Commit ab24971a authored by Olivier Crête's avatar Olivier Crête 👻

speed: Various 1.0 style additions

Use GstAudioInfo everywhere, etc
parent 98ffddb0
......@@ -61,12 +61,7 @@ enum
/* assumption here: sizeof (gfloat) = 4 */
#define GST_SPEED_AUDIO_CAPS \
"audio/x-raw, " \
"format = (string) " GST_AUDIO_NE (F32) ", " \
"rate = (int) [ 8000, MAX ], " \
"channels = (int) [ 1, 2 ]; " \
\
"audio/x-raw, " \
"format = (string) " GST_AUDIO_NE (S16) ", " \
"format = {" GST_AUDIO_NE (F32) ", " GST_AUDIO_NE (S16) "}, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ]"
......@@ -101,7 +96,6 @@ static gboolean speed_sink_event (GstPad * pad, GstObject * parent,
static gboolean speed_src_event (GstPad * pad, GstObject * parent,
GstEvent * event);
#define gst_speed_parent_class parent_class
G_DEFINE_TYPE (GstSpeed, gst_speed, GST_TYPE_ELEMENT);
static gboolean
......@@ -122,37 +116,12 @@ speed_setcaps (GstPad * pad, GstCaps * caps)
static gboolean
speed_parse_caps (GstSpeed * filter, const GstCaps * caps)
{
GstAudioInfo info;
gint rate, chans, width;
g_return_val_if_fail (filter != NULL, FALSE);
g_return_val_if_fail (caps != NULL, FALSE);
gst_audio_info_from_caps (&info, caps);
if (GST_AUDIO_INFO_IS_FLOAT (&info))
filter->format = GST_SPEED_FORMAT_FLOAT;
else if (GST_AUDIO_INFO_IS_INTEGER (&info))
filter->format = GST_SPEED_FORMAT_INT;
else
if (!gst_audio_info_from_caps (&filter->info, caps))
return FALSE;
rate = info.rate;
width = GST_AUDIO_INFO_WIDTH (&info);
chans = info.channels;
if (!rate || !width || !chans)
return FALSE;
filter->rate = rate;
filter->width = width;
filter->channels = chans;
if (filter->format == GST_SPEED_FORMAT_FLOAT) {
filter->sample_size = filter->channels * filter->width / 8;
} else {
/* our caps only allow width == depth for now */
filter->sample_size = filter->channels * filter->width / 8;
}
return TRUE;
}
......@@ -208,33 +177,32 @@ speed_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
}
static gboolean
gst_speed_convert (GstPad * pad, GstFormat src_format, gint64 src_value,
gst_speed_convert (GstSpeed * filter, GstFormat src_format, gint64 src_value,
GstFormat * dest_format, gint64 * dest_value)
{
gboolean ret = TRUE;
guint scale = 1;
GstSpeed *filter;
if (src_format == *dest_format) {
*dest_value = src_value;
return TRUE;
}
filter = GST_SPEED (GST_PAD_PARENT (pad));
switch (src_format) {
case GST_FORMAT_BYTES:
switch (*dest_format) {
case GST_FORMAT_DEFAULT:
if (filter->sample_size == 0) {
if (GST_AUDIO_INFO_BPF (&filter->info) == 0) {
ret = FALSE;
break;
}
*dest_value = src_value / filter->sample_size;
*dest_value = src_value / GST_AUDIO_INFO_BPF (&filter->info);
break;
case GST_FORMAT_TIME:
{
gint byterate = filter->sample_size * filter->rate;
gint byterate =
GST_AUDIO_INFO_BPF (&filter->info) *
GST_AUDIO_INFO_RATE (&filter->info);
if (byterate == 0) {
ret = FALSE;
......@@ -250,14 +218,15 @@ gst_speed_convert (GstPad * pad, GstFormat src_format, gint64 src_value,
case GST_FORMAT_DEFAULT:
switch (*dest_format) {
case GST_FORMAT_BYTES:
*dest_value = src_value * filter->sample_size;
*dest_value = src_value * GST_AUDIO_INFO_BPF (&filter->info);
break;
case GST_FORMAT_TIME:
if (filter->rate == 0) {
if (GST_AUDIO_INFO_RATE (&filter->info) == 0) {
ret = FALSE;
break;
}
*dest_value = src_value * GST_SECOND / filter->rate;
*dest_value =
src_value * GST_SECOND / GST_AUDIO_INFO_RATE (&filter->info);
