/* GStreamer * Copyright (C) 2003 Benjamin Otte * * gstaudioconvert.c: Convert audio to different audio formats automatically * * 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., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ /* Element-Checklist-Version: 5 */ /* * design decisions: * - audioconvert converts buffers in a set of supported caps. If it supports * a caps, it supports conversion from these caps to any other caps it * supports. (example: if it does A=>B and A=>C, it also does B=>C) * - audioconvert does not save state between buffers. Every incoming buffer is * converted and the converted buffer is pushed out. * conclusion: * audioconvert is not supposed to be a one-element-does-anything solution for * audio conversions. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include "gstchannelmix.h" #include "plugin.h" GST_DEBUG_CATEGORY (audio_convert_debug); /*** DEFINITIONS **************************************************************/ static GstElementDetails audio_convert_details = { "Audio Conversion", "Filter/Converter/Audio", "Convert audio to different formats", "Benjamin Otte ", }; /* type functions */ static void gst_audio_convert_base_init (gpointer g_class); static void gst_audio_convert_class_init (GstAudioConvertClass * klass); static void gst_audio_convert_init (GstAudioConvert * audio_convert); static void gst_audio_convert_dispose (GObject * obj); /* gstreamer functions */ static GstFlowReturn gst_audio_convert_chain (GstPad * pad, GstBuffer * buffer); static gboolean gst_audio_convert_link_src (GstAudioConvert * this, GstCaps * sinkcaps, GstAudioConvertCaps * sink_ac_caps); static gboolean gst_audio_convert_setcaps (GstPad * pad, GstCaps * caps); static void gst_audio_convert_fixate (GstPad * pad, GstCaps * caps); static GstCaps *gst_audio_convert_getcaps (GstPad * pad); static GstElementStateReturn gst_audio_convert_change_state (GstElement * element); static GstBuffer *gst_audio_convert_buffer_to_default_format (GstAudioConvert * this, GstBuffer * buf); static GstBuffer *gst_audio_convert_buffer_from_default_format (GstAudioConvert * this, GstBuffer * buf); static GstBuffer *gst_audio_convert_channels (GstAudioConvert * this, GstBuffer * buf); /* AudioConvert signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { ARG_0, ARG_AGGRESSIVE }; #define DEBUG_INIT(bla) \ GST_DEBUG_CATEGORY_INIT (audio_convert_debug, "audioconvert", 0, "audio conversion element"); GST_BOILERPLATE_FULL (GstAudioConvert, gst_audio_convert, GstElement, GST_TYPE_ELEMENT, DEBUG_INIT); /*** GSTREAMER PROTOTYPES *****************************************************/ #define STATIC_CAPS \ GST_STATIC_CAPS ( \ "audio/x-raw-float, " \ "rate = (int) [ 1, MAX ], " \ "channels = (int) [ 1, 8 ], " \ "endianness = (int) BYTE_ORDER, " \ "width = (int) 32, " \ "buffer-frames = (int) [ 0, MAX ];" \ "audio/x-raw-int, " \ "rate = (int) [ 1, MAX ], " \ "channels = (int) [ 1, 8 ], " \ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \ "width = (int) 32, " \ "depth = (int) [ 1, 32 ], " \ "signed = (boolean) { true, false }; " \ "audio/x-raw-int, " \ "rate = (int) [ 1, MAX ], " \ "channels = (int) [ 1, 8 ], " \ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \ "width = (int) 24, " \ "depth = (int) [ 1, 24 ], " \ "signed = (boolean) { true, false }; " \ "audio/x-raw-int, " \ "rate = (int) [ 1, MAX ], " \ "channels = (int) [ 1, 8 ], " \ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \ "width = (int) 16, " \ "depth = (int) [ 1, 16 ], " \ "signed = (boolean) { true, false }; " \ "audio/x-raw-int, " \ "rate = (int) [ 1, MAX ], " \ "channels = (int) [ 1, 8 ], " \ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \ "width = (int) 8, " \ "depth = (int) [ 1, 8 ], " \ "signed = (boolean) { true, false } " \ ) static GstAudioChannelPosition *supported_positions; static GstStaticPadTemplate gst_audio_convert_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, STATIC_CAPS); static GstStaticPadTemplate gst_audio_convert_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, STATIC_CAPS); /*** TYPE FUNCTIONS ***********************************************************/ static void gst_audio_convert_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_audio_convert_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_audio_convert_sink_template)); gst_element_class_set_details (element_class, &audio_convert_details); } static void gst_audio_convert_class_init (GstAudioConvertClass * klass) { GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); GObjectClass *gobject_class = G_OBJECT_CLASS (klass); gint i; gstelement_class->change_state = gst_audio_convert_change_state; gobject_class->dispose = gst_audio_convert_dispose; supported_positions = g_new0 (GstAudioChannelPosition, GST_AUDIO_CHANNEL_POSITION_NUM); for (i = 0; i < GST_AUDIO_CHANNEL_POSITION_NUM; i++) supported_positions[i] = i; } static void gst_audio_convert_init (GstAudioConvert * this) { /* sinkpad */ this->sink = gst_pad_new_from_template (gst_static_pad_template_get (&gst_audio_convert_sink_template), "sink"); gst_pad_set_getcaps_function (this->sink, gst_audio_convert_getcaps); gst_pad_set_setcaps_function (this->sink, gst_audio_convert_setcaps); gst_pad_set_fixatecaps_function (this->sink, gst_audio_convert_fixate); gst_element_add_pad (GST_ELEMENT (this), this->sink); /* srcpad */ this->src = gst_pad_new_from_template (gst_static_pad_template_get (&gst_audio_convert_src_template), "src"); gst_pad_set_getcaps_function (this->src, gst_audio_convert_getcaps); //gst_pad_set_setcaps_function (this->src, gst_audio_convert_setcaps); gst_pad_set_fixatecaps_function (this->src, gst_audio_convert_fixate); gst_element_add_pad (GST_ELEMENT (this), this->src); gst_pad_set_chain_function (this->sink, gst_audio_convert_chain); /* clear important variables */ this->convert_internal = NULL; this->sinkcaps.pos = NULL; this->srccaps.pos = NULL; this->matrix = NULL; } static void gst_audio_convert_dispose (GObject * obj) { GstAudioConvert *this = GST_AUDIO_CONVERT (obj); if (this->sinkcaps.pos) { g_free (this->sinkcaps.pos); this->sinkcaps.pos = NULL; } if (this->srccaps.pos) { g_free (this->srccaps.pos); this->srccaps.pos = NULL; } G_OBJECT_CLASS (parent_class)->dispose (obj); } /*** GSTREAMER FUNCTIONS ******************************************************/ static GstFlowReturn gst_audio_convert_chain (GstPad * pad, GstBuffer * buf) { GstAudioConvert *this; GstFlowReturn ret; this = GST_AUDIO_CONVERT (GST_OBJECT_PARENT (pad)); /** * Theory of operation: * - convert the format (endianness, signedness, width, depth) to * (G_BYTE_ORDER, TRUE, 32, 32) * - convert rate and channels * - convert back to output format */ if (!GST_PAD_CAPS (this->sink)) { goto not_negotiated; } else if (!GST_PAD_CAPS (this->src)) { if (!gst_audio_convert_link_src (this, GST_PAD_CAPS (this->sink), &this->sinkcaps)) goto no_format; } else if (!this->matrix) { gst_audio_convert_setup_matrix (this); } buf = gst_audio_convert_buffer_to_default_format (this, buf); buf = gst_audio_convert_channels (this, buf); buf = gst_audio_convert_buffer_from_default_format (this, buf); ret = gst_pad_push (this->src, buf); return ret; not_negotiated: { GST_ELEMENT_ERROR (this, CORE, NEGOTIATION, (NULL), ("Pad not negotiated before chain function was called")); gst_buffer_unref (buf); return GST_FLOW_NOT_NEGOTIATED; } no_format: { GST_ELEMENT_ERROR (this, CORE, NEGOTIATION, (NULL), ("Could not negotiate format")); gst_buffer_unref (buf); return GST_FLOW_ERROR; } } static GstCaps * gst_audio_convert_caps_remove_format_info (GstPad * pad, GstCaps * caps) { int i, size; GstAudioConvert *this; this = GST_AUDIO_CONVERT (GST_OBJECT_PARENT (pad)); size = gst_caps_get_size (caps); caps = gst_caps_make_writable (caps); for (i = size - 1; i >= 0; i--) { GstStructure *structure; structure = gst_caps_get_structure (caps, i); gst_structure_remove_field (structure, "channels"); gst_structure_remove_field (structure, "channel-positions"); gst_structure_remove_field (structure, "endianness"); gst_structure_remove_field (structure, "width"); gst_structure_remove_field (structure, "depth"); gst_structure_remove_field (structure, "signed"); structure = gst_structure_copy (structure); if (strcmp (gst_structure_get_name (structure), "audio/x-raw-int") == 0) { gst_structure_set_name (structure, "audio/x-raw-float"); if (pad == this->sink) { gst_structure_set (structure, "buffer-frames", GST_TYPE_INT_RANGE, 0, G_MAXINT, NULL); } else { gst_structure_set (structure, "buffer-frames", G_TYPE_INT, 0, NULL); } } else { gst_structure_set_name (structure, "audio/x-raw-int"); gst_structure_remove_field (structure, "buffer-frames"); } gst_caps_append_structure (caps, structure); } return caps; } /* this function is complicated now, but it will be unnecessary when we convert * rate. */ static GstCaps * gst_audio_convert_getcaps (GstPad * pad) { GstAudioConvert *this; GstPad *otherpad; GstCaps *othercaps, *caps; const GstCaps *templcaps; this = GST_AUDIO_CONVERT (GST_OBJECT_PARENT (pad)); otherpad = (pad == this->src) ? this->sink : this->src; /* we can do all our peer can */ othercaps = gst_pad_peer_get_caps (otherpad); if (othercaps != NULL) { /* without the format info even */ othercaps = gst_audio_convert_caps_remove_format_info (pad, othercaps); /* but filtered against our template */ templcaps = gst_pad_get_pad_template_caps (pad); caps = gst_caps_intersect (othercaps, templcaps); gst_caps_unref (othercaps); } else { /* no peer, then our template is enough */ caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad)); } /* Get the channel positions in as well. */ gst_audio_set_caps_channel_positions_list (caps, supported_positions, GST_AUDIO_CHANNEL_POSITION_NUM); return caps; } static gboolean gst_audio_convert_parse_caps (const GstCaps * gst_caps, GstAudioConvertCaps * caps) { GstStructure *structure = gst_caps_get_structure (gst_caps, 0); GST_DEBUG ("parse caps %p and %" GST_PTR_FORMAT, gst_caps, gst_caps); g_return_val_if_fail (gst_caps_is_fixed (gst_caps), FALSE); g_return_val_if_fail (caps != NULL, FALSE); /* cleanup old */ if (caps->pos) { g_free (caps->pos); caps->pos = NULL; } caps->endianness = G_BYTE_ORDER; caps->is_int = (strcmp (gst_structure_get_name (structure), "audio/x-raw-int") == 0); if (!gst_structure_get_int (structure, "channels", &caps->channels) || !