/* -*- c-basic-offset: 2 -*- * GStreamer * Copyright (C) 1999-2001 Erik Walthinsen * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library 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 * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:element-speed * * Plays an audio stream at a different speed (by resampling the audio). * * Do not use this element. Either use the 'pitch' element, or do a seek with * a non-1.0 rate parameter, this will have the same effect as using the speed * element (but relies on the decoder/demuxer to handle this correctly, also * requires a fairly up-to-date gst-plugins-base, as of February 2007). * * * Example launch line * |[ * gst-launch-1.0 filesrc location=test.ogg ! decodebin ! audioconvert ! speed speed=1.5 ! audioconvert ! audioresample ! autoaudiosink * ]| Plays an .ogg file at 1.5x speed. * */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include #include "gstspeed.h" GST_DEBUG_CATEGORY_STATIC (speed_debug); #define GST_CAT_DEFAULT speed_debug enum { PROP_0, PROP_SPEED }; /* assumption here: sizeof (gfloat) = 4 */ #define GST_SPEED_AUDIO_CAPS \ "audio/x-raw, " \ "format = {" GST_AUDIO_NE (F32) ", " GST_AUDIO_NE (S16) "}, " \ "rate = (int) [ 1, MAX ], " \ "channels = (int) [ 1, MAX ]" static GstStaticPadTemplate gst_speed_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS (GST_SPEED_AUDIO_CAPS) ); static GstStaticPadTemplate gst_speed_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (GST_SPEED_AUDIO_CAPS) ); static void speed_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void speed_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static gboolean speed_parse_caps (GstSpeed * filter, const GstCaps * caps); static GstFlowReturn speed_chain (GstPad * pad, GstObject * parent, GstBuffer * buf); static GstStateChangeReturn speed_change_state (GstElement * element, GstStateChange transition); static gboolean speed_sink_event (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean speed_src_event (GstPad * pad, GstObject * parent, GstEvent * event); G_DEFINE_TYPE (GstSpeed, gst_speed, GST_TYPE_ELEMENT); static gboolean speed_setcaps (GstPad * pad, GstCaps * caps) { GstSpeed *filter; gboolean ret; filter = GST_SPEED (gst_pad_get_parent (pad)); ret = speed_parse_caps (filter, caps); gst_object_unref (filter); return ret; } static gboolean speed_parse_caps (GstSpeed * filter, const GstCaps * caps) { g_return_val_if_fail (filter != NULL, FALSE); g_return_val_if_fail (caps != NULL, FALSE); if (!gst_audio_info_from_caps (&filter->info, caps)) return FALSE; return TRUE; } static gboolean speed_src_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstSpeed *filter; gboolean ret = FALSE; filter = GST_SPEED (parent); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEEK:{ gdouble rate; GstFormat format; GstSeekFlags flags; GstSeekType start_type, stop_type; gint64 start, stop; gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start, &stop_type, &stop); gst_event_unref (event); if (format != GST_FORMAT_TIME) { GST_DEBUG_OBJECT (filter, "only support seeks in TIME format"); break; } if (start_type != GST_SEEK_TYPE_NONE && start != -1) { start *= filter->speed; } if (stop_type != GST_SEEK_TYPE_NONE && stop != -1) { stop *= filter->speed; } event = gst_event_new_seek (rate, format, flags, start_type, start, stop_type, stop); GST_LOG ("sending seek event: %" GST_PTR_FORMAT, gst_event_get_structure (event)); ret = gst_pad_send_event (GST_PAD_PEER (filter->sinkpad), event); break; } default: ret = gst_pad_event_default (pad, parent, event); break; } return ret; } static gboolean gst_speed_convert (GstSpeed * filter, GstFormat src_format, gint64 src_value, GstFormat * dest_format, gint64 * dest_value) { gboolean ret = TRUE; guint scale = 1; if (src_format == *dest_format) { *dest_value = src_value; return TRUE; } switch (src_format) { case GST_FORMAT_BYTES: switch (*dest_format) { case GST_FORMAT_DEFAULT: if (GST_AUDIO_INFO_BPF (&filter->info) == 0) { ret = FALSE; break; } *dest_value = src_value / GST_AUDIO_INFO_BPF (&filter->info); break; case GST_FORMAT_TIME: { gint byterate = GST_AUDIO_INFO_BPF (&filter->info) * GST_AUDIO_INFO_RATE (&filter->info); if (byterate == 0) { ret = FALSE; break; } *dest_value = src_value * GST_SECOND / byterate; break; } default: ret = FALSE; } break; case GST_FORMAT_DEFAULT: switch (*dest_format) { case GST_FORMAT_BYTES: *dest_value = src_value * GST_AUDIO_INFO_BPF (&filter->info); break; case GST_FORMAT_TIME: if (GST_AUDIO_INFO_RATE (&filter->info) == 0) { ret = FALSE; break; } *dest_value = src_value * GST_SECOND / GST_AUDIO_INFO_RATE (&filter->info); break; default: ret = FALSE; } break; case GST_FORMAT_TIME: switch (*dest_format) { case GST_FORMAT_BYTES: scale = GST_AUDIO_INFO_BPF (&filter->info); /* fallthrough */ case GST_FORMAT_DEFAULT: *dest_value = src_value * scale * GST_AUDIO_INFO_RATE (&filter->info) / GST_SECOND; break; default: ret = FALSE; } break; default: ret = FALSE; } return ret; } static gboolean speed_src_query (GstPad * pad, GstObject * parent, GstQuery * query) { gboolean ret = TRUE; GstSpeed *filter; filter = GST_SPEED (parent); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_POSITION: { GstFormat format; GstFormat rformat = GST_FORMAT_TIME; gint64 cur; GstFormat conv_format = GST_FORMAT_TIME; /* save requested format */ gst_query_parse_position (query, &format, NULL); /* query peer for current position in time */ gst_query_set_position (query, GST_FORMAT_TIME, -1); if (!gst_pad_peer_query_position (filter->sinkpad, rformat, &cur)) { GST_LOG_OBJECT (filter, "TIME query on peer pad failed, trying BYTES"); rformat = GST_FORMAT_BYTES; if (!gst_pad_peer_query_position (filter->sinkpad, rformat, &cur)) { GST_LOG_OBJECT (filter, "BYTES query on peer pad failed too"); goto error; } } if (rformat == GST_FORMAT_BYTES) GST_LOG_OBJECT (filter, "peer pad returned current=%" G_GINT64_FORMAT " bytes", cur); else if (rformat == GST_FORMAT_TIME) GST_LOG_OBJECT (filter, "peer pad returned time=%" G_GINT64_FORMAT, cur); /* convert to time format */ if (!gst_speed_convert (filter, rformat, cur, &conv_format, &cur)) { ret = FALSE; break; } /* adjust for speed factor */ cur /= filter->speed; /* convert to time format */ if (!gst_speed_convert (filter, conv_format, cur, &format, &cur)) { ret = FALSE; break; } gst_query_set_position (query, format, cur); GST_LOG_OBJECT (filter, "position query: we return %" G_GUINT64_FORMAT " (format %u)", cur, format); break; } case GST_QUERY_DURATION: { GstFormat format; GstFormat rformat = GST_FORMAT_TIME; gint64 end; GstFormat conv_format = GST_FORMAT_TIME; /* save requested format */ gst_query_parse_duration (query, &format, NULL); /* query peer for total length in time */ gst_query_set_duration (query, GST_FORMAT_TIME, -1); if (!gst_pad_peer_query_duration (filter->sinkpad, rformat, &end)) { GST_LOG_OBJECT (filter, "TIME query on peer pad failed, trying BYTES"); rformat = GST_FORMAT_BYTES; if (!gst_pad_peer_query_duration (filter->sinkpad, rformat, &end)) { GST_LOG_OBJECT (filter, "BYTES query on peer pad failed too"); goto error; } } if (rformat == GST_FORMAT_BYTES) GST_LOG_OBJECT (filter, "peer pad returned total=%" G_GINT64_FORMAT " bytes", end); else if (rformat == GST_FORMAT_TIME) GST_LOG_OBJECT (filter, "peer pad returned time=%" G_GINT64_FORMAT, end); /* convert to time format */ if (!gst_speed_convert (filter, rformat, end, &conv_format, &end)) { ret = FALSE; break; } /* adjust for speed factor */ end /= filter->speed; /* convert to time format */ if (!gst_speed_convert (filter, conv_format, end, &format, &end)) { ret = FALSE; break; } gst_query_set_duration (query, format, end); GST_LOG_OBJECT (filter, "duration query: we return %" G_GUINT64_FORMAT " (format %u)", end, format); break; } default: ret = FALSE; break; } return ret; error: gst_object_unref (filter); GST_DEBUG ("error handling query"); return FALSE; } static void gst_speed_class_init (GstSpeedClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstElementClass *gstelement_class = (GstElementClass *) klass; gobject_class->set_property = speed_set_property; gobject_class->get_property = speed_get_property; gstelement_class->change_state = speed_change_state; g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SPEED, g_param_spec_float ("speed", "speed", "speed", 0.1f, 40.0, 1.0, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); gst_element_class_set_static_metadata (gstelement_class, "Speed", "Filter/Effect/Audio", "Set speed/pitch on audio/raw streams (resampler)", "Andy Wingo , " "Tim-Philipp Müller "); gst_element_class_add_static_pad_template (gstelement_class, &gst_speed_src_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_speed_sink_template); } static void gst_speed_init (GstSpeed * filter) { filter->sinkpad = gst_pad_new_from_static_template (&gst_speed_sink_template, "sink"); gst_pad_set_chain_function (filter->sinkpad, speed_chain); gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad); gst_pad_set_event_function (filter->sinkpad, speed_sink_event); GST_PAD_SET_PROXY_CAPS (filter->sinkpad); filter->srcpad = gst_pad_new_from_static_template (&gst_speed_src_template, "src"); gst_pad_set_query_function (filter->srcpad, speed_src_query); gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad); gst_pad_set_event_function (filter->srcpad, speed_src_event); GST_PAD_SET_PROXY_CAPS (filter->srcpad); filter->offset = 0; filter->timestamp = 0; } static inline guint speed_chain_int16 (GstSpeed * filter, GstBuffer * in_buf, GstBuffer * out_buf, guint c, guint in_samples) { gint16 *in_data, *out_data; gfloat interp, lower, i_float; guint i, j; GstMapInfo in_info, out_info; gst_buffer_map (in_buf, &in_info, GST_MAP_READ); gst_buffer_map (out_buf, &out_info, GST_MAP_WRITE); in_data = (gint16 *) in_info.data + c; out_data = (gint16 *) out_info.data + c; lower = in_data[0]; i_float = 0.5 * (filter->speed - 1.0); i = (guint) ceil (i_float); j = 0; while (i < in_samples) { interp = i_float - floor (i_float); 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 * GST_AUDIO_INFO_CHANNELS (&filter->info)]; i_float += filter->speed; i = (guint) ceil (i_float); ++j; } gst_buffer_unmap (in_buf, &in_info); gst_buffer_unmap (out_buf, &out_info); return j; } static inline guint speed_chain_float32 (GstSpeed * filter, GstBuffer * in_buf, GstBuffer * out_buf, guint c, guint in_samples) { gfloat *in_data, *out_data; gfloat interp, lower, i_float; guint i, j; GstMapInfo in_info, out_info; gst_buffer_map (in_buf, &in_info, GST_MAP_WRITE); gst_buffer_map (out_buf, &out_info, GST_MAP_WRITE); in_data = (gfloat *) in_info.data + c; out_data = (gfloat *) out_info.data + c; lower = in_data[0]; i_float = 0.5 * (filter->speed - 1.0); i = (guint) ceil (i_float); j = 0; while (i < in_samples) { interp = i_float - floor (i_float); 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 * GST_AUDIO_INFO_CHANNELS (&filter->info)]; i_float += filter->speed; i = (guint) ceil (i_float); ++j; } gst_buffer_unmap (in_buf, &in_info); gst_buffer_unmap (out_buf, &out_info); return j; } static gboolean