* Calculate the number of samples and their memory size that
* should be pushed from the residue */
outsamples = self->nsamples_in - (self->nsamples_out - self->latency);
if (outsamples <= 0) {
self->buffer_fill = 0;
g_free (self->buffer);
self->buffer = NULL;
return;
}
outsize = outsamples * channels * bps;
if (!self->fft || self->low_latency) {
gint64 diffsize, diffsamples;
/* 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 =
((gint64) self->latency) - ((gint64) self->buffer_fill) / channels;
if (diffsamples > 0) {
diffsize = diffsamples * channels * bps;
in = g_new0 (guint8, diffsize);
out = g_new0 (guint8, diffsize);
self->nsamples_out += self->process (self, in, out, diffsamples);
g_free (in);
g_free (out);
}
outbuf = gst_buffer_new_and_alloc (outsize);
/* Convolve the residue with zeros to get the actual remaining data */
in = g_new0 (guint8, outsize);
gst_buffer_map (outbuf, &map, GST_MAP_READWRITE);
self->nsamples_out += self->process (self, in, map.data, outsamples);
gst_buffer_unmap (outbuf, &map);
g_free (in);
} else {
guint gensamples = 0;
outbuf = gst_buffer_new_and_alloc (outsize);
gst_buffer_map (outbuf, &map, GST_MAP_READWRITE);
while (gensamples < outsamples) {
guint step_insamples = self->block_length - self->buffer_fill;
guint8 *zeroes = g_new0 (guint8, step_insamples * channels * bps);
guint8 *out = g_new (guint8, self->block_length * channels * bps);
guint step_gensamples;
step_gensamples = self->process (self, zeroes, out, step_insamples);
g_free (zeroes);
memcpy (map.data + gensamples * bps, out, MIN (step_gensamples,
outsamples - gensamples) * bps);
gensamples += MIN (step_gensamples, outsamples - gensamples);
g_free (out);
}
self->nsamples_out += gensamples;
gst_buffer_unmap (outbuf, &map);
}
/* Set timestamp, offset, etc from the values we
* saved when processing the regular buffers */
if (GST_CLOCK_TIME_IS_VALID (self->start_ts))
GST_BUFFER_TIMESTAMP (outbuf) = self->start_ts;
else
GST_BUFFER_TIMESTAMP (outbuf) = 0;
GST_BUFFER_TIMESTAMP (outbuf) +=
gst_util_uint64_scale_int (self->nsamples_out - outsamples -
self->latency, GST_SECOND, rate);
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (outsamples, GST_SECOND, rate);
if (self->start_off != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) =
self->start_off + self->nsamples_out - outsamples - self->latency;
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET (outbuf) + outsamples;
}
GST_DEBUG_OBJECT (self,
"Pushing residue 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), outsamples);
res = gst_pad_push (GST_BASE_TRANSFORM_CAST (self)->srcpad, outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK)) {
GST_WARNING_OBJECT (self, "failed to push residue");
}
self->buffer_fill = 0;
}
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 void
gst_audio_wsinclimit_build_kernel (GstAudioWSincLimit * self,
const GstAudioInfo * info)
{
gint i = 0;
gdouble sum = 0.0;
gint len = 0;
gdouble w;
gdouble *kernel = NULL;
gint rate, channels;
len = self->kernel_length;
if (info) {
rate = GST_AUDIO_INFO_RATE (info);
channels = GST_AUDIO_INFO_CHANNELS (info);
} else {