(caps->pos = gst_audio_get_channel_positions (structure)) || !gst_structure_get_int (structure, "width", &caps->width) || !gst_structure_get_int (structure, "rate", &caps->rate) || (caps->is_int && (!gst_structure_get_boolean (structure, "signed", &caps->sign) || !gst_structure_get_int (structure, "depth", &caps->depth) || (caps->width != 8 && !gst_structure_get_int (structure, "endianness", &caps->endianness)))) || (!caps->is_int && !gst_structure_get_int (structure, "buffer-frames", &caps->buffer_frames))) { GST_DEBUG ("could not get some values from structure"); g_free (caps->pos); caps->pos = NULL; return FALSE; } if (caps->is_int && caps->depth > caps->width) { GST_DEBUG ("width > depth, not allowed - make us advertise correct caps"); g_free (caps->pos); caps->pos = NULL; return FALSE; } return TRUE; } static gboolean gst_audio_convert_link_src (GstAudioConvert * this, GstCaps * sinkcaps, GstAudioConvertCaps * sink_ac_caps) { GstAudioConvertCaps ac_caps = { 0 }; if (gst_pad_peer_accept_caps (this->src, sinkcaps)) { /* great, so that will be our suggestion then */ this->src_prefered = gst_caps_ref (sinkcaps); gst_caps_replace (&GST_PAD_CAPS (this->src), sinkcaps); ac_caps = *sink_ac_caps; if (ac_caps.pos) { ac_caps.pos = g_memdup (ac_caps.pos, sizeof (gint) * ac_caps.channels); } } else { /* nope, find something we can convert to and the peer can * accept. */ GstCaps *othercaps = gst_pad_peer_get_caps (this->src); if (othercaps) { /* peel off first one */ GstCaps *targetcaps = gst_caps_copy_nth (othercaps, 0); GstStructure *structure = gst_caps_get_structure (targetcaps, 0); gst_caps_unref (othercaps); /* set the rate on the caps, this has to work */ gst_structure_set (structure, "rate", G_TYPE_INT, sink_ac_caps->rate, "channels", G_TYPE_INT, sink_ac_caps->channels, NULL); if (strcmp (gst_structure_get_name (structure), "audio/x-raw-float") == 0) { if (!sink_ac_caps->is_int) { /* copy over */ gst_structure_set (structure, "buffer-frames", G_TYPE_INT, ac_caps.buffer_frames, NULL); } else { /* set to anything */ gst_structure_set (structure, "buffer-frames", G_TYPE_INT, 0, NULL); } } /* this will be our suggestion */ this->src_prefered = targetcaps; if (!gst_audio_convert_parse_caps (targetcaps, &ac_caps)) return FALSE; gst_caps_replace (&GST_PAD_CAPS (this->src), targetcaps); } } this->srccaps = ac_caps; GST_DEBUG_OBJECT (this, "negotiated pad to %" GST_PTR_FORMAT, sinkcaps); return TRUE; } static gboolean gst_audio_convert_setcaps (GstPad * pad, GstCaps * caps) { GstAudioConvert *this; GstAudioConvertCaps ac_caps = { 0 }; gboolean res; g_return_val_if_fail (GST_IS_PAD (pad), FALSE); g_return_val_if_fail (GST_IS_AUDIO_CONVERT (GST_OBJECT_PARENT (pad)), FALSE); g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE); this = GST_AUDIO_CONVERT (GST_OBJECT_PARENT (pad)); /* we'll need a new matrix after every new negotiation */ gst_audio_convert_unset_matrix (this); ac_caps.pos = NULL; if (!gst_audio_convert_parse_caps (caps, &ac_caps)) return FALSE; this->sink_prefered = caps; if ((res = gst_audio_convert_link_src (this, caps, &ac_caps))) { this->sinkcaps = ac_caps; GST_DEBUG_OBJECT (this, "negotiated pad to %" GST_PTR_FORMAT, caps); } return res; } /* tries to fixate the given field of the given caps to the given int value */ gboolean _fixate_caps_to_int (GstCaps * caps, const gchar * field, gint value) { gboolean changed = FALSE; guint i; /* FIXME: why don't we already return here when ret == TRUE ? */ for (i = 0; i < gst_caps_get_size (caps); i++) { GstStructure *structure = gst_caps_get_structure (caps, i); if (gst_structure_has_field (structure, field)) changed |= gst_caps_structure_fixate_field_nearest_int (structure, field, value); } return changed; } static void gst_audio_convert_fixate (GstPad * pad, GstCaps * caps) { const GValue *pos_val; GstPad *otherpad; GstAudioConvert *this; GstAudioConvertCaps try, ac_caps; this = GST_AUDIO_CONVERT (GST_PAD_PARENT (pad)); otherpad = (pad == this->sink ? this->src : this->sink); ac_caps = (pad == this->sink ? this->srccaps : this->sinkcaps); if (!GST_PAD_IS_IN_SETCAPS (otherpad)) { try.channels = ac_caps.channels; try.width = ac_caps.is_int ? ac_caps.width : 16; try.depth = ac_caps.is_int ? ac_caps.depth : 16; try.endianness = ac_caps.is_int ? ac_caps.endianness : G_BYTE_ORDER; } else { try.channels = 2; try.width = 16; try.depth = 16; try.endianness = G_BYTE_ORDER; } if (_fixate_caps_to_int (caps, "channels", try.channels)) { int n, c; gst_structure_get_int (gst_caps_get_structure (caps, 0), "channels", &c); if (c > 2) { /* make sure we have a channelpositions structure or array here */ GstStructure *str; for (n = 0; n < gst_caps_get_size (caps); n++) { str = gst_caps_get_structure (caps, n); if (!gst_structure_get_value (str, "channel-positions")) { /* first try otherpad's positions, else anything */ if (ac_caps.pos != NULL && c == ac_caps.channels) { gst_audio_set_channel_positions (str, ac_caps.pos); } else { gst_audio_set_structure_channel_positions_list (str, supported_positions, GST_AUDIO_CHANNEL_POSITION_NUM); /* FIXME: fixate (else we'll be less fixed than we used to) */ } } } } else { /* make sure we don't */ for (n = 0; n < gst_caps_get_size (caps); n++) { gst_structure_remove_field (gst_caps_get_structure (caps, n), "channel-positions"); } } } _fixate_caps_to_int (caps, "width", try.width); if (gst_structure_get_name (gst_caps_get_structure (caps, 0))[12] == 'i') { _fixate_caps_to_int (caps, "depth", try.depth); } _fixate_caps_to_int (caps, "endianness", try.endianness); if ((pos_val = gst_structure_get_value (gst_caps_get_structure (caps, 0), "channel-positions")) != NULL) { GstAudioChannelPosition *pos; const GValue *pos_val_entry; gint i; for (i = 0; i < gst_value_list_get_size (pos_val); i++) { pos_val_entry = gst_value_list_get_value (pos_val, i); if (G_VALUE_TYPE (pos_val_entry) == GST_TYPE_LIST) { /* unfixed */ pos = gst_audio_fixate_channel_positions (gst_caps_get_structure (caps, 0)); if (pos) { gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos); g_free (pos); pos = NULL; } } } } } static GstElementStateReturn gst_audio_convert_change_state (GstElement * element) { GstElementStateReturn ret; GstAudioConvert *this = GST_AUDIO_CONVERT (element); gint transition; transition = GST_STATE_TRANSITION (element); switch (transition) { default: break; } ret = parent_class->change_state (element); switch (transition) { case GST_STATE_PAUSED_TO_READY: this->convert_internal = NULL; gst_audio_convert_unset_matrix (this); gst_caps_replace (&GST_PAD_CAPS (this->sink), NULL); gst_caps_replace (&GST_PAD_CAPS (this->src), NULL); break; default: break; } return ret; } /* return a writable buffer of size which ideally is the same as before - You must unref the new buffer - The size of the old buffer is undefined after this operation */ static GstBuffer * gst_audio_convert_get_buffer (GstBuffer * buf, guint size) { GstBuffer *ret; g_assert (GST_IS_BUFFER (buf)); GST_LOG ("new buffer of size %u requested. Current is: data: %p - size: %u", size, buf->data, buf->size); if (buf->size >= size && gst_buffer_is_writable (buf)) { gst_buffer_ref (buf); buf->size = size; GST_LOG ("returning same buffer with adjusted values. data: %p - size: %u", buf->data, buf->size); return buf; } else { ret = gst_buffer_new_and_alloc (size); g_assert (ret); gst_buffer_stamp (ret, buf); GST_LOG ("returning new buffer. data: %p - size: %u", ret->data, ret->size); return ret; } } static inline guint8 GUINT8_IDENTITY (guint8 x) { return x; } static inline guint8 GINT8_IDENTITY (gint8 x) { return x; } #define CONVERT_TO(to, from, type, sign, endianness, LE_FUNC, BE_FUNC) \ G_STMT_START{ \ type value; \ memcpy (&value, from, sizeof (type)); \ from -= sizeof (type); \ value = (endianness == G_LITTLE_ENDIAN) ? LE_FUNC (value) : BE_FUNC (value); \ if (sign) { \ to = value; \ } else { \ to = (gint64) value - (1 << (sizeof (type) * 8 - 1)); \ } \ }G_STMT_END; static GstBuffer * gst_audio_convert_buffer_to_default_format (GstAudioConvert * this, GstBuffer * buf) { GstBuffer *ret; gint i, count; gint64 cur = 0; gint32 write; gint32 *dest; guint8 *src; if (this->sinkcaps.is_int) { if (this->sinkcaps.width == 32 && this->sinkcaps.depth == 32 && this->sinkcaps.endianness == G_BYTE_ORDER && this->sinkcaps.sign == TRUE) return buf; ret = gst_audio_convert_get_buffer (buf, buf->size * 32 / this->sinkcaps.width); gst_buffer_set_caps (ret, GST_PAD_CAPS (this->src)); count = ret->size / 4; src = buf->data + (count - 1) * (this->sinkcaps.width / 8); dest = (gint32 *) ret->data; for (i = count - 1; i >= 0; i--) { switch (this->sinkcaps.width) { case 8: if (this->sinkcaps.sign) { CONVERT_TO (cur, src, gint8, this->sinkcaps.sign, this->sinkcaps.endianness, GINT8_IDENTITY, GINT8_IDENTITY); } else { CONVERT_TO (cur, src, guint8, this->sinkcaps.sign, this->sinkcaps.endianness, GUINT8_IDENTITY, GUINT8_IDENTITY); } break; case 16: if (this->sinkcaps.sign) { CONVERT_TO (cur, src, gint16, this->sinkcaps.sign, this->sinkcaps.endianness, GINT16_FROM_LE, GINT16_FROM_BE); } else { CONVERT_TO (cur, src, guint16, this->sinkcaps.sign, this->sinkcaps.endianness, GUINT16_FROM_LE, GUINT16_FROM_BE); } break; case 24: { /* Read 24-bits LE/BE into signed 64 host-endian */ if (this->sinkcaps.endianness == G_LITTLE_ENDIAN) { cur = src[0] | (src[1] << 8) | (src[2] << 16); } else { cur = src[2] | (src[1] << 8) | (src[0] << 16); } /* Sign extend */ if ((this->sinkcaps.sign) && (cur & (1 << (this->sinkcaps.depth - 1)))) cur |= ((gint64) (-1)) ^ ((1 << this->sinkcaps.depth) - 1); src -= 3; } break; case 32: if (this->sinkcaps.sign) { CONVERT_TO (cur, src, gint32, this->sinkcaps.sign, this->sinkcaps.endianness, GINT32_FROM_LE, GINT32_FROM_BE); } else { CONVERT_TO (cur, src, guint32, this->sinkcaps.sign, this->sinkcaps.endianness, GUINT32_FROM_LE, GUINT32_FROM_BE); } break; default: g_assert_not_reached (); } cur = cur * ((gint64) 1 << (32 - this->sinkcaps.depth)); cur = CLAMP (cur, -((gint64) 1 << 32), (gint64) 0x7FFFFFFF); write = cur; memcpy (&dest[i], &write, 4); } } else { /* float2int */ gfloat *in; gint32 *out; float temp; /* should just give the same buffer, unless it's not writable -- float is * already 32 bits */ ret = gst_audio_convert_get_buffer (buf, buf->size); gst_buffer_set_caps (ret, GST_PAD_CAPS (this->src)); in = (gfloat *) GST_BUFFER_DATA (buf); out = (gint32 *) GST_BUFFER_DATA (ret); for (i = buf->size / sizeof (float); i > 0; i--) { temp = *in * 2147483647.0f + .5; *out = (gint32) CLAMP ((gint64) temp, -2147483648ll, 2147483647ll); out++; in++; } } gst_buffer_unref (buf); return ret; } #define POPULATE(out, format, be_func, le_func) G_STMT_START{ \ format val; \ format* p = (format *) out; \ int_value >>= (32 - this->srccaps.depth); \ if (this->srccaps.sign) { \ val = (format) int_value; \ } else { \ val = (format) int_value + (1 << (this->srccaps.depth - 1)); \ } \ switch (this->srccaps.endianness) { \ case G_LITTLE_ENDIAN: \ val = le_func (val); \ break; \ case G_BIG_ENDIAN: \ val = be_func (val); \ break; \ default: \ g_assert_not_reached (); \ }; \ *p = val; \ p ++; \ out = (guint8 *) p; \ }G_STMT_END static GstBuffer * gst_audio_convert_buffer_from_default_format (GstAudioConvert * this, GstBuffer * buf) { GstBuffer *ret; guint count, i; gint32 *src; if (this->srccaps.is_int && this->srccaps.width == 32 && this->srccaps.depth == 32 && this->srccaps.endianness == G_BYTE_ORDER && this->srccaps.sign == TRUE) return buf; if (this->srccaps.is_int) { guint8 *dest; count = buf->size / 4; /* size is undefined after gst_audio_convert_get_buffer! */ ret = gst_audio_convert_get_buffer (buf, buf->size * this->srccaps.width / 32); gst_buffer_set_caps (ret, GST_PAD_CAPS (this->src)); dest = ret->data; src = (gint32 *) buf->data; for (i = 0; i < count; i++) { gint32 int_value = *src; src++; switch (this->srccaps.width) { case 8: if (this->srccaps.sign) { POPULATE (dest, gint8, GINT8_IDENTITY, GINT8_IDENTITY); } else { POPULATE (dest, guint8, GUINT8_IDENTITY, GUINT8_IDENTITY); } break; case 16: if (this->srccaps.sign) { POPULATE (dest, gint16, GINT16_TO_BE, GINT16_TO_LE); } else { POPULATE (dest, guint16, GUINT16_TO_BE, GUINT16_TO_LE); } break; case 24: { guint8 tmp[4]; guint8 *tmpp = tmp; /* Write out big endian array */ if (this->srccaps.sign) { POPULATE (tmpp, gint32, GINT32_TO_BE, GINT32_TO_BE); } else { POPULATE (tmpp, guint32, GUINT32_TO_BE, GUINT32_TO_BE); } if (this->srccaps.endianness == G_LITTLE_ENDIAN) { dest[2] = tmp[1]; dest[1] = tmp[2]; dest[0] = tmp[3]; } else { memcpy (dest, tmp + 1, 3); } dest += 3; } break; case 32: if (this->srccaps.sign) { POPULATE (dest, gint32, GINT32_TO_BE, GINT32_TO_LE); } else { POPULATE (dest, guint32, GUINT32_TO_BE, GUINT32_TO_LE); } break; default: g_assert_not_reached (); } } } else { gfloat *dest; /* 1 / (2^31-1) * i */ #define INT2FLOAT(i) (4.6566128752457969e-10 * ((gfloat)i)) count = buf->size / 4; /* size is undefined after gst_audio_convert_get_buffer! */ ret = gst_audio_convert_get_buffer (buf, buf->size * this->srccaps.width / 32); gst_buffer_set_caps (ret, GST_PAD_CAPS (this->src)); dest = (gfloat *) ret->data; src = (gint32 *) buf->data; for (i = 0; i < count; i++) { *dest = (4.6566128752457969e-10 * ((gfloat) * src)); dest++; src++; } } gst_buffer_unref (buf); return ret; } static GstBuffer * gst_audio_convert_channels (GstAudioConvert * this, GstBuffer * buf) { GstBuffer *ret; gint count; g_assert (this->matrix != NULL); /* check for passthrough */ if (gst_audio_convert_passthrough (this)) return buf; /* convert */ count = GST_BUFFER_SIZE (buf) / 4 / this->sinkcaps.channels; ret = gst_audio_convert_get_buffer (buf, count * 4 * this->srccaps.channels); gst_buffer_set_caps (ret, GST_PAD_CAPS (this->src)); gst_audio_convert_mix (this, (gint32 *) GST_BUFFER_DATA (buf), (gint32 *) GST_BUFFER_DATA (ret), count); gst_buffer_unref (buf); return ret; }