/* GStreamer * Copyright (C) 2004 Ronald Bultje * Copyright (C) 2008 Sebastian Dröge * * audio-channel-mixer.c: setup of channel conversion matrices * * 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. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include "audio-channel-mixer.h" #ifndef GST_DISABLE_GST_DEBUG #define GST_CAT_DEFAULT ensure_debug_category() static GstDebugCategory * ensure_debug_category (void) { static gsize cat_gonce = 0; if (g_once_init_enter (&cat_gonce)) { gsize cat_done; cat_done = (gsize) _gst_debug_category_new ("audio-channel-mixer", 0, "audio-channel-mixer object"); g_once_init_leave (&cat_gonce, cat_done); } return (GstDebugCategory *) cat_gonce; } #else #define ensure_debug_category() /* NOOP */ #endif /* GST_DISABLE_GST_DEBUG */ #define PRECISION_INT 10 typedef void (*MixerFunc) (GstAudioChannelMixer * mix, const gpointer src[], gpointer dst[], gint samples); struct _GstAudioChannelMixer { gint in_channels; gint out_channels; /* channel conversion matrix, m[in_channels][out_channels]. * If identity matrix, passthrough applies. */ gfloat **matrix; /* channel conversion matrix with int values, m[in_channels][out_channels]. * this is matrix * (2^10) as integers */ gint **matrix_int; MixerFunc func; }; /** * gst_audio_channel_mixer_free: * @mix: a #GstAudioChannelMixer * * Free memory allocated by @mix. */ void gst_audio_channel_mixer_free (GstAudioChannelMixer * mix) { gint i; /* free */ for (i = 0; i < mix->in_channels; i++) g_free (mix->matrix[i]); g_free (mix->matrix); mix->matrix = NULL; for (i = 0; i < mix->in_channels; i++) g_free (mix->matrix_int[i]); g_free (mix->matrix_int); mix->matrix_int = NULL; g_free (mix); } /* * Detect and fill in identical channels. E.g. * forward the left/right front channels in a * 5.1 to 2.0 conversion. */ static void gst_audio_channel_mixer_fill_identical (gfloat ** matrix, gint in_channels, GstAudioChannelPosition * in_position, gint out_channels, GstAudioChannelPosition * out_position, GstAudioChannelMixerFlags flags) { gint ci, co; /* Apart from the compatible channel assignments, we can also have * same channel assignments. This is much simpler, we simply copy * the value from source to dest! */ for (co = 0; co < out_channels; co++) { /* find a channel in input with same position */ for (ci = 0; ci < in_channels; ci++) { /* If the input was unpositioned, we're simply building * an identity matrix */ if (flags & GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_IN) { matrix[ci][co] = ci == co ? 1.0 : 0.0; } else if (in_position[ci] == out_position[co]) { matrix[ci][co] = 1.0; } } } } /* * Detect and fill in compatible channels. E.g. * forward left/right front to mono (or the other * way around) when going from 2.0 to 1.0. */ static void gst_audio_channel_mixer_fill_compatible (gfloat ** matrix, gint in_channels, GstAudioChannelPosition * in_position, gint out_channels, GstAudioChannelPosition * out_position) { /* Conversions from one-channel to compatible two-channel configs */ struct { GstAudioChannelPosition pos1[2]; GstAudioChannelPosition pos2[1]; } conv[] = { /* front: mono <-> stereo */ {{ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, { GST_AUDIO_CHANNEL_POSITION_MONO}}, /* front center: 2 <-> 1 */ {{ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER}, { GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}}, /* rear: 2 <-> 1 */ {{ GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, { GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}}, {{ GST_AUDIO_CHANNEL_POSITION_INVALID}} }; gint c; /* conversions from compatible (but not the same) channel schemes */ for (c = 0; conv[c].pos1[0] != GST_AUDIO_CHANNEL_POSITION_INVALID; c++) { gint pos1_0 = -1, pos1_1 = -1, pos1_2 = -1; gint pos2_0 = -1, pos2_1 = -1, pos2_2 = -1; gint n; for (n = 0; n < in_channels; n++) { if (in_position[n] == conv[c].pos1[0]) pos1_0 = n; else if (in_position[n] == conv[c].pos1[1]) pos1_1 = n; else if (in_position[n] == conv[c].pos2[0]) pos1_2 = n; } for (n = 0; n < out_channels; n++) { if (out_position[n] == conv[c].pos1[0]) pos2_0 = n; else if (out_position[n] == conv[c].pos1[1]) pos2_1 = n; else if (out_position[n] == conv[c].pos2[0]) pos2_2 = n; } /* The general idea here is to fill in channels from the same position * as good as possible. This means mixing left<->center and right<->center. */ /* left -> center */ if (pos1_0 != -1 && pos1_2 == -1 && pos2_0 == -1 && pos2_2 != -1) matrix[pos1_0][pos2_2] = 1.0; else if (pos1_0 != -1 && pos1_2 != -1 && pos2_0 == -1 && pos2_2 != -1) matrix[pos1_0][pos2_2] = 0.5; else if (pos1_0 != -1 && pos1_2 == -1 && pos2_0 != -1 && pos2_2 != -1) matrix[pos1_0][pos2_2] = 1.0; /* right -> center */ if (pos1_1 != -1 && pos1_2 == -1 && pos2_1 == -1 && pos2_2 != -1) matrix[pos1_1][pos2_2] = 1.0; else if (pos1_1 != -1 && pos1_2 != -1 && pos2_1 == -1 && pos2_2 != -1) matrix[pos1_1][pos2_2] = 0.5; else if (pos1_1 != -1 && pos1_2 == -1 && pos2_1 != -1 && pos2_2 != -1) matrix[pos1_1][pos2_2] = 1.0; /* center -> left */ if (pos1_2 != -1 && pos1_0 == -1 && pos2_2 == -1 && pos2_0 != -1) matrix[pos1_2][pos2_0] = 1.0; else if (pos1_2 != -1 && pos1_0 != -1 && pos2_2 == -1 && pos2_0 != -1) matrix[pos1_2][pos2_0] = 0.5; else if (pos1_2 != -1 && pos1_0 == -1 && pos2_2 != -1 && pos2_0 != -1) matrix[pos1_2][pos2_0] = 1.0; /* center -> right */ if (pos1_2 != -1 && pos1_1 == -1 && pos2_2 == -1 && pos2_1 != -1) matrix[pos1_2][pos2_1] = 1.0; else if (pos1_2 != -1 && pos1_1 != -1 && pos2_2 == -1 && pos2_1 != -1) matrix[pos1_2][pos2_1] = 0.5; else if (pos1_2 != -1 && pos1_1 == -1 && pos2_2 != -1 && pos2_1 != -1) matrix[pos1_2][pos2_1] = 1.0; } } /* * Detect and fill in channels not handled by the * above two, e.g. center to left/right front in * 5.1 to 2.0 (or the other way around). * * Unfortunately, limited to static conversions * for now. */ static void gst_audio_channel_mixer_detect_pos (gint channels, GstAudioChannelPosition position[64], gint * f, gboolean * has_f, gint * c, gboolean * has_c, gint * r, gboolean * has_r, gint * s, gboolean * has_s, gint * b, gboolean * has_b) { gint n; for (n = 0; n < channels; n++) { switch (position[n]) { case GST_AUDIO_CHANNEL_POSITION_MONO: f[1] = n; *has_f = TRUE; break; case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT: f[0] = n; *has_f = TRUE; break; case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT: f[2] = n; *has_f = TRUE; break; case GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER: c[1] = n; *has_c = TRUE; break; case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER: c[0] = n; *has_c = TRUE; break; case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER: c[2] = n; *has_c = TRUE; break; case GST_AUDIO_CHANNEL_POSITION_REAR_CENTER: r[1] = n; *has_r = TRUE; break; case GST_AUDIO_CHANNEL_POSITION_REAR_LEFT: r[0] = n; *has_r = TRUE; break; case GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT: r[2] = n; *has_r = TRUE; break; case GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT: s[0] = n; *has_s = TRUE; break; case GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT: s[2] = n; *has_s = TRUE; break; case GST_AUDIO_CHANNEL_POSITION_LFE1: *has_b = TRUE; b[1] = n; break; default: break; } } } static void gst_audio_channel_mixer_fill_one_other (gfloat ** matrix, gint * from_idx, gint * to_idx, gfloat ratio) { /* src & dst have center => passthrough */ if (from_idx[1] != -1 && to_idx[1] != -1) { matrix[from_idx[1]][to_idx[1]] = ratio; } /* src & dst have left => passthrough */ if (from_idx[0] != -1 && to_idx[0] != -1) { matrix[from_idx[0]][to_idx[0]] = ratio; } /* src & dst have right => passthrough */ if (from_idx[2] != -1 && to_idx[2] != -1) { matrix[from_idx[2]][to_idx[2]] = ratio; } /* src has left & dst has center => put into center */ if (from_idx[0] != -1 && to_idx[1] != -1 && from_idx[1] != -1) { matrix[from_idx[0]][to_idx[1]] = 0.5 * ratio; } else if (from_idx[0] != -1 && to_idx[1] != -1 && from_idx[1] == -1) { matrix[from_idx[0]][to_idx[1]] = ratio; } /* src has right & dst has center => put into center */ if (from_idx[2] != -1 && to_idx[1] != -1 && from_idx[1] != -1) { matrix[from_idx[2]][to_idx[1]] = 0.5 * ratio; } else if (from_idx[2] != -1 && to_idx[1] != -1 && from_idx[1] == -1) { matrix[from_idx[2]][to_idx[1]] = ratio; } /* src has center & dst has left => passthrough */ if (from_idx[1] != -1 && to_idx[0] != -1 && from_idx[0] != -1) { matrix[from_idx[1]][to_idx[0]] = 0.5 * ratio; } else if (from_idx[1] != -1 && to_idx[0] != -1 && from_idx[0] == -1) { matrix[from_idx[1]][to_idx[0]] = ratio; } /* src has center & dst has right => passthrough */ if (from_idx[1] != -1 && to_idx[2] != -1 && from_idx[2] != -1) { matrix[from_idx[1]][to_idx[2]] = 0.5 * ratio; } else if (from_idx[1] != -1 && to_idx[2] != -1 && from_idx[2] == -1) { matrix[from_idx[1]][to_idx[2]] = ratio; } } #define RATIO_CENTER_FRONT (1.0 / sqrt (2.0)) #define RATIO_CENTER_SIDE (1.0 / 2.0) #define RATIO_CENTER_REAR (1.0 / sqrt (8.0)) #define RATIO_FRONT_CENTER (1.0 / sqrt (2.0)) #define RATIO_FRONT_SIDE (1.0 / sqrt (2.0)) #define RATIO_FRONT_REAR (1.0 / 2.0) #define RATIO_SIDE_CENTER (1.0 / 2.0) #define RATIO_SIDE_FRONT (1.0 / sqrt (2.0)) #define RATIO_SIDE_REAR (1.0 / sqrt (2.0)) #define RATIO_CENTER_BASS (1.0 / sqrt (2.0)) #define RATIO_FRONT_BASS (1.0) #define RATIO_SIDE_BASS (1.0 / sqrt (2.0)) #define RATIO_REAR_BASS (1.0 / sqrt (2.0)) static void gst_audio_channel_mixer_fill_others (gfloat ** matrix, gint in_channels, GstAudioChannelPosition * in_position, gint out_channels, GstAudioChannelPosition * out_position) { gboolean in_has_front = FALSE, out_has_front = FALSE, in_has_center = FALSE, out_has_center = FALSE, in_has_rear = FALSE, out_has_rear = FALSE, in_has_side = FALSE, out_has_side = FALSE, in_has_bass = FALSE, out_has_bass = FALSE; /* LEFT, RIGHT, MONO */ gint in_f[3] = { -1, -1, -1 }; gint out_f[3] = { -1, -1, -1 }; /* LOC, ROC, CENTER */ gint in_c[3] = { -1, -1, -1 }; gint out_c[3] = { -1, -1, -1 }; /* RLEFT, RRIGHT, RCENTER */ gint in_r[3] = { -1, -1, -1 }; gint out_r[3] = { -1, -1, -1 }; /* SLEFT, INVALID, SRIGHT */ gint in_s[3] = { -1, -1, -1 }; gint out_s[3] = { -1, -1, -1 }; /* INVALID, LFE, INVALID */ gint in_b[3] = { -1, -1, -1 }; gint out_b[3] = { -1, -1, -1 }; /* First see where (if at all) the various channels from/to * which we want to convert are located in our matrix/array. */ gst_audio_channel_mixer_detect_pos (in_channels, in_position, in_f, &in_has_front, in_c, &in_has_center, in_r, &in_has_rear, in_s, &in_has_side, in_b, &in_has_bass); gst_audio_channel_mixer_detect_pos (out_channels, out_position, out_f, &out_has_front, out_c, &out_has_center, out_r, &out_has_rear, out_s, &out_has_side, out_b, &out_has_bass); /* The general idea here is: * - if the source has a channel that the destination doesn't have mix * it into the nearest available destination channel * - if the destination has a channel that the source doesn't have mix * the nearest source channel into the destination channel * * The ratio for the mixing becomes lower as the distance between the * channels gets larger */ /* center <-> front/side/rear */ if (!in_has_center && in_has_front && out_has_center) { gst_audio_channel_mixer_fill_one_other (matrix, in_f, out_c, RATIO_CENTER_FRONT); } else if (!in_has_center && !in_has_front && in_has_side && out_has_center) { gst_audio_channel_mixer_fill_one_other (matrix, in_s, out_c, RATIO_CENTER_SIDE); } else if (!in_has_center && !in_has_front && !in_has_side && in_has_rear && out_has_center) { gst_audio_channel_mixer_fill_one_other (matrix, in_r, out_c, RATIO_CENTER_REAR); } else if (in_has_center && !out_has_center && out_has_front) { gst_audio_channel_mixer_fill_one_other (matrix, in_c, out_f, RATIO_CENTER_FRONT); } else if (in_has_center && !out_has_center && !out_has_front && out_has_side) { gst_audio_channel_mixer_fill_one_other (matrix, in_c, out_s, RATIO_CENTER_SIDE); } else if (in_has_center && !out_has_center && !out_has_front && !out_has_side && out_has_rear) { gst_audio_channel_mixer_fill_one_other (matrix, in_c, out_r, RATIO_CENTER_REAR); } /* front <-> center/side/rear */ if (!in_has_front && in_has_center && !in_has_side && out_has_front) { gst_audio_channel_mixer_fill_one_other (matrix, in_c, out_f, RATIO_CENTER_FRONT); } else if (!in_has_front && !in_has_center && in_has_side && out_has_front) { gst_audio_channel_mixer_fill_one_other (matrix, in_s, out_f, RATIO_FRONT_SIDE); } else if (!in_has_front && in_has_center && in_has_side && out_has_front) { gst_audio_channel_mixer_fill_one_other (matrix, in_c, out_f, 0.5 * RATIO_CENTER_FRONT); gst_audio_channel_mixer_fill_one_other (matrix, in_s, out_f, 0.5 * RATIO_FRONT_SIDE); } else if (!in_has_front && !in_has_center && !in_has_side && in_has_rear && out_has_front) { gst_audio_channel_mixer_fill_one_other (matrix, in_r, out_f, RATIO_FRONT_REAR); } else if (in_has_front && out_has_center && !out_has_side && !out_has_front) { gst_audio_channel_mixer_fill_one_other (matrix, in_f, out_c, RATIO_CENTER_FRONT); } else if (in_has_front && !out_has_center && out_has_side && !out_has_front) { gst_audio_channel_mixer_fill_one_other (matrix, in_f, out_s, RATIO_FRONT_SIDE); } else if (in_has_front && out_has_center && out_has_side && !out_has_front) { gst_audio_channel_mixer_fill_one_other (matrix, in_f, out_c, 0.5 * RATIO_CENTER_FRONT); gst_audio_channel_mixer_fill_one_other (matrix, in_f, out_s, 0.5 * RATIO_FRONT_SIDE); } else if (in_has_front && !out_has_center && !out_has_side && !out_has_front && out_has_rear) { gst_audio_channel_mixer_fill_one_other (matrix, in_f, out_r, RATIO_FRONT_REAR); } /* side <-> center/front/rear */ if (!in_has_side && in_has_front && !in_has_rear && out_has_side) { gst_audio_channel_mixer_fill_one_other (matrix, in_f, out_s, RATIO_FRONT_SIDE); } else if (!in_has_side && !in_has_front && in_has_rear && out_has_side) { gst_audio_channel_mixer_fill_one_other (matrix, in_r, out_s, RATIO_SIDE_REAR); } else if (!in_has_side && in_has_front && in_has_rear && out_has_side) { gst_audio_channel_mixer_fill_one_other (matrix, in_f, out_s, 0.5 * RATIO_FRONT_SIDE); gst_audio_channel_mixer_fill_one_other (matrix, in_r, out_s, 0.5 * RATIO_SIDE_REAR); } else if (!in_has_side && !in_has_front && !in_has_rear && in_has_center && out_has_side) { gst_audio_channel_mixer_fill_one_other (matrix, in_c, out_s, RATIO_CENTER_SIDE); } else if (in_has_side && out_has_front && !out_has_rear && !out_has_side) { gst_audio_channel_mixer_fill_one_other (matrix, in_s, out_f, RATIO_FRONT_SIDE); } else if (in_has_side && !out_has_front && out_has_rear && !out_has_side) { gst_audio_channel_mixer_fill_one_other (matrix, in_s, out_r, RATIO_SIDE_REAR); } else if (in_has_side && out_has_front && out_has_rear && !out_has_side) { gst_audio_channel_mixer_fill_one_other (matrix, in_s, out_f, 0.5 * RATIO_FRONT_SIDE); gst_audio_channel_mixer_fill_one_other (matrix, in_s, out_r, 0.5 * RATIO_SIDE_REAR); } else if (in_has_side && !out_has_front && !out_has_rear && out_has_center && !out_has_side) { gst_audio_channel_mixer_fill_one_other (matrix, in_s, out_c, RATIO_CENTER_SIDE); } /* rear <-> center/front/side */ if (!in_has_rear && in_has_side && out_has_rear) { gst_audio_channel_mixer_fill_one_other (matrix, in_s, out_r, RATIO_SIDE_REAR); } else if (!in_has_rear && !in_has_side && in_has_front && out_has_rear) { gst_audio_channel_mixer_fill_one_other (matrix, in_f, out_r, RATIO_FRONT_REAR); } else if (!in_has_rear && !in_has_side && !in_has_front && in_has_center && out_has_rear) { gst_audio_channel_mixer_fill_one_other (matrix, in_c, out_r, RATIO_CENTER_REAR); } else if (in_has_rear && !out_has_rear && out_has_side) { gst_audio_channel_mixer_fill_one_other (matrix, in_r, out_s, RATIO_SIDE_REAR); } else if (in_has_rear && !out_has_rear && !out_has_side && out_has_front) { gst_audio_channel_mixer_fill_one_other (matrix, in_r, out_f, RATIO_FRONT_REAR); } else if (in_has_rear && !out_has_rear && !out_has_side && !out_has_front && out_has_center) { gst_audio_channel_mixer_fill_one_other (matrix, in_r, out_c, RATIO_CENTER_REAR); } /* bass <-> any */ if (in_has_bass && !out_has_bass) { if (out_has_center) { gst_audio_channel_mixer_fill_one_other (matrix, in_b, out_c, RATIO_CENTER_BASS); } if (out_has_front) { gst_audio_channel_mixer_fill_one_other (matrix, in_b, out_f, RATIO_FRONT_BASS); } if (out_has_side) { gst_audio_channel_mixer_fill_one_other (matrix, in_b, out_s, RATIO_SIDE_BASS); } if (out_has_rear) { gst_audio_channel_mixer_fill_one_other (matrix, in_b, out_r, RATIO_REAR_BASS); } } else if (!in_has_bass && out_has_bass) { if (in_has_center) { gst_audio_channel_mixer_fill_one_other (matrix, in_c, out_b, RATIO_CENTER_BASS); } if (in_has_front) { gst_audio_channel_mixer_fill_one_other (matrix, in_f, out_b, RATIO_FRONT_BASS); } if (in_has_side) { gst_audio_channel_mixer_fill_one_other (matrix, in_s, out_b, RATIO_REAR_BASS); } if (in_has_rear) { gst_audio_channel_mixer_fill_one_other (matrix, in_r, out_b, RATIO_REAR_BASS); } } } /* * Normalize output values. */ static void gst_audio_channel_mixer_fill_normalize (gfloat ** matrix, gint in_channels, gint out_channels) { gfloat sum, top = 0; gint i, j; for (j = 0; j < out_channels; j++) { /* calculate sum */ sum = 0.0; for (i = 0; i < in_channels; i++) { sum += fabs (matrix[i][j]); } if (sum > top) { top = sum; } } /* normalize to mix */ if (top == 0.0) return; for (j = 0; j < out_channels; j++) { for (i = 0; i < in_channels; i++) { matrix[i][j] /= top; } } } static gboolean gst_audio_channel_mixer_fill_special (gfloat ** matrix, gint in_channels, GstAudioChannelPosition * in_position, gint out_channels, GstAudioChannelPosition * out_position) { /* Special, standard conversions here */ /* Mono<->Stereo, just a fast-path */ if (in_channels == 2 && out_channels == 1 && ((in_position[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT && in_position[1] == GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT) || (in_position[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT && in_position[1] == GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT)) && out_position[0] == GST_AUDIO_CHANNEL_POSITION_MONO) { matrix[0][0] = 0.5; matrix[1][0] = 0.5; return TRUE; } else if (in_channels == 1 && out_channels == 2 && ((out_position[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT && out_position[1] == GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT) || (out_position[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT && out_position[1] == GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT)) && in_position[0] == GST_AUDIO_CHANNEL_POSITION_MONO) { matrix[0][0] = 1.0; matrix[0][1] = 1.0; return TRUE; } /* TODO: 5.1 <-> Stereo and other standard conversions */ return FALSE; } /* * Automagically generate conversion matrix. */ typedef enum { GST_AUDIO_CHANNEL_MIXER_VIRTUAL_INPUT_NONE = 0, GST_AUDIO_CHANNEL_MIXER_VIRTUAL_INPUT_MONO, GST_AUDIO_CHANNEL_MIXER_VIRTUAL_INPUT_STEREO } GstAudioChannelMixerVirtualInput; /* Detects specific input channels configurations introduced in the * audioconvert element (since version 1.26) with the * `GstAudioConvertInputChannelsReorder` configurations. * * If all input channels are positioned to GST_AUDIO_CHANNEL_POSITION_MONO, * the automatic mixing matrix should be configured like if there was only one * virtual input mono channel. This virtual mono channel is the mix of all the * real mono channels. * * If all input channels with an even index are positioned to * GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT and all input channels with an odd * index are positioned to GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, then the * automatic mixing matrix should be configured like if there were only one * virtual input left channel and one virtual input right channel. This virtual * left or right channel is the mix of all the real left or right channels. */ static gboolean gst_audio_channel_mixer_detect_virtual_input_channels (gint channels, GstAudioChannelPosition * position, GstAudioChannelMixerVirtualInput * virtual_input) { g_return_val_if_fail (position != NULL, FALSE); g_return_val_if_fail (virtual_input != NULL, FALSE); *virtual_input = GST_AUDIO_CHANNEL_MIXER_VIRTUAL_INPUT_NONE; if (channels < 2) return FALSE; static const GstAudioChannelPosition alternate_positions[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT }; gboolean is_mono = TRUE; gboolean is_alternate = TRUE; for (gint i = 0; i < channels; ++i) { if (position[i] != GST_AUDIO_CHANNEL_POSITION_MONO) is_mono = FALSE; if (position[i] != alternate_positions[i % 2]) is_alternate = FALSE; if (!is_mono && !is_alternate) return FALSE; } if (is_mono) { g_assert (!is_alternate); *virtual_input = GST_AUDIO_CHANNEL_MIXER_VIRTUAL_INPUT_MONO; return TRUE; } if (is_alternate && (channels > 2)) { g_assert (!is_mono); *virtual_input = GST_AUDIO_CHANNEL_MIXER_VIRTUAL_INPUT_STEREO; return TRUE; } return FALSE; } static void gst_audio_channel_mixer_fill_matrix (gfloat ** matrix, GstAudioChannelMixerFlags flags, gint in_channels, GstAudioChannelPosition * in_position, gint out_channels, GstAudioChannelPosition * out_position) { if (gst_audio_channel_mixer_fill_special (matrix, in_channels, in_position, out_channels, out_position)) return; /* If all input channels are positioned to mono, the mix matrix should be * configured like if there was only one virtual input mono channel. This * virtual mono channel is the mix of all the real input mono channels. * * If all input channels are positioned to left and right alternately, the mix * matrix should be configured like if there were only two virtual input * channels: one left and one right. This virtual left or right channel is the * mix of all the real input left or right channels. */ gint in_size = in_channels; GstAudioChannelMixerVirtualInput virtual_input = GST_AUDIO_CHANNEL_MIXER_VIRTUAL_INPUT_NONE; if (gst_audio_channel_mixer_detect_virtual_input_channels (in_size, in_position, &virtual_input)) { switch (virtual_input) { case GST_AUDIO_CHANNEL_MIXER_VIRTUAL_INPUT_MONO: in_size = 1; break; case GST_AUDIO_CHANNEL_MIXER_VIRTUAL_INPUT_STEREO: in_size = 2; break; default: break; } } gst_audio_channel_mixer_fill_identical (matrix, in_size, in_position, out_channels, out_position, flags); if (!