mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-27 20:21:24 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/audiovisualizers/gstbaseaudiovisualizer.c
Wim Taymans 940de5b195 add parent to pad functions
2011-11-17 15:53:42 +01:00

1133 lines
33 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) <2011> Stefan Kost <ensonic@users.sf.net>
*
* gstbaseaudiovisualizer.h: base class for audio visualisation elements
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
/**
* SECTION:gstbaseaudiovisualizer
*
* A basclass for scopes (visualizers). It takes care of re-fitting the
* audio-rate to video-rate and handles renegotiation (downstream video size
* changes).
*
* It also provides several background shading effects. These effects are
* applied to a previous picture before the render() implementation can draw a
* new frame.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstbaseaudiovisualizer.h"
GST_DEBUG_CATEGORY_STATIC (base_audio_visualizer_debug);
#define GST_CAT_DEFAULT (base_audio_visualizer_debug)
#define DEFAULT_SHADER GST_BASE_AUDIO_VISUALIZER_SHADER_FADE
#define DEFAULT_SHADE_AMOUNT 0x000a0a0a
enum
{
PROP_0,
PROP_SHADER,
PROP_SHADE_AMOUNT
};
static GstBaseTransformClass *parent_class = NULL;
static void gst_base_audio_visualizer_class_init (GstBaseAudioVisualizerClass *
klass);
static void gst_base_audio_visualizer_init (GstBaseAudioVisualizer * scope,
GstBaseAudioVisualizerClass * g_class);
static void gst_base_audio_visualizer_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_base_audio_visualizer_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static void gst_base_audio_visualizer_dispose (GObject * object);
static gboolean gst_base_audio_visualizer_src_negotiate (GstBaseAudioVisualizer
* scope);
static gboolean gst_base_audio_visualizer_src_setcaps (GstBaseAudioVisualizer *
scope, GstCaps * caps);
static gboolean gst_base_audio_visualizer_sink_setcaps (GstBaseAudioVisualizer *
scope, GstCaps * caps);
static GstFlowReturn gst_base_audio_visualizer_chain (GstPad * pad,
GstObject * parent, GstBuffer * buffer);
static gboolean gst_base_audio_visualizer_src_event (GstPad * pad,
GstObject * parent, GstEvent * event);
static gboolean gst_base_audio_visualizer_sink_event (GstPad * pad,
GstObject * parent, GstEvent * event);
static gboolean gst_base_audio_visualizer_src_query (GstPad * pad,
GstObject * parent, GstQuery * query);
static gboolean gst_base_audio_visualizer_sink_query (GstPad * pad,
GstObject * parent, GstQuery * query);
static GstStateChangeReturn gst_base_audio_visualizer_change_state (GstElement *
element, GstStateChange transition);
/* shading functions */
#define GST_TYPE_BASE_AUDIO_VISUALIZER_SHADER (gst_base_audio_visualizer_shader_get_type())
static GType
gst_base_audio_visualizer_shader_get_type (void)
{
static GType shader_type = 0;
static const GEnumValue shaders[] = {
{GST_BASE_AUDIO_VISUALIZER_SHADER_NONE, "None", "none"},
{GST_BASE_AUDIO_VISUALIZER_SHADER_FADE, "Fade", "fade"},
{GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_UP, "Fade and move up",
"fade-and-move-up"},
{GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_DOWN, "Fade and move down",
"fade-and-move-down"},
{GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_LEFT, "Fade and move left",
"fade-and-move-left"},
{GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_RIGHT,
"Fade and move right",
"fade-and-move-right"},
{GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_OUT,
"Fade and move horizontally out", "fade-and-move-horiz-out"},
{GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_IN,
"Fade and move horizontally in", "fade-and-move-horiz-in"},
{GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_OUT,
"Fade and move vertically out", "fade-and-move-vert-out"},
{GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_IN,
"Fade and move vertically in", "fade-and-move-vert-in"},
{0, NULL, NULL},
};
if (G_UNLIKELY (shader_type == 0)) {
shader_type =
g_enum_register_static ("GstBaseAudioVisualizerShader", shaders);
}
return shader_type;
}
/* we're only supporting GST_VIDEO_FORMAT_xRGB right now) */
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define SHADE1(_d, _s, _i, _r, _g, _b) \
G_STMT_START { \
_d[_i] = (_s[_i] > _b) ? _s[_i] - _b : 0; \
_i++; \
_d[_i] = (_s[_i] > _g) ? _s[_i] - _g : 0; \
_i++; \
_d[_i] = (_s[_i] > _r) ? _s[_i] - _r : 0; \
_i++; \
_d[_i++] = 0; \
} G_STMT_END
#define SHADE2(_d, _s, _j, _i, _r, _g, _b) \
G_STMT_START { \
_d[_j++] = (_s[_i] > _b) ? _s[_i] - _b : 0; \
_i++; \
_d[_j++] = (_s[_i] > _g) ? _s[_i] - _g : 0; \
_i++; \
_d[_j++] = (_s[_i] > _r) ? _s[_i] - _r : 0; \
_i++; \
_d[_j++] = 0; \
_i++; \
} G_STMT_END
#else
#define SHADE1(_d, _s, _i, _r, _g, _b) \
G_STMT_START { \
_d[_i++] = 0; \
_d[_i] = (_s[_i] > _r) ? _s[_i] - _r : 0; \
_i++; \
_d[_i] = (_s[_i] > _g) ? _s[_i] - _g : 0; \
_i++; \
_d[_i] = (_s[_i] > _b) ? _s[_i] - _b : 0; \
_i++; \
} G_STMT_END
#define SHADE2(_d, _s, _j, _i, _r, _g, _b) \
G_STMT_START { \
_d[_j++] = 0; \
_i++; \
_d[_j++] = (_s[_i] > _r) ? _s[_i] - _r : 0; \
_i++; \
_d[_j++] = (_s[_i] > _g) ? _s[_i] - _g : 0; \
_i++; \
_d[_j++] = (_s[_i] > _b) ? _s[_i] - _b : 0; \
_i++; \
} G_STMT_END
#endif
static void
shader_fade (GstBaseAudioVisualizer * scope, const guint8 * s, guint8 * d)
{
guint i, bpf = scope->bpf;
guint r = (scope->shade_amount >> 16) & 0xff;
guint g = (scope->shade_amount >> 8) & 0xff;
guint b = (scope->shade_amount >> 0) & 0xff;
for (i = 0; i < bpf;) {
SHADE1 (d, s, i, r, g, b);
}
}
static void
shader_fade_and_move_up (GstBaseAudioVisualizer * scope, const guint8 * s,
guint8 * d)
{
guint i, j, bpf = scope->bpf;
guint bpl = 4 * scope->width;
guint r = (scope->shade_amount >> 16) & 0xff;
guint g = (scope->shade_amount >> 8) & 0xff;
guint b = (scope->shade_amount >> 0) & 0xff;
for (j = 0, i = bpl; i < bpf;) {
SHADE2 (d, s, j, i, r, g, b);
}
}
static void
shader_fade_and_move_down (GstBaseAudioVisualizer * scope, const guint8 * s,
guint8 * d)
{
guint i, j, bpf = scope->bpf;
guint bpl = 4 * scope->width;
guint r = (scope->shade_amount >> 16) & 0xff;
guint g = (scope->shade_amount >> 8) & 0xff;
guint b = (scope->shade_amount >> 0) & 0xff;
for (j = bpl, i = 0; j < bpf;) {
SHADE2 (d, s, j, i, r, g, b);
}
}
static void
shader_fade_and_move_left (GstBaseAudioVisualizer * scope,
const guint8 * s, guint8 * d)
{
guint i, j, k, bpf = scope->bpf;
guint w = scope->width;
guint r = (scope->shade_amount >> 16) & 0xff;
guint g = (scope->shade_amount >> 8) & 0xff;
guint b = (scope->shade_amount >> 0) & 0xff;
/* move to the left */
for (j = 0, i = 4; i < bpf;) {
for (k = 0; k < w - 1; k++) {
SHADE2 (d, s, j, i, r, g, b);
}
i += 4;
j += 4;
}
}
static void
shader_fade_and_move_right (GstBaseAudioVisualizer * scope,
const guint8 * s, guint8 * d)
{
guint i, j, k, bpf = scope->bpf;
guint w = scope->width;
guint r = (scope->shade_amount >> 16) & 0xff;