rate = GST_AUDIO_FILTER_RATE (self);
channels = GST_AUDIO_FILTER_CHANNELS (self);
}
if (rate == 0) {
GST_DEBUG ("rate not set yet");
return;
}
if (channels == 0) {
GST_DEBUG ("channels not set yet");
return;
}
/* Clamp cutoff frequency between 0 and the nyquist frequency */
self->cutoff = CLAMP (self->cutoff, 0.0, rate / 2);
GST_DEBUG ("gst_audio_wsinclimit_: initializing filter kernel of length %d "
"with cutoff %.2lf Hz "
"for mode %s",
len, self->cutoff,
(self->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass");
/* fill the kernel */
w = 2 * G_PI * (self->cutoff / rate);
kernel = g_new (gdouble, len);
for (i = 0; i < len; ++i) {
if (i == (len - 1) / 2.0)
kernel[i] = w;
else
kernel[i] = sin (w * (i - (len - 1) / 2)) / (i - (len - 1) / 2.0);
/* windowing */
switch (self->window) {
case WINDOW_HAMMING:
kernel[i] *= (0.54 - 0.46 * cos (2 * G_PI * i / (len - 1)));
break;
case WINDOW_BLACKMAN:
kernel[i] *= (0.42 - 0.5 * cos (2 * G_PI * i / (len - 1)) +
0.08 * cos (4 * G_PI * i / (len - 1)));
break;
case WINDOW_GAUSSIAN:
kernel[i] *= exp (-0.5 * POW2 (3.0 / len * (2 * i - (len - 1))));
break;
case WINDOW_COSINE:
kernel[i] *= cos (G_PI * i / (len - 1) - G_PI / 2);
break;
case WINDOW_HANN:
kernel[i] *= 0.5 * (1 - cos (2 * G_PI * i / (len - 1)));
break;
}
}
/* normalize for unity gain at DC */
for (i = 0; i < len; ++i)
sum += kernel[i];
for (i = 0; i < len; ++i)
kernel[i] /= sum;
/* convert to highpass if specified */
if (self->mode == MODE_HIGH_PASS) {
for (i = 0; i < len; ++i)
kernel[i] = -kernel[i];
if (len % 2 == 1) {
kernel[(len - 1) / 2] += 1.0;
} else {
kernel[len / 2 - 1] += 0.5;
kernel[len / 2] += 0.5;
}
}
gst_audio_fx_base_fir_filter_set_kernel (GST_AUDIO_FX_BASE_FIR_FILTER (self),
kernel, self->kernel_length, (len - 1) / 2, info);
}
static GstFlowReturn
gst_spectrum_transform_ip (GstBaseTransform * trans, GstBuffer * buffer)
{
GstSpectrum *spectrum = GST_SPECTRUM (trans);
guint rate = GST_AUDIO_FILTER_RATE (spectrum);
guint channels = GST_AUDIO_FILTER_CHANNELS (spectrum);
guint bps = GST_AUDIO_FILTER_BPS (spectrum);
guint bpf = GST_AUDIO_FILTER_BPF (spectrum);
guint output_channels = spectrum->multi_channel ? channels : 1;
guint c;
gfloat max_value = (1UL << ((bps << 3) - 1)) - 1;
guint bands = spectrum->bands;
guint nfft = 2 * bands - 2;
guint input_pos;
gfloat *input;
GstMapInfo map;
const guint8 *data;
gsize size;
guint fft_todo, msg_todo, block_size;
gboolean have_full_interval;
GstSpectrumChannel *cd;
GstSpectrumInputData input_data;
g_mutex_lock (&spectrum->lock);
gst_buffer_map (buffer, &map, GST_MAP_READ);
data = map.data;
size = map.size;
GST_LOG_OBJECT (spectrum, "input size: %" G_GSIZE_FORMAT " bytes", size);
if (GST_BUFFER_IS_DISCONT (buffer)) {
GST_DEBUG_OBJECT (spectrum, "Discontinuity detected -- flushing");
gst_spectrum_flush (spectrum);
}
/* If we don't have a FFT context yet (or it was reset due to parameter
* changes) get one and allocate memory for everything
*/
if (spectrum->channel_data == NULL) {
GST_DEBUG_OBJECT (spectrum, "allocating for bands %u", bands);
gst_spectrum_alloc_channel_data (spectrum);
/* number of sample frames we process before posting a message