speed_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstSpeed *filter = GST_SPEED (parent); gboolean ret = FALSE; switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEGMENT:{ gdouble rate; GstFormat format; gint64 start_value, stop_value, base; const GstSegment *segment; GstSegment seg; gst_event_parse_segment (event, &segment); rate = segment->rate; format = segment->format; start_value = segment->start; stop_value = segment->stop; base = segment->base; gst_event_unref (event); if (format != GST_FORMAT_TIME) { GST_WARNING_OBJECT (filter, "newsegment event not in TIME format!"); break; } g_assert (filter->speed > 0); if (start_value >= 0) start_value /= filter->speed; if (stop_value >= 0) stop_value /= filter->speed; base /= filter->speed; /* this would only really be correct if we clipped incoming data */ filter->timestamp = start_value; /* set to NONE so it gets reset later based on the timestamp when we have * the samplerate */ filter->offset = GST_BUFFER_OFFSET_NONE; gst_segment_init (&seg, GST_FORMAT_TIME); seg.rate = rate; seg.start = start_value; seg.stop = stop_value; seg.time = segment->time; ret = gst_pad_push_event (filter->srcpad, gst_event_new_segment (&seg)); break; } case GST_EVENT_CAPS: { GstCaps *caps; gst_event_parse_caps (event, &caps); ret = speed_setcaps (pad, caps); if (!ret) { gst_event_unref (event); return ret; } } /* Fallthrough so that the caps event gets forwarded */ default: ret = gst_pad_event_default (pad, parent, event); break; } return ret; } static GstFlowReturn speed_chain (GstPad * pad, GstObject * parent, GstBuffer * in_buf) { GstBuffer *out_buf; GstSpeed *filter = GST_SPEED (parent); guint c, in_samples, out_samples, out_size; GstFlowReturn flow; gsize size; if (G_UNLIKELY (filter->offset == GST_BUFFER_OFFSET_NONE)) { filter->offset = gst_util_uint64_scale_int (filter->timestamp, GST_AUDIO_INFO_RATE (&filter->info), GST_SECOND); } /* buffersize has to be aligned to a frame */ out_size = ceil ((gfloat) gst_buffer_get_size (in_buf) / filter->speed); 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) / GST_AUDIO_INFO_BPF (&filter->info); out_samples = 0; 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 out_samples = speed_chain_float32 (filter, in_buf, out_buf, c, in_samples); } size = out_samples * GST_AUDIO_INFO_BPF (&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 / GST_AUDIO_INFO_BPF (&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); if (G_UNLIKELY (flow != GST_FLOW_OK)) GST_DEBUG_OBJECT (filter, "flow: %s", gst_flow_get_name (flow)); gst_buffer_unref (in_buf); return flow; } static void speed_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstSpeed *filter = GST_SPEED (object); switch (prop_id) { case PROP_SPEED: filter->speed = g_value_get_float (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void speed_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstSpeed *filter = GST_SPEED (object); switch (prop_id) { case PROP_SPEED: g_value_set_float (value, filter->speed); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstStateChangeReturn speed_change_state (GstElement * element, GstStateChange transition) { GstSpeed *speed = GST_SPEED (element); switch (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; } return GST_ELEMENT_CLASS (gst_speed_parent_class)->change_state (element, transition); } static gboolean plugin_init (GstPlugin * plugin) { GST_DEBUG_CATEGORY_INIT (speed_debug, "speed", 0, "speed element"); return gst_element_register (plugin, "speed", GST_RANK_NONE, GST_TYPE_SPEED); } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, speed, "Set speed/pitch on audio/raw streams (resampler)", plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)