guint g = (scope->shade_amount >> 8) & 0xff;
guint b = (scope->shade_amount >> 0) & 0xff;
/* move to the left */
for (j = 4, i = 0; i < bpf;) {
for (k = 0; k < w - 1; k++) {
SHADE2 (d, s, j, i, r, g, b);
}
i += 4;
j += 4;
}
}
static void
shader_fade_and_move_horiz_out (GstBaseAudioVisualizer * scope,
const guint8 * s, guint8 * d)
{
guint i, j, bpf = scope->bpf / 2;
guint bpl = 4 * scope->width;
guint r = (scope->shade_amount >> 16) & 0xff;
guint g = (scope->shade_amount >> 8) & 0xff;
guint b = (scope->shade_amount >> 0) & 0xff;
/* move upper half up */
for (j = 0, i = bpl; i < bpf;) {
SHADE2 (d, s, j, i, r, g, b);
}
/* move lower half down */
for (j = bpf + bpl, i = bpf; j < bpf + bpf;) {
SHADE2 (d, s, j, i, r, g, b);
}
}
static void
shader_fade_and_move_horiz_in (GstBaseAudioVisualizer * scope,
const guint8 * s, guint8 * d)
{
guint i, j, bpf = scope->bpf / 2;
guint bpl = 4 * scope->width;
guint r = (scope->shade_amount >> 16) & 0xff;
guint g = (scope->shade_amount >> 8) & 0xff;
guint b = (scope->shade_amount >> 0) & 0xff;
/* move upper half down */
for (i = 0, j = bpl; i < bpf;) {
SHADE2 (d, s, j, i, r, g, b);
}
/* move lower half up */
for (i = bpf + bpl, j = bpf; i < bpf + bpf;) {
SHADE2 (d, s, j, i, r, g, b);
}
}
static void
shader_fade_and_move_vert_out (GstBaseAudioVisualizer * scope,
const guint8 * s, guint8 * d)
{
guint i, j, k, bpf = scope->bpf;
guint m = scope->width / 2;
guint r = (scope->shade_amount >> 16) & 0xff;
guint g = (scope->shade_amount >> 8) & 0xff;
guint b = (scope->shade_amount >> 0) & 0xff;
/* move left half to the left */
for (j = 0, i = 4; i < bpf;) {
for (k = 0; k < m; k++) {
SHADE2 (d, s, j, i, r, g, b);
}
j += 4 * m;
i += 4 * m;
}
/* move right half to the right */
for (j = 4 * (m + 1), i = 4 * m; j < bpf;) {
for (k = 0; k < m; k++) {
SHADE2 (d, s, j, i, r, g, b);
}
j += 4 * m;
i += 4 * m;
}
}
static void
shader_fade_and_move_vert_in (GstBaseAudioVisualizer * scope,
const guint8 * s, guint8 * d)
{
guint i, j, k, bpf = scope->bpf;
guint m = scope->width / 2;
guint r = (scope->shade_amount >> 16) & 0xff;
guint g = (scope->shade_amount >> 8) & 0xff;
guint b = (scope->shade_amount >> 0) & 0xff;
/* move left half to the right */
for (j = 4, i = 0; j < bpf;) {
for (k = 0; k < m; k++) {
SHADE2 (d, s, j, i, r, g, b);
}
j += 4 * m;
i += 4 * m;
}
/* move right half to the left */
for (j = 4 * m, i = 4 * (m + 1); i < bpf;) {
for (k = 0; k < m; k++) {
SHADE2 (d, s, j, i, r, g, b);
}
j += 4 * m;
i += 4 * m;
}
}
static void
gst_base_audio_visualizer_change_shader (GstBaseAudioVisualizer * scope)
{
switch (scope->shader_type) {
case GST_BASE_AUDIO_VISUALIZER_SHADER_NONE:
scope->shader = NULL;
break;
case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE:
scope->shader = shader_fade;
break;
case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_UP:
scope->shader = shader_fade_and_move_up;
break;
case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_DOWN:
scope->shader = shader_fade_and_move_down;
break;
case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_LEFT:
scope->shader = shader_fade_and_move_left;
break;
case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_RIGHT:
scope->shader = shader_fade_and_move_right;
break;
case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_OUT:
scope->shader = shader_fade_and_move_horiz_out;
break;
case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_IN:
scope->shader = shader_fade_and_move_horiz_in;
break;
case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_OUT:
scope->shader = shader_fade_and_move_vert_out;
break;
case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_IN:
scope->shader = shader_fade_and_move_vert_in;
break;
default:
GST_ERROR ("invalid shader function");
scope->shader = NULL;
break;
}
}
/* base class */
GType
gst_base_audio_visualizer_get_type (void)
{
static volatile gsize base_audio_visualizer_type = 0;
if (g_once_init_enter (&base_audio_visualizer_type)) {
static const GTypeInfo base_audio_visualizer_info = {
sizeof (GstBaseAudioVisualizerClass),
NULL,
NULL,
(GClassInitFunc) gst_base_audio_visualizer_class_init,
NULL,
NULL,
sizeof (GstBaseAudioVisualizer),
0,
(GInstanceInitFunc) gst_base_audio_visualizer_init,
};
GType _type;
_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstBaseAudioVisualizer", &base_audio_visualizer_info,
G_TYPE_FLAG_ABSTRACT);
g_once_init_leave (&base_audio_visualizer_type, _type);
}
return (GType) base_audio_visualizer_type;
}
static void
gst_base_audio_visualizer_class_init (GstBaseAudioVisualizerClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *element_class = (GstElementClass *) klass;
parent_class = g_type_class_peek_parent (klass);
GST_DEBUG_CATEGORY_INIT (base_audio_visualizer_debug, "baseaudiovisualizer",
0, "scope audio visualisation base class");
gobject_class->set_property = gst_base_audio_visualizer_set_property;
gobject_class->get_property = gst_base_audio_visualizer_get_property;
gobject_class->dispose = gst_base_audio_visualizer_dispose;
element_class->change_state =
GST_DEBUG_FUNCPTR (gst_base_audio_visualizer_change_state);
g_object_class_install_property (gobject_class, PROP_SHADER,
g_param_spec_enum ("shader", "shader type",
"Shader function to apply on each frame",
GST_TYPE_BASE_AUDIO_VISUALIZER_SHADER, DEFAULT_SHADER,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SHADE_AMOUNT,
g_param_spec_uint ("shade-amount", "shade amount",
"Shading color to use (big-endian ARGB)", 0, G_MAXUINT32,
DEFAULT_SHADE_AMOUNT,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
}
static void
gst_base_audio_visualizer_init (GstBaseAudioVisualizer * scope,
GstBaseAudioVisualizerClass * g_class)
{
GstPadTemplate *pad_template;
/* create the sink and src pads */
pad_template =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "sink");
g_return_if_fail (pad_template != NULL);
scope->sinkpad = gst_pad_new_from_template (pad_template, "sink");
gst_pad_set_chain_function (scope->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_audio_visualizer_chain));
gst_pad_set_event_function (scope->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_audio_visualizer_sink_event));
gst_pad_set_query_function (scope->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_audio_visualizer_sink_query));
gst_element_add_pad (GST_ELEMENT (scope), scope->sinkpad);
pad_template =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "src");
g_return_if_fail (pad_template != NULL);
scope->srcpad = gst_pad_new_from_template (pad_template, "src");
gst_pad_set_event_function (scope->srcpad,
GST_DEBUG_FUNCPTR (gst_base_audio_visualizer_src_event));
gst_pad_set_query_function (scope->srcpad,
GST_DEBUG_FUNCPTR (gst_base_audio_visualizer_src_query));
gst_element_add_pad (GST_ELEMENT (scope), scope->srcpad);
scope->adapter = gst_adapter_new ();
scope->inbuf = gst_buffer_new ();
/* properties */
scope->shader_type = DEFAULT_SHADER;
gst_base_audio_visualizer_change_shader (scope);
scope->shade_amount = DEFAULT_SHADE_AMOUNT;
/* reset the initial video state */
scope->width = 320;
scope->height = 200;
scope->fps_n = 25; /* desired frame rate */
scope->fps_d = 1;
scope->frame_duration = GST_CLOCK_TIME_NONE;
/* reset the initial audio state */
scope->rate = GST_AUDIO_DEF_RATE;
scope->channels = 2;
scope->config_lock = g_mutex_new ();
}
static void
gst_base_audio_visualizer_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstBaseAudioVisualizer *scope = GST_BASE_AUDIO_VISUALIZER (object);
switch (prop_id) {
case PROP_SHADER:
scope->shader_type = g_value_get_enum (value);
gst_base_audio_visualizer_change_shader (scope);
break;
case PROP_SHADE_AMOUNT:
scope->shade_amount = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_base_audio_visualizer_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstBaseAudioVisualizer *scope = GST_BASE_AUDIO_VISUALIZER (object);
switch (prop_id) {
case PROP_SHADER:
g_value_set_enum (value, scope->shader_type);
break;
case PROP_SHADE_AMOUNT:
g_value_set_uint (value, scope->shade_amount);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_base_audio_visualizer_dispose (GObject * object)
{
GstBaseAudioVisualizer *scope = GST_BASE_AUDIO_VISUALIZER (object);
if (scope->adapter) {
g_object_unref (scope->adapter);
scope->adapter = NULL;
}
if (scope->inbuf) {
gst_buffer_unref (scope->inbuf);
scope->inbuf = NULL;
}
if (scope->pixelbuf) {
g_free (scope->pixelbuf);
scope->pixelbuf = NULL;
}
if (scope->config_lock) {
g_mutex_free (scope->config_lock);
scope->config_lock = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_base_audio_visualizer_reset (GstBaseAudioVisualizer * scope)
{
gst_adapter_clear (scope->adapter);
gst_segment_init (&scope->segment, GST_FORMAT_UNDEFINED);
GST_OBJECT_LOCK (scope);
scope->proportion = 1.0;
scope->earliest_time = -1;
GST_OBJECT_UNLOCK (scope);
}
static gboolean
gst_base_audio_visualizer_sink_setcaps (GstBaseAudioVisualizer * scope,
GstCaps * caps)
{
GstStructure *structure;
gint channels;
gint rate;
gboolean res = TRUE;
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "channels", &channels) ||
!gst_structure_get_int (structure, "rate", &rate))
goto missing_caps_details;
if (channels != 2)
goto wrong_channels;
if (rate <= 0)
goto wrong_rate;
scope->channels = channels;
scope->rate = rate;
GST_DEBUG_OBJECT (scope, "audio: channels %d, rate %d",
scope->channels, scope->rate);
done:
return res;
/* Errors */
missing_caps_details:
{
GST_WARNING_OBJECT (scope, "missing channels or rate in the caps");
res = FALSE;
goto done;
}
wrong_channels:
{
GST_WARNING_OBJECT (scope, "number of channels must be 2, but is %d",
channels);
res = FALSE;
goto done;
}
wrong_rate:
{
GST_WARNING_OBJECT (scope, "sample rate must be >0, but is %d", rate);
res = FALSE;
goto done;
}
}
static gboolean
gst_base_audio_visualizer_src_setcaps (GstBaseAudioVisualizer * scope,
GstCaps * caps)
{
GstBaseAudioVisualizerClass *klass;
GstStructure *structure;
gboolean res;
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "width", &scope->width) ||
!gst_structure_get_int (structure, "height", &scope->height) ||
!gst_structure_get_fraction (structure, "framerate", &scope->fps_n,
&scope->fps_d))
goto error;
klass = GST_BASE_AUDIO_VISUALIZER_CLASS (G_OBJECT_GET_CLASS (scope));
//scope->video_format = format; ??