* interval is in ns */
spectrum->frames_per_interval =
gst_util_uint64_scale (spectrum->interval, rate, GST_SECOND);
spectrum->frames_todo = spectrum->frames_per_interval;
/* rounding error for frames_per_interval in ns,
* aggregated it in accumulated_error */
spectrum->error_per_interval = (spectrum->interval * rate) % GST_SECOND;
if (spectrum->frames_per_interval == 0)
spectrum->frames_per_interval = 1;
GST_INFO_OBJECT (spectrum, "interval %" GST_TIME_FORMAT ", fpi %"
G_GUINT64_FORMAT ", error %" GST_TIME_FORMAT,
GST_TIME_ARGS (spectrum->interval), spectrum->frames_per_interval,
GST_TIME_ARGS (spectrum->error_per_interval));
spectrum->input_pos = 0;
gst_spectrum_flush (spectrum);
}
if (spectrum->num_frames == 0)
spectrum->message_ts = GST_BUFFER_TIMESTAMP (buffer);
input_pos = spectrum->input_pos;
input_data = spectrum->input_data;
while (size >= bpf) {
/* run input_data for a chunk of data */
fft_todo = nfft - (spectrum->num_frames % nfft);
msg_todo = spectrum->frames_todo - spectrum->num_frames;
GST_LOG_OBJECT (spectrum,
"message frames todo: %u, fft frames todo: %u, input frames %"
G_GSIZE_FORMAT, msg_todo, fft_todo, (size / bpf));
block_size = msg_todo;
if (block_size > (size / bpf))
block_size = (size / bpf);
if (block_size > fft_todo)
block_size = fft_todo;
for (c = 0; c < output_channels; c++) {
cd = &spectrum->channel_data[c];
input = cd->input;
/* Move the current frames into our ringbuffers */
input_data (data + c * bps, input, block_size, channels, max_value,
input_pos, nfft);
}
data += block_size * bpf;
size -= block_size * bpf;
input_pos = (input_pos + block_size) % nfft;
spectrum->num_frames += block_size;
have_full_interval = (spectrum->num_frames == spectrum->frames_todo);
GST_LOG_OBJECT (spectrum,
"size: %" G_GSIZE_FORMAT ", do-fft = %d, do-message = %d", size,
(spectrum->num_frames % nfft == 0), have_full_interval);
/* If we have enough frames for an FFT or we have all frames required for
* the interval and we haven't run a FFT, then run an FFT */
if ((spectrum->num_frames % nfft == 0) ||
(have_full_interval && !spectrum->num_fft)) {
for (c = 0; c < output_channels; c++) {
cd = &spectrum->channel_data[c];
gst_spectrum_run_fft (spectrum, cd, input_pos);
}
spectrum->num_fft++;
}
/* Do we have the FFTs for one interval? */
if (have_full_interval) {
GST_DEBUG_OBJECT (spectrum, "nfft: %u frames: %" G_GUINT64_FORMAT
" fpi: %" G_GUINT64_FORMAT " error: %" GST_TIME_FORMAT, nfft,
spectrum->num_frames, spectrum->frames_per_interval,
GST_TIME_ARGS (spectrum->accumulated_error));
spectrum->frames_todo = spectrum->frames_per_interval;
if (spectrum->accumulated_error >= GST_SECOND) {
spectrum->accumulated_error -= GST_SECOND;
spectrum->frames_todo++;
}
spectrum->accumulated_error += spectrum->error_per_interval;
if (spectrum->post_messages) {
GstMessage *m;
for (c = 0; c < output_channels; c++) {
cd = &spectrum->channel_data[c];
gst_spectrum_prepare_message_data (spectrum, cd);
}
m = gst_spectrum_message_new (spectrum, spectrum->message_ts,
spectrum->interval);
gst_element_post_message (GST_ELEMENT (spectrum), m);
}
if (GST_CLOCK_TIME_IS_VALID (spectrum->message_ts))