break;
default:
ret = FALSE;
......@@ -266,10 +235,12 @@ gst_speed_convert (GstPad * pad, GstFormat src_format, gint64 src_value,
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
scale = filter->sample_size;
scale = GST_AUDIO_INFO_BPF (&filter->info);
/* fallthrough */
case GST_FORMAT_DEFAULT:
*dest_value = src_value * scale * filter->rate / GST_SECOND;
*dest_value =
src_value * scale * GST_AUDIO_INFO_RATE (&filter->info) /
GST_SECOND;
break;
default:
ret = FALSE;
......@@ -318,7 +289,7 @@ speed_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
cur);
/* convert to time format */
if (!gst_speed_convert (pad, rformat, cur, &conv_format, &cur)) {
if (!gst_speed_convert (filter, rformat, cur, &conv_format, &cur)) {
ret = FALSE;
break;
}
......@@ -327,7 +298,7 @@ speed_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
cur /= filter->speed;
/* convert to time format */
if (!gst_speed_convert (pad, conv_format, cur, &format, &cur)) {
if (!gst_speed_convert (filter, conv_format, cur, &format, &cur)) {
ret = FALSE;
break;
}
......@@ -365,7 +336,7 @@ speed_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
end);
/* convert to time format */
if (!gst_speed_convert (pad, rformat, end, &conv_format, &end)) {
if (!gst_speed_convert (filter, rformat, end, &conv_format, &end)) {
ret = FALSE;
break;
}
......@@ -374,7 +345,7 @@ speed_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
end /= filter->speed;
/* convert to time format */
if (!gst_speed_convert (pad, conv_format, end, &format, &end)) {
if (!gst_speed_convert (filter, conv_format, end, &format, &end)) {
ret = FALSE;
break;
}
......@@ -447,8 +418,6 @@ gst_speed_init (GstSpeed * filter)
filter->offset = 0;
filter->timestamp = 0;
filter->sample_size = 0;
}
static inline guint
......@@ -474,10 +443,11 @@ speed_chain_int16 (GstSpeed * filter, GstBuffer * in_buf, GstBuffer * out_buf,
while (i < in_samples) {
interp = i_float - floor (i_float);
out_data[j * filter->channels] =
lower * (1 - interp) + in_data[i * filter->channels] * interp;
out_data[j * GST_AUDIO_INFO_CHANNELS (&filter->info)] =
lower * (1 - interp) +
in_data[i * GST_AUDIO_INFO_CHANNELS (&filter->info)] * interp;
lower = in_data[i * filter->channels];
lower = in_data[i * GST_AUDIO_INFO_CHANNELS (&filter->info)];
i_float += filter->speed;
i = (guint) ceil (i_float);
......@@ -513,10 +483,11 @@ speed_chain_float32 (GstSpeed * filter, GstBuffer * in_buf, GstBuffer * out_buf,
while (i < in_samples) {
interp = i_float - floor (i_float);
out_data[j * filter->channels] =
lower * (1 - interp) + in_data[i * filter->channels] * interp;
out_data[j * GST_AUDIO_INFO_CHANNELS (&filter->info)] =
lower * (1 - interp) +
in_data[i * GST_AUDIO_INFO_CHANNELS (&filter->info)] * interp;
lower = in_data[i * filter->channels];
lower = in_data[i * GST_AUDIO_INFO_CHANNELS (&filter->info)];
i_float += filter->speed;
i = (guint) ceil (i_float);
......@@ -532,11 +503,9 @@ speed_chain_float32 (GstSpeed * filter, GstBuffer * in_buf, GstBuffer * out_buf,
static gboolean
speed_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstSpeed *filter;
GstSpeed *filter = GST_SPEED (parent);
gboolean ret = FALSE;
filter = GST_SPEED (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:{
gdouble rate;
......@@ -607,60 +576,51 @@ static GstFlowReturn
speed_chain (GstPad * pad, GstObject * parent, GstBuffer * in_buf)