scope->frame_duration = gst_util_uint64_scale_int (GST_SECOND,
scope->fps_d, scope->fps_n);
scope->spf = gst_util_uint64_scale_int (scope->rate,
scope->fps_d, scope->fps_n);
scope->req_spf = scope->spf;
scope->bpf = scope->width * scope->height * 4;
if (scope->pixelbuf)
g_free (scope->pixelbuf);
scope->pixelbuf = g_malloc0 (scope->bpf);
if (klass->setup)
res = klass->setup (scope);
GST_DEBUG_OBJECT (scope, "video: dimension %dx%d, framerate %d/%d",
scope->width, scope->height, scope->fps_n, scope->fps_d);
GST_DEBUG_OBJECT (scope, "blocks: spf %u, req_spf %u",
scope->spf, scope->req_spf);
res = gst_pad_push_event (scope->srcpad, gst_event_new_caps (caps));
return res;
/* ERRORS */
error:
{
GST_DEBUG_OBJECT (scope, "error parsing caps");
return FALSE;
}
}
static gboolean
gst_base_audio_visualizer_src_negotiate (GstBaseAudioVisualizer * scope)
{
GstCaps *othercaps, *target;
GstStructure *structure;
GstCaps *templ;
GstQuery *query;
GstBufferPool *pool = NULL;
guint size, min, max, prefix, alignment;
templ = gst_pad_get_pad_template_caps (scope->srcpad);
GST_DEBUG_OBJECT (scope, "performing negotiation");
/* see what the peer can do */
othercaps = gst_pad_peer_query_caps (scope->srcpad, NULL);
if (othercaps) {
target = gst_caps_intersect (othercaps, templ);
gst_caps_unref (othercaps);
gst_caps_unref (templ);
if (gst_caps_is_empty (target))
goto no_format;
gst_caps_truncate (target);
} else {
target = gst_caps_ref ((GstCaps *) templ);
}
structure = gst_caps_get_structure (target, 0);
gst_structure_fixate_field_nearest_int (structure, "width", scope->width);
gst_structure_fixate_field_nearest_int (structure, "height", scope->height);
gst_structure_fixate_field_nearest_fraction (structure, "framerate",
scope->fps_n, scope->fps_d);
GST_DEBUG_OBJECT (scope, "final caps are %" GST_PTR_FORMAT, target);
gst_base_audio_visualizer_src_setcaps (scope, target);
/* try to get a bufferpool now */
/* find a pool for the negotiated caps now */
query = gst_query_new_allocation (target, TRUE);
if (gst_pad_peer_query (scope->srcpad, query)) {
/* we got configuration from our peer, parse them */
gst_query_parse_allocation_params (query, &size, &min, &max, &prefix,
&alignment, &pool);
} else {
size = scope->bpf;
min = max = 0;
prefix = 0;
alignment = 0;
}
if (pool == NULL) {
GstStructure *config;
/* we did not get a pool, make one ourselves then */
pool = gst_buffer_pool_new ();
config = gst_buffer_pool_get_config (pool);
gst_buffer_pool_config_set (config, target, size, min, max, prefix,
alignment);
gst_buffer_pool_set_config (pool, config);
}
if (scope->pool)
gst_object_unref (scope->pool);
scope->pool = pool;
/* and activate */
gst_buffer_pool_set_active (pool, TRUE);
gst_caps_unref (target);
return TRUE;
no_format:
{
gst_caps_unref (target);
return FALSE;
}
}
/* make sure we are negotiated */
static GstFlowReturn
gst_base_audio_visualizer_ensure_negotiated (GstBaseAudioVisualizer * scope)
{
gboolean reconfigure;
reconfigure = gst_pad_check_reconfigure (scope->srcpad);
/* we don't know an output format yet, pick one */
if (reconfigure || !gst_pad_has_current_caps (scope->srcpad)) {
if (!gst_base_audio_visualizer_src_negotiate (scope))
return GST_FLOW_NOT_NEGOTIATED;
}
return GST_FLOW_OK;
}
static GstFlowReturn
gst_base_audio_visualizer_chain (GstPad * pad, GstObject * parent,
GstBuffer * buffer)
{
GstFlowReturn ret = GST_FLOW_OK;
GstBaseAudioVisualizer *scope;
GstBaseAudioVisualizerClass *klass;
GstBuffer *inbuf;
guint64 dist, ts;
guint avail, sbpf;
gpointer adata, vdata;
gboolean (*render) (GstBaseAudioVisualizer * scope, GstBuffer * audio,
GstBuffer * video);
scope = GST_BASE_AUDIO_VISUALIZER (parent);
klass = GST_BASE_AUDIO_VISUALIZER_CLASS (G_OBJECT_GET_CLASS (scope));
render = klass->render;
GST_LOG_OBJECT (scope, "chainfunc called");
/* resync on DISCONT */
if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
gst_adapter_clear (scope->adapter);
}
if (scope->bps == 0) {
ret = GST_FLOW_NOT_NEGOTIATED;
goto beach;
}
/* Make sure have an output format */
ret = gst_base_audio_visualizer_ensure_negotiated (scope);
if (ret != GST_FLOW_OK) {
gst_buffer_unref (buffer);
goto beach;
}
gst_adapter_push (scope->adapter, buffer);
g_mutex_lock (scope->config_lock);
/* this is what we want */
sbpf = scope->req_spf * scope->channels * sizeof (gint16);
inbuf = scope->inbuf;
/* FIXME: the timestamp in the adapter would be different */
gst_buffer_copy_into (inbuf, buffer, GST_BUFFER_COPY_METADATA, 0, -1);
/* this is what we have */
avail = gst_adapter_available (scope->adapter);
GST_LOG_OBJECT (scope, "avail: %u, bpf: %u", avail, sbpf);
while (avail >= sbpf) {
GstBuffer *outbuf;
/* get timestamp of the current adapter content */
ts = gst_adapter_prev_timestamp (scope->adapter, &dist);
if (GST_CLOCK_TIME_IS_VALID (ts)) {
/* convert bytes to time */
dist /= scope->bps;
ts += gst_util_uint64_scale_int (dist, GST_SECOND, scope->rate);
}
if (GST_CLOCK_TIME_IS_VALID (ts)) {
gint64 qostime;
gboolean need_skip;
qostime =
gst_segment_to_running_time (&scope->segment, GST_FORMAT_TIME, ts) +
scope->frame_duration;
GST_OBJECT_LOCK (scope);
/* check for QoS, don't compute buffers that are known to be late */
need_skip = scope->earliest_time != -1 && qostime <= scope->earliest_time;
GST_OBJECT_UNLOCK (scope);
if (need_skip) {
GST_WARNING_OBJECT (scope,
"QoS: skip ts: %" GST_TIME_FORMAT ", earliest: %" GST_TIME_FORMAT,
GST_TIME_ARGS (qostime), GST_TIME_ARGS (scope->earliest_time));
goto skip;
}
}
g_mutex_unlock (scope->config_lock);
ret = gst_buffer_pool_acquire_buffer (scope->pool, &outbuf, NULL);
g_mutex_lock (scope->config_lock);
/* recheck as the value could have changed */
sbpf = scope->req_spf * scope->channels * sizeof (gint16);
/* no buffer allocated, we don't care why. */
if (ret != GST_FLOW_OK)
break;
/* sync controlled properties */
gst_object_sync_values (GST_OBJECT (scope), ts);
GST_BUFFER_TIMESTAMP (outbuf) = ts;
GST_BUFFER_DURATION (outbuf) = scope->frame_duration;
vdata = gst_buffer_map (outbuf, NULL, NULL, GST_MAP_WRITE);
if (scope->shader) {
memcpy (vdata, scope->pixelbuf, scope->bpf);
} else {
memset (vdata, 0, scope->bpf);
}
/* this can fail as the data size we need could have changed */
if (!(adata = (gpointer) gst_adapter_map (scope->adapter, sbpf)))
break;
gst_buffer_take_memory (inbuf, -1,
gst_memory_new_wrapped (GST_MEMORY_FLAG_READONLY, adata, NULL, sbpf, 0,
sbpf));
/* call class->render() vmethod */
if (render) {
if (!render (scope, inbuf, outbuf)) {
ret = GST_FLOW_ERROR;
} else {
/* run various post processing (shading and geometri transformation */
if (scope->shader) {
scope->shader (scope, vdata, scope->pixelbuf);
}
}
}
gst_buffer_unmap (outbuf, vdata, scope->bpf);
g_mutex_unlock (scope->config_lock);
ret = gst_pad_push (scope->srcpad, outbuf);
outbuf = NULL;
g_mutex_lock (scope->config_lock);
skip:
/* recheck as the value could have changed */
sbpf = scope->req_spf * scope->channels * sizeof (gint16);
GST_LOG_OBJECT (scope, "avail: %u, bpf: %u", avail, sbpf);
/* we want to take less or more, depending on spf : req_spf */
if (avail - sbpf >= sbpf) {
gst_adapter_flush (scope->adapter, sbpf);
gst_adapter_unmap (scope->adapter);
} else if (avail - sbpf >= 0) {
/* just flush a bit and stop */
gst_adapter_flush (scope->adapter, (avail - sbpf));
gst_adapter_unmap (scope->adapter);
break;
}
avail = gst_adapter_available (scope->adapter);
if (ret != GST_FLOW_OK)
break;
}
g_mutex_unlock (scope->config_lock);
beach:
return ret;
}
static gboolean
gst_base_audio_visualizer_src_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
gboolean res;
GstBaseAudioVisualizer *scope;
