/* GStreamer * Copyright (C) <2011> Stefan Kost * * gstaudiovisualizer.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:gstaudiovisualizer * * A baseclass 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 /* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex * with newer GLib versions (>= 2.31.0) */ #define GLIB_DISABLE_DEPRECATION_WARNINGS #include #include "gstaudiovisualizer.h" GST_DEBUG_CATEGORY_STATIC (audio_visualizer_debug); #define GST_CAT_DEFAULT (audio_visualizer_debug) #define DEFAULT_SHADER GST_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_audio_visualizer_class_init (GstAudioVisualizerClass * klass); static void gst_audio_visualizer_init (GstAudioVisualizer * scope, GstAudioVisualizerClass * g_class); static void gst_audio_visualizer_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_audio_visualizer_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_audio_visualizer_dispose (GObject * object); static gboolean gst_audio_visualizer_src_negotiate (GstAudioVisualizer * scope); static gboolean gst_audio_visualizer_src_setcaps (GstAudioVisualizer * scope, GstCaps * caps); static gboolean gst_audio_visualizer_sink_setcaps (GstAudioVisualizer * scope, GstCaps * caps); static GstFlowReturn gst_audio_visualizer_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer); static gboolean gst_audio_visualizer_src_event (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean gst_audio_visualizer_sink_event (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean gst_audio_visualizer_src_query (GstPad * pad, GstObject * parent, GstQuery * query); static gboolean gst_audio_visualizer_sink_query (GstPad * pad, GstObject * parent, GstQuery * query); static GstStateChangeReturn gst_audio_visualizer_change_state (GstElement * element, GstStateChange transition); /* shading functions */ #define GST_TYPE_AUDIO_VISUALIZER_SHADER (gst_audio_visualizer_shader_get_type()) static GType gst_audio_visualizer_shader_get_type (void) { static GType shader_type = 0; static const GEnumValue shaders[] = { {GST_AUDIO_VISUALIZER_SHADER_NONE, "None", "none"}, {GST_AUDIO_VISUALIZER_SHADER_FADE, "Fade", "fade"}, {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_UP, "Fade and move up", "fade-and-move-up"}, {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_DOWN, "Fade and move down", "fade-and-move-down"}, {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_LEFT, "Fade and move left", "fade-and-move-left"}, {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_RIGHT, "Fade and move right", "fade-and-move-right"}, {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_OUT, "Fade and move horizontally out", "fade-and-move-horiz-out"}, {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_IN, "Fade and move horizontally in", "fade-and-move-horiz-in"}, {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_OUT, "Fade and move vertically out", "fade-and-move-vert-out"}, {GST_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)) { /* TODO: rename when exporting it as a library */ shader_type = g_enum_register_static ("GstAudioVisualizerShader-BadGstAudioVisualizers", 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 (GstAudioVisualizer * 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 (GstAudioVisualizer * 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 (GstAudioVisualizer * 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 (GstAudioVisualizer * 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 (GstAudioVisualizer * 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 (GstAudioVisualizer * 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 (GstAudioVisualizer * 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 (GstAudioVisualizer * 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 (GstAudioVisualizer * 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_audio_visualizer_change_shader (GstAudioVisualizer * scope) { switch (scope->shader_type) { case GST_AUDIO_VISUALIZER_SHADER_NONE: scope->shader = NULL; break; case GST_AUDIO_VISUALIZER_SHADER_FADE: scope->shader = shader_fade; break; case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_UP: scope->shader = shader_fade_and_move_up; break; case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_DOWN: scope->shader = shader_fade_and_move_down; break; case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_LEFT: scope->shader = shader_fade_and_move_left; break; case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_RIGHT: scope->shader = shader_fade_and_move_right; break; case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_OUT: scope->shader = shader_fade_and_move_horiz_out; break; case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_IN: scope->shader = shader_fade_and_move_horiz_in; break; case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_OUT: scope->shader = shader_fade_and_move_vert_out; break; case GST_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_audio_visualizer_get_type (void) { static volatile gsize audio_visualizer_type = 0; if (g_once_init_enter (&audio_visualizer_type)) { static const GTypeInfo audio_visualizer_info = { sizeof (GstAudioVisualizerClass), NULL, NULL, (GClassInitFunc) gst_audio_visualizer_class_init, NULL, NULL, sizeof (GstAudioVisualizer), 0, (GInstanceInitFunc) gst_audio_visualizer_init, }; GType _type; /* TODO: rename when exporting it as a library */ _type = g_type_register_static (GST_TYPE_ELEMENT, "GstAudioVisualizer-BadGstAudioVisualizers", &audio_visualizer_info, G_TYPE_FLAG_ABSTRACT); g_once_init_leave (&audio_visualizer_type, _type); } return (GType) audio_visualizer_type; } static void gst_audio_visualizer_class_init (GstAudioVisualizerClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstElementClass *element_class = (GstElementClass *) klass; parent_class = g_type_class_peek_parent (klass); GST_DEBUG_CATEGORY_INIT (audio_visualizer_debug, "baseaudiovisualizer", 0, "scope audio visualisation base class"); gobject_class->set_property = gst_audio_visualizer_set_property; gobject_class->get_property = gst_audio_visualizer_get_property; gobject_class->dispose = gst_audio_visualizer_dispose; element_class->change_state = GST_DEBUG_FUNCPTR (gst_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_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_audio_visualizer_init (GstAudioVisualizer * scope, GstAudioVisualizerClass * 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_audio_visualizer_chain)); gst_pad_set_event_function (scope->sinkpad, GST_DEBUG_FUNCPTR (gst_audio_visualizer_sink_event)); gst_pad_set_query_function (scope->sinkpad, GST_DEBUG_FUNCPTR (gst_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_audio_visualizer_src_event)); gst_pad_set_query_function (scope->srcpad, GST_DEBUG_FUNCPTR (gst_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_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 state */ gst_audio_info_init (&scope->ainfo); gst_video_info_init (&scope->vinfo); g_mutex_init (&scope->config_lock); } static void gst_audio_visualizer_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioVisualizer *scope = GST_AUDIO_VISUALIZER (object); switch (prop_id) { case PROP_SHADER: scope->shader_type = g_value_get_enum (value); gst_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_audio_visualizer_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioVisualizer *scope = GST_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_audio_visualizer_dispose (GObject * object) { GstAudioVisualizer *scope = GST_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.p) { g_mutex_clear (&scope->config_lock); scope->config_lock.p = NULL; } G_OBJECT_CLASS (parent_class)->dispose (object); } static void gst_audio_visualizer_reset (GstAudioVisualizer * 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_audio_visualizer_sink_setcaps (GstAudioVisualizer * scope, GstCaps * caps) { GstAudioInfo info; gboolean res = TRUE; if (!gst_audio_info_from_caps (&info, caps)) goto wrong_caps; scope->ainfo = info; GST_DEBUG_OBJECT (scope, "audio: channels %d, rate %d", GST_AUDIO_INFO_CHANNELS (&info), GST_AUDIO_INFO_RATE (&info)); done: return res; /* Errors */ wrong_caps: { GST_WARNING_OBJECT (scope, "could not parse caps"); res = FALSE; goto done; } } static gboolean gst_audio_visualizer_src_setcaps (GstAudioVisualizer * scope, GstCaps * caps) { GstVideoInfo info; GstAudioVisualizerClass *klass; GstStructure *structure; gboolean res; if (!gst_video_info_from_caps (&info, caps)) goto wrong_caps; 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 wrong_caps; klass = GST_AUDIO_VISUALIZER_CLASS (G_OBJECT_GET_CLASS (scope)); scope->vinfo = info; scope->video_format = info.finfo->format; scope->frame_duration = gst_util_uint64_scale_int (GST_SECOND, scope->fps_d, scope->fps_n); scope->spf = gst_util_uint64_scale_int (GST_AUDIO_INFO_RATE (&scope->ainfo), 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_set_caps (scope->srcpad, caps); return res; /* ERRORS */ wrong_caps: { GST_DEBUG_OBJECT (scope, "error parsing caps"); return FALSE; } } static gboolean gst_audio_visualizer_src_negotiate (GstAudioVisualizer * scope) { GstCaps *othercaps, *target; GstStructure *structure; GstCaps *templ; GstQuery *query; GstBufferPool *pool; GstStructure *config; guint size, min, max; 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; target = gst_caps_truncate (target); } else { target = templ; } target = gst_caps_make_writable (target); 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_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)) { /* not a problem, we use the query defaults */ GST_DEBUG_OBJECT (scope, "allocation query failed"); } if (gst_query_get_n_allocation_pools (query) > 0) { /* we got configuration from our peer, parse them */ gst_query_parse_nth_allocation_pool (query, 0, &pool, &size, &min, &max); } else { pool = NULL; size = scope->bpf; min = max = 0; } if (pool == NULL) { /* 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_params (config, target, size, min, max); gst_buffer_pool_set_config (pool, config); if (scope->pool) { gst_buffer_pool_set_active (scope->pool, FALSE); 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_audio_visualizer_ensure_negotiated (GstAudioVisualizer * 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_audio_visualizer_src_negotiate (scope)) return GST_FLOW_NOT_NEGOTIATED; } return GST_FLOW_OK; } static GstFlowReturn gst_audio_visualizer_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstFlowReturn ret = GST_FLOW_OK; GstAudioVisualizer *scope; GstAudioVisualizerClass *klass; GstBuffer *inbuf; guint64 dist, ts; guint avail, sbpf; gpointer adata; gboolean (*render) (GstAudioVisualizer * scope, GstBuffer * audio, GstBuffer * video); gint bps, channels, rate; scope = GST_AUDIO_VISUALIZER (parent); klass = GST_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); } /* Make sure have an output format */ ret = gst_audio_visualizer_ensure_negotiated (scope); if (ret != GST_FLOW_OK) { gst_buffer_unref (buffer); goto beach; } channels = GST_AUDIO_INFO_CHANNELS (&scope->ainfo); rate = GST_AUDIO_INFO_RATE (&scope->ainfo); bps = GST_AUDIO_INFO_BPS (&scope->ainfo); if (bps == 0) { ret = GST_FLOW_NOT_NEGOTIATED; goto beach; } gst_adapter_push (scope->adapter, buffer); g_mutex_lock (&scope->config_lock); /* this is what we want */ sbpf = scope->req_spf * 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; GstMapInfo map; /* 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 /= bps; ts += gst_util_uint64_scale_int (dist, GST_SECOND, 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 * 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; gst_buffer_map (outbuf, &map, GST_MAP_WRITE); if (scope->shader) { memcpy (map.data, scope->pixelbuf, scope->bpf); } else { memset (map.data, 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_replace_all_memory (inbuf, gst_memory_new_wrapped (GST_MEMORY_FLAG_READONLY, adata, sbpf, 0, sbpf, NULL, NULL)); /* 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, map.data, scope->pixelbuf); } } } gst_buffer_unmap (outbuf, &map); gst_buffer_resize (outbuf, 0, 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 * 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) { /* 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_audio_visualizer_src_event (GstPad * pad, GstObject * parent, GstEvent * event) { gboolean res; GstAudioVisualizer *scope; scope = GST_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, ×tamp); /* 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; } case GST_EVENT_RECONFIGURE: /* dont't forward */ gst_event_unref (event); res = TRUE; break; default: res = gst_pad_push_event (scope->sinkpad, event); break; } return res; } static gboolean gst_audio_visualizer_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) { gboolean res; GstAudioVisualizer *scope; scope = GST_AUDIO_VISUALIZER (parent); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CAPS: { GstCaps *caps; gst_event_parse_caps (event, &caps); res = gst_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_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_audio_visualizer_src_query (GstPad * pad, GstObject * parent, GstQuery * query) { gboolean res = FALSE; GstAudioVisualizer *scope; scope = GST_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; gint rate = GST_AUDIO_INFO_RATE (&scope->ainfo); if (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, 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_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_audio_visualizer_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret; GstAudioVisualizer *scope; scope = GST_AUDIO_VISUALIZER (element); switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: gst_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; }