{
GstBuffer *out_buf;
GstSpeed *filter;
GstSpeed *filter = GST_SPEED (parent);
guint c, in_samples, out_samples, out_size;
GstFlowReturn flow;
gsize size;
filter = GST_SPEED (parent);
if (G_UNLIKELY (filter->sample_size == 0 || filter->rate == 0)) {
flow = GST_FLOW_NOT_NEGOTIATED;
goto done;
}
if (G_UNLIKELY (filter->offset == GST_BUFFER_OFFSET_NONE)) {
filter->offset =
gst_util_uint64_scale_int (filter->timestamp, filter->rate, GST_SECOND);
filter->offset = gst_util_uint64_scale_int (filter->timestamp,
GST_AUDIO_INFO_RATE (&filter->info), GST_SECOND);
}
/* buffersize has to be aligned by samplesize */
/* buffersize has to be aligned to a frame */
out_size = ceil ((gfloat) gst_buffer_get_size (in_buf) / filter->speed);
out_size = ((out_size + filter->sample_size - 1) / filter->sample_size) *
filter->sample_size;
out_size = ((out_size + GST_AUDIO_INFO_BPF (&filter->info) - 1) /
GST_AUDIO_INFO_BPF (&filter->info)) * GST_AUDIO_INFO_BPF (&filter->info);
out_buf = gst_buffer_new_and_alloc (out_size);
in_samples = gst_buffer_get_size (in_buf) / filter->sample_size;
in_samples = gst_buffer_get_size (in_buf) /
GST_AUDIO_INFO_BPS (&filter->info);
out_samples = 0;
for (c = 0; c < filter->channels; ++c) {
if (filter->format == GST_SPEED_FORMAT_INT) {
for (c = 0; c < GST_AUDIO_INFO_CHANNELS (&filter->info); ++c) {
if (GST_AUDIO_INFO_IS_INTEGER (&filter->info))
out_samples = speed_chain_int16 (filter, in_buf, out_buf, c, in_samples);
} else {
else
out_samples =
speed_chain_float32 (filter, in_buf, out_buf, c, in_samples);
}
}
size = out_samples * filter->sample_size;
size = out_samples * GST_AUDIO_INFO_BPS (&filter->info);
gst_buffer_set_size (out_buf, size);
GST_BUFFER_OFFSET (out_buf) = filter->offset;
GST_BUFFER_TIMESTAMP (out_buf) = filter->timestamp;
filter->offset += size / filter->sample_size;
filter->timestamp =
gst_util_uint64_scale_int (filter->offset, GST_SECOND, filter->rate);
filter->offset += size / GST_AUDIO_INFO_BPS (&filter->info);
filter->timestamp = gst_util_uint64_scale_int (filter->offset, GST_SECOND,
GST_AUDIO_INFO_RATE (&filter->info));
/* make sure it's at least nominally a perfect stream */
GST_BUFFER_DURATION (out_buf) =
filter->timestamp - GST_BUFFER_TIMESTAMP (out_buf);
flow = gst_pad_push (filter->srcpad, out_buf);
done:
if (G_UNLIKELY (flow != GST_FLOW_OK))
GST_DEBUG_OBJECT (filter, "flow: %s", gst_flow_get_name (flow));
......@@ -711,15 +671,14 @@ speed_change_state (GstElement * element, GstStateChange transition)
case GST_STATE_CHANGE_READY_TO_PAUSED:
speed->offset = GST_BUFFER_OFFSET_NONE;
speed->timestamp = 0;
gst_audio_info_init (&speed->info);
break;
default:
break;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
return GST_STATE_CHANGE_SUCCESS;
return GST_ELEMENT_CLASS (gst_speed_parent_class)->change_state (element,
transition);
}
static gboolean
......
......@@ -40,11 +40,6 @@ G_BEGIN_DECLS
typedef struct _GstSpeed GstSpeed;
typedef struct _GstSpeedClass GstSpeedClass;
enum _GstSpeedFormat {
GST_SPEED_FORMAT_INT,
GST_SPEED_FORMAT_FLOAT
};
typedef enum _GstSpeedFormat GstSpeedFormat;
struct _GstSpeed {
......@@ -58,12 +53,7 @@ struct _GstSpeed {
gint64 offset;
gint64 timestamp;
guint rate;
guint channels;
guint width;
guint sample_size;
GstSpeedFormat format;
GstAudioInfo info;
};
struct _GstSpeedClass {
......
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