scope = GST_BASE_AUDIO_VISUALIZER (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_QOS:
{
gdouble proportion;
GstClockTimeDiff diff;
GstClockTime timestamp;
gst_event_parse_qos (event, NULL, &proportion, &diff, &timestamp);
/* save stuff for the _chain() function */
GST_OBJECT_LOCK (scope);
scope->proportion = proportion;
if (diff >= 0)
/* we're late, this is a good estimate for next displayable
* frame (see part-qos.txt) */
scope->earliest_time = timestamp + 2 * diff + scope->frame_duration;
else
scope->earliest_time = timestamp + diff;
GST_OBJECT_UNLOCK (scope);
res = gst_pad_push_event (scope->sinkpad, event);
break;
}
default:
res = gst_pad_push_event (scope->sinkpad, event);
break;
}
return res;
}
static gboolean
gst_base_audio_visualizer_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
gboolean res;
GstBaseAudioVisualizer *scope;
scope = GST_BASE_AUDIO_VISUALIZER (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
res = gst_base_audio_visualizer_sink_setcaps (scope, caps);
break;
}
case GST_EVENT_FLUSH_START:
res = gst_pad_push_event (scope->srcpad, event);
break;
case GST_EVENT_FLUSH_STOP:
gst_base_audio_visualizer_reset (scope);
res = gst_pad_push_event (scope->srcpad, event);
break;
case GST_EVENT_SEGMENT:
{
/* the newsegment values are used to clip the input samples
* and to convert the incomming timestamps to running time so
* we can do QoS */
gst_event_copy_segment (event, &scope->segment);
res = gst_pad_push_event (scope->srcpad, event);
break;
}
default:
res = gst_pad_push_event (scope->srcpad, event);
break;
}
return res;
}
static gboolean
gst_base_audio_visualizer_src_query (GstPad * pad, GstObject * parent,
GstQuery * query)
{
gboolean res = FALSE;
GstBaseAudioVisualizer *scope;
scope = GST_BASE_AUDIO_VISUALIZER (parent);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
/* We need to send the query upstream and add the returned latency to our
* own */
GstClockTime min_latency, max_latency;
gboolean us_live;
GstClockTime our_latency;
guint max_samples;
if (scope->rate == 0)
break;
if ((res = gst_pad_peer_query (scope->sinkpad, query))) {
gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
GST_DEBUG_OBJECT (scope, "Peer latency: min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
/* the max samples we must buffer buffer */
max_samples = MAX (scope->req_spf, scope->spf);
our_latency =
gst_util_uint64_scale_int (max_samples, GST_SECOND, scope->rate);
GST_DEBUG_OBJECT (scope, "Our latency: %" GST_TIME_FORMAT,
GST_TIME_ARGS (our_latency));
/* we add some latency but only if we need to buffer more than what
* upstream gives us */
min_latency += our_latency;
if (max_latency != -1)
max_latency += our_latency;
GST_DEBUG_OBJECT (scope, "Calculated total latency : min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
gst_query_set_latency (query, TRUE, min_latency, max_latency);
}
break;
}
default:
res = gst_pad_query_default (pad, parent, query);
break;
}
return res;
}
static gboolean
gst_base_audio_visualizer_sink_query (GstPad * pad, GstObject * parent,
GstQuery * query)
{
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
default:
res = gst_pad_query_default (pad, parent, query);
break;
}
return res;
}
static GstStateChangeReturn
gst_base_audio_visualizer_change_state (GstElement * element,
GstStateChange transition)
{
GstStateChangeReturn ret;
GstBaseAudioVisualizer *scope;
scope = GST_BASE_AUDIO_VISUALIZER (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_base_audio_visualizer_reset (scope);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
if (scope->pool) {
gst_buffer_pool_set_active (scope->pool, FALSE);
gst_object_replace ((GstObject **) & scope->pool, NULL);
}
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
Reference in a new issue View git blame Copy permalink