spectrum->message_ts +=
gst_util_uint64_scale (spectrum->num_frames, GST_SECOND, rate);
for (c = 0; c < output_channels; c++) {
cd = &spectrum->channel_data[c];
gst_spectrum_reset_message_data (spectrum, cd);
}
spectrum->num_frames = 0;
spectrum->num_fft = 0;
}
}
spectrum->input_pos = input_pos;
gst_buffer_unmap (buffer, &map);
g_mutex_unlock (&spectrum->lock);
g_assert (size == 0);
return GST_FLOW_OK;
}
static GstMessage *
gst_spectrum_message_new (GstSpectrum * spectrum, GstClockTime timestamp,
GstClockTime duration)
{
GstBaseTransform *trans = GST_BASE_TRANSFORM_CAST (spectrum);
GstSpectrumChannel *cd;
GstStructure *s;
GValue *mcv = NULL, *pcv = NULL;
GstClockTime endtime, running_time, stream_time;
GST_DEBUG_OBJECT (spectrum, "preparing message, bands =%d ", spectrum->bands);
running_time = gst_segment_to_running_time (&trans->segment, GST_FORMAT_TIME,
timestamp);
stream_time = gst_segment_to_stream_time (&trans->segment, GST_FORMAT_TIME,
timestamp);
/* endtime is for backwards compatibility */
endtime = stream_time + duration;
s = gst_structure_new ("spectrum",
"endtime", GST_TYPE_CLOCK_TIME, endtime,
"timestamp", G_TYPE_UINT64, timestamp,
"stream-time", G_TYPE_UINT64, stream_time,
"running-time", G_TYPE_UINT64, running_time,
"duration", G_TYPE_UINT64, duration, NULL);
if (!spectrum->multi_channel) {
cd = &spectrum->channel_data[0];
if (spectrum->message_magnitude) {
/* FIXME 0.11: this should be an array, not a list */
mcv = gst_spectrum_message_add_container (s, GST_TYPE_LIST, "magnitude");
gst_spectrum_message_add_list (mcv, cd->spect_magnitude, spectrum->bands);
}
if (spectrum->message_phase) {
/* FIXME 0.11: this should be an array, not a list */
pcv = gst_spectrum_message_add_container (s, GST_TYPE_LIST, "phase");
gst_spectrum_message_add_list (pcv, cd->spect_phase, spectrum->bands);
}
} else {
guint c;
guint channels = GST_AUDIO_FILTER_CHANNELS (spectrum);
if (spectrum->message_magnitude) {
mcv = gst_spectrum_message_add_container (s, GST_TYPE_ARRAY, "magnitude");
}
if (spectrum->message_phase) {
pcv = gst_spectrum_message_add_container (s, GST_TYPE_ARRAY, "phase");
}
for (c = 0; c < channels; c++) {
cd = &spectrum->channel_data[c];
if (spectrum->message_magnitude) {
gst_spectrum_message_add_array (mcv, cd->spect_magnitude,
spectrum->bands);
}
if (spectrum->message_phase) {
gst_spectrum_message_add_array (pcv, cd->spect_magnitude,
spectrum->bands);
}
}
}
return gst_message_new_element (GST_OBJECT (spectrum), s);
}
static void
gst_audio_wsincband_build_kernel (GstAudioWSincBand * self,
const GstAudioInfo * info)
{
gint i = 0;
gdouble sum = 0.0;
gint len = 0;
gdouble *kernel_lp, *kernel_hp;
gdouble w;
gdouble *kernel;
gint rate, channels;
len = self->kernel_length;
if (info) {
rate = GST_AUDIO_INFO_RATE (info);
channels = GST_AUDIO_INFO_CHANNELS (info);
} else {
rate = GST_AUDIO_FILTER_RATE (self);
channels = GST_AUDIO_FILTER_CHANNELS (self);
}
if (rate == 0) {
GST_DEBUG ("rate not set yet");
return;
}
if (channels == 0) {
GST_DEBUG ("channels not set yet");
return;
}
/* Clamp frequencies */
self->lower_frequency = CLAMP (self->lower_frequency, 0.0, rate / 2);
self->upper_frequency = CLAMP (self->upper_frequency, 0.0, rate / 2);
if (self->lower_frequency > self->upper_frequency) {
gint tmp = self->lower_frequency;
self->lower_frequency = self->upper_frequency;
self->upper_frequency = tmp;
}
GST_DEBUG ("gst_audio_wsincband: initializing filter kernel of length %d "
"with lower frequency %.2lf Hz "
", upper frequency %.2lf Hz for mode %s",
len, self->lower_frequency, self->upper_frequency,
(self->mode == MODE_BAND_PASS) ? "band-pass" : "band-reject");
/* fill the lp kernel */
w = 2 * G_PI * (self->lower_frequency / rate);
kernel_lp = g_new (gdouble, len);
for (i = 0; i < len; ++i) {
if (i == (len - 1) / 2.0)
kernel_lp[i] = w;
else
kernel_lp[i] = sin (w * (i - (len - 1) / 2.0)) / (i - (len - 1) / 2.0);
/* windowing */
switch (self->window) {
case WINDOW_HAMMING:
kernel_lp[i] *= (0.54 - 0.46 * cos (2 * G_PI * i / (len - 1)));
break;
case WINDOW_BLACKMAN:
kernel_lp[i] *= (0.42 - 0.5 * cos (2 * G_PI * i / (len - 1)) +
0.08 * cos (4 * G_PI * i / (len - 1)));
break;
case WINDOW_GAUSSIAN:
kernel_lp[i] *= exp (-0.5 * POW2 (3.0 / len * (2 * i - (len - 1))));
break;
case WINDOW_COSINE:
kernel_lp[i] *= cos (G_PI * i / (len - 1) - G_PI / 2);
break;
case WINDOW_HANN:
kernel_lp[i] *= 0.5 * (1 - cos (2 * G_PI * i / (len - 1)));
break;
}
}
/* normalize for unity gain at DC */
sum = 0.0;
for (i = 0; i < len; ++i)
sum += kernel_lp[i];
for (i = 0; i < len; ++i)
kernel_lp[i] /= sum;
/* fill the hp kernel */
w = 2 * G_PI * (self->upper_frequency / rate);
kernel_hp = g_new (gdouble, len);
for (i = 0; i < len; ++i) {
if (i == (len - 1) / 2.0)
kernel_hp[i] = w;
else
kernel_hp[i] = sin (w * (i - (len - 1) / 2.0)) / (i - (len - 1) / 2.0);
/* Windowing */
switch (self->window) {
case WINDOW_HAMMING:
kernel_hp[i] *= (0.54 - 0.46 * cos (2 * G_PI * i / (len - 1)));
break;
case WINDOW_BLACKMAN:
kernel_hp[i] *= (0.42 - 0.5 * cos (2 * G_PI * i / (len - 1)) +
0.08 * cos (4 * G_PI * i / (len - 1)));
break;
case WINDOW_GAUSSIAN:
kernel_hp[i] *= exp (-0.5 * POW2 (3.0 / len * (2 * i - (len - 1))));
break;
case WINDOW_COSINE:
kernel_hp[i] *= cos (G_PI * i / (len - 1) - G_PI / 2);
break;
case WINDOW_HANN:
kernel_hp[i] *= 0.5 * (1 - cos (2 * G_PI * i / (len - 1)));
break;
}
}
/* normalize for unity gain at DC */
sum = 0.0;
for (i = 0; i < len; ++i)
sum += kernel_hp[i];
for (i = 0; i < len; ++i)
kernel_hp[i] /= sum;
/* do spectral inversion to go from lowpass to highpass */
for (i = 0; i < len; ++i)
kernel_hp[i] = -kernel_hp[i];
if (len % 2 == 1) {
kernel_hp[(len - 1) / 2] += 1.0;
} else {
kernel_hp[len / 2 - 1] += 0.5;
kernel_hp[len / 2] += 0.5;
}
/* combine the two kernels */
kernel = g_new (gdouble, len);
for (i = 0; i < len; ++i)
kernel[i] = kernel_lp[i] + kernel_hp[i];
/* free the helper kernels */
g_free (kernel_lp);
g_free (kernel_hp);
/* do spectral inversion to go from bandreject to bandpass
* if specified */
if (self->mode == MODE_BAND_PASS) {
for (i = 0; i < len; ++i)
kernel[i] = -kernel[i];
kernel[len / 2] += 1;
}
gst_audio_fx_base_fir_filter_set_kernel (GST_AUDIO_FX_BASE_FIR_FILTER (self),
kernel, self->kernel_length, (len - 1) / 2, info);
}