(flags & GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_IN)) { gst_audio_channel_mixer_fill_compatible (matrix, in_size, in_position, out_channels, out_position); gst_audio_channel_mixer_fill_others (matrix, in_size, in_position, out_channels, out_position); gst_audio_channel_mixer_fill_normalize (matrix, in_size, out_channels); } switch (virtual_input) { case GST_AUDIO_CHANNEL_MIXER_VIRTUAL_INPUT_MONO:{ for (gint out = 0; out < out_channels; ++out) matrix[0][out] /= in_channels; for (gint in = 1; in < in_channels; ++in) memcpy (matrix[in], matrix[0], out_channels * sizeof (gfloat)); break; } case GST_AUDIO_CHANNEL_MIXER_VIRTUAL_INPUT_STEREO:{ gint right_channels = in_channels >> 1; gint left_channels = right_channels + (in_channels % 2); for (gint out = 0; out < out_channels; ++out) { matrix[0][out] /= left_channels; matrix[1][out] /= right_channels; } for (gint in = 2; in < in_channels; ++in) memcpy (matrix[in], matrix[in % 2], out_channels * sizeof (gfloat)); break; } default: break; } } /* only call mix after mix->matrix is fully set up and normalized */ static void gst_audio_channel_mixer_setup_matrix_int (GstAudioChannelMixer * mix) { gint i, j; gfloat tmp; gfloat factor = (1 << PRECISION_INT); mix->matrix_int = g_new0 (gint *, mix->in_channels); for (i = 0; i < mix->in_channels; i++) { mix->matrix_int[i] = g_new (gint, mix->out_channels); for (j = 0; j < mix->out_channels; j++) { tmp = mix->matrix[i][j] * factor; mix->matrix_int[i][j] = (gint) tmp; } } } static gfloat ** gst_audio_channel_mixer_setup_matrix (GstAudioChannelMixerFlags flags, gint in_channels, GstAudioChannelPosition * in_position, gint out_channels, GstAudioChannelPosition * out_position) { gint i, j; gfloat **matrix = g_new0 (gfloat *, in_channels); for (i = 0; i < in_channels; i++) { matrix[i] = g_new (gfloat, out_channels); for (j = 0; j < out_channels; j++) matrix[i][j] = 0.; } /* setup the matrix' internal values */ gst_audio_channel_mixer_fill_matrix (matrix, flags, in_channels, in_position, out_channels, out_position); return matrix; } #define DEFINE_GET_DATA_FUNCS(type) \ static inline type \ _get_in_data_interleaved_##type (const type * in_data[], \ gint sample, gint channel, gint total_channels) \ { \ return in_data[0][sample * total_channels + channel]; \ } \ \ static inline type * \ _get_out_data_interleaved_##type (type * out_data[], \ gint sample, gint channel, gint total_channels) \ { \ return &out_data[0][sample * total_channels + channel]; \ } \ \ static inline type \ _get_in_data_planar_##type (const type * in_data[], \ gint sample, gint channel, gint total_channels) \ { \ (void) total_channels; \ return in_data[channel][sample]; \ } \ \ static inline type * \ _get_out_data_planar_##type (type * out_data[], \ gint sample, gint channel, gint total_channels) \ { \ (void) total_channels; \ return &out_data[channel][sample]; \ } #define DEFINE_INTEGER_MIX_FUNC(bits, resbits, inlayout, outlayout) \ static void \ gst_audio_channel_mixer_mix_int##bits##_##inlayout##_##outlayout ( \ GstAudioChannelMixer * mix, const gint##bits * in_data[], \ gint##bits * out_data[], gint samples) \ { \ gint in, out, n; \ gint##resbits res; \ gint inchannels, outchannels; \ \ inchannels = mix->in_channels; \ outchannels = mix->out_channels; \ \ for (n = 0; n < samples; n++) { \ for (out = 0; out < outchannels; out++) { \ /* convert */ \ res = 0; \ for (in = 0; in < inchannels; in++) \ res += \ _get_in_data_##inlayout##_gint##bits (in_data, n, in, inchannels) * \ (gint##resbits) mix->matrix_int[in][out]; \ \ /* remove factor from int matrix */ \ res = (res + (1 << (PRECISION_INT - 1))) >> PRECISION_INT; \ *_get_out_data_##outlayout##_gint##bits (out_data, n, out, outchannels) = \ CLAMP (res, G_MININT##bits, G_MAXINT##bits); \ } \ } \ } #define DEFINE_FLOAT_MIX_FUNC(type, inlayout, outlayout) \ static void \ gst_audio_channel_mixer_mix_##type##_##inlayout##_##outlayout ( \ GstAudioChannelMixer * mix, const g##type * in_data[], \ g##type * out_data[], gint samples) \ { \ gint in, out, n; \ g##type res; \ gint inchannels, outchannels; \ \ inchannels = mix->in_channels; \ outchannels = mix->out_channels; \ \ for (n = 0; n < samples; n++) { \ for (out = 0; out < outchannels; out++) { \ /* convert */ \ res = 0.0; \ for (in = 0; in < inchannels; in++) \ res += \ _get_in_data_##inlayout##_g##type (in_data, n, in, inchannels) * \ mix->matrix[in][out]; \ \ *_get_out_data_##outlayout##_g##type (out_data, n, out, outchannels) = res; \ } \ } \ } DEFINE_GET_DATA_FUNCS (gint16); DEFINE_INTEGER_MIX_FUNC (16, 32, interleaved, interleaved); DEFINE_INTEGER_MIX_FUNC (16, 32, interleaved, planar); DEFINE_INTEGER_MIX_FUNC (16, 32, planar, interleaved); DEFINE_INTEGER_MIX_FUNC (16, 32, planar, planar); DEFINE_GET_DATA_FUNCS (gint32); DEFINE_INTEGER_MIX_FUNC (32, 64, interleaved, interleaved); DEFINE_INTEGER_MIX_FUNC (32, 64, interleaved, planar); DEFINE_INTEGER_MIX_FUNC (32, 64, planar, interleaved); DEFINE_INTEGER_MIX_FUNC (32, 64, planar, planar); DEFINE_GET_DATA_FUNCS (gfloat); DEFINE_FLOAT_MIX_FUNC (float, interleaved, interleaved); DEFINE_FLOAT_MIX_FUNC (float, interleaved, planar); DEFINE_FLOAT_MIX_FUNC (float, planar, interleaved); DEFINE_FLOAT_MIX_FUNC (float, planar, planar); DEFINE_GET_DATA_FUNCS (gdouble); DEFINE_FLOAT_MIX_FUNC (double, interleaved, interleaved); DEFINE_FLOAT_MIX_FUNC (double, interleaved, planar); DEFINE_FLOAT_MIX_FUNC (double, planar, interleaved); DEFINE_FLOAT_MIX_FUNC (double, planar, planar); /** * gst_audio_channel_mixer_new_with_matrix: (skip): * @flags: #GstAudioChannelMixerFlags * @in_channels: number of input channels * @out_channels: number of output channels * @matrix: (transfer full) (nullable): channel conversion matrix, m[@in_channels][@out_channels]. * If identity matrix, passthrough applies. If %NULL, a (potentially truncated) * identity matrix is generated. * * Create a new channel mixer object for the given parameters. * * Returns: a new #GstAudioChannelMixer object. * Free with gst_audio_channel_mixer_free() after usage. * * Since: 1.14 */ GstAudioChannelMixer * gst_audio_channel_mixer_new_with_matrix (GstAudioChannelMixerFlags flags, GstAudioFormat format, gint in_channels, gint out_channels, gfloat ** matrix) { GstAudioChannelMixer *mix; g_return_val_if_fail (format == GST_AUDIO_FORMAT_S16 || format == GST_AUDIO_FORMAT_S32 || format == GST_AUDIO_FORMAT_F32 || format == GST_AUDIO_FORMAT_F64, NULL); mix = g_new0 (GstAudioChannelMixer, 1); mix->in_channels = in_channels; mix->out_channels = out_channels; if (!matrix) { /* Generate (potentially truncated) identity matrix */ gint i, j; mix->matrix = g_new0 (gfloat *, in_channels); for (i = 0; i < in_channels; i++) { mix->matrix[i] = g_new (gfloat, out_channels); for (j = 0; j < out_channels; j++) { mix->matrix[i][j] = i == j ? 1.0 : 0.0; } } } else { mix->matrix = matrix; } gst_audio_channel_mixer_setup_matrix_int (mix); #ifndef GST_DISABLE_GST_DEBUG /* debug */ { GString *s; gint i, j; s = g_string_new ("Matrix for"); g_string_append_printf (s, " %d -> %d: ", mix->in_channels, mix->out_channels); g_string_append (s, "{"); for (i = 0; i < mix->in_channels; i++) { if (i != 0) g_string_append (s, ","); g_string_append (s, " {"); for (j = 0; j < mix->out_channels; j++) { if (j != 0) g_string_append (s, ","); g_string_append_printf (s, " %f", mix->matrix[i][j]); } g_string_append (s, " }"); } g_string_append (s, " }"); GST_DEBUG ("%s", s->str); g_string_free (s, TRUE); } #endif switch (format) { case GST_AUDIO_FORMAT_S16: if (flags & GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_IN) { if (flags & GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_OUT) { mix->func = (MixerFunc) gst_audio_channel_mixer_mix_int16_planar_planar; } else { mix->func = (MixerFunc) gst_audio_channel_mixer_mix_int16_planar_interleaved; } } else { if (flags & GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_OUT) { mix->func = (MixerFunc) gst_audio_channel_mixer_mix_int16_interleaved_planar; } else { mix->func = (MixerFunc) gst_audio_channel_mixer_mix_int16_interleaved_interleaved; } } break; case GST_AUDIO_FORMAT_S32: if (flags & GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_IN) { if (flags & GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_OUT) { mix->func = (MixerFunc) gst_audio_channel_mixer_mix_int32_planar_planar; } else { mix->func = (MixerFunc) gst_audio_channel_mixer_mix_int32_planar_interleaved; } } else { if (flags & GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_OUT) { mix->func = (MixerFunc) gst_audio_channel_mixer_mix_int32_interleaved_planar; } else { mix->func = (MixerFunc) gst_audio_channel_mixer_mix_int32_interleaved_interleaved; } } break; case GST_AUDIO_FORMAT_F32: if (flags & GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_IN) { if (flags & GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_OUT) { mix->func = (MixerFunc) gst_audio_channel_mixer_mix_float_planar_planar; } else { mix->func = (MixerFunc) gst_audio_channel_mixer_mix_float_planar_interleaved; } } else { if (flags & GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_OUT) { mix->func = (MixerFunc) gst_audio_channel_mixer_mix_float_interleaved_planar; } else { mix->func = (MixerFunc) gst_audio_channel_mixer_mix_float_interleaved_interleaved; } } break; case GST_AUDIO_FORMAT_F64: if (flags & GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_IN) { if (flags & GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_OUT) { mix->func = (MixerFunc) gst_audio_channel_mixer_mix_double_planar_planar; } else { mix->func = (MixerFunc) gst_audio_channel_mixer_mix_double_planar_interleaved; } } else { if (flags & GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_OUT) { mix->func = (MixerFunc) gst_audio_channel_mixer_mix_double_interleaved_planar; } else { mix->func = (MixerFunc) gst_audio_channel_mixer_mix_double_interleaved_interleaved; } } break; default: g_assert_not_reached (); break; } return mix; } /** * gst_audio_channel_mixer_new: (skip): * @flags: #GstAudioChannelMixerFlags * @in_channels: number of input channels * @in_position: positions of input channels * @out_channels: number of output channels * @out_position: positions of output channels * * Create a new channel mixer object for the given parameters. * * Returns: a new #GstAudioChannelMixer object. * Free with gst_audio_channel_mixer_free() after usage. */ GstAudioChannelMixer * gst_audio_channel_mixer_new (GstAudioChannelMixerFlags flags, GstAudioFormat format, gint in_channels, GstAudioChannelPosition * in_position, gint out_channels, GstAudioChannelPosition * out_position) { gfloat **matrix; g_return_val_if_fail (format == GST_AUDIO_FORMAT_S16 || format == GST_AUDIO_FORMAT_S32 || format == GST_AUDIO_FORMAT_F32 || format == GST_AUDIO_FORMAT_F64, NULL); matrix = gst_audio_channel_mixer_setup_matrix (flags, in_channels, in_position, out_channels, out_position); return gst_audio_channel_mixer_new_with_matrix (flags, format, in_channels, out_channels, matrix); } /** * gst_audio_channel_mixer_is_passthrough: * @mix: a #GstAudioChannelMixer * * Check if @mix is in passthrough. * * Only N x N mix identity matrices are considered passthrough, * this is determined by comparing the contents of the matrix * with 0.0 and 1.0. * * As this is floating point comparisons, if the values have been * generated, they should be rounded up or down by explicit * assignment of 0.0 or 1.0 to values within a user-defined * epsilon, this code doesn't make assumptions as to what may * constitute an appropriate epsilon. * * Returns: %TRUE is @mix is passthrough. */ gboolean gst_audio_channel_mixer_is_passthrough (GstAudioChannelMixer * mix) { gint i, j; gboolean res; /* only NxN matrices can be identities */ if (mix->in_channels != mix->out_channels) return FALSE; res = TRUE; for (i = 0; i < mix->in_channels; i++) { for (j = 0; j < mix->out_channels; j++) { if ((i == j && mix->matrix[i][j] != 1.0f) || (i != j && mix->matrix[i][j] != 0.0f)) { res = FALSE; break; } } } return res; } /** * gst_audio_channel_mixer_samples: * @mix: a #GstAudioChannelMixer * @in: input samples * @out: output samples * @samples: number of samples * * In case the samples are interleaved, @in and @out must point to an * array with a single element pointing to a block of interleaved samples. * * If non-interleaved samples are used, @in and @out must point to an * array with pointers to memory blocks, one for each channel. * * Perform channel mixing on @in_data and write the result to @out_data. * @in_data and @out_data need to be in @format and @layout. */ void gst_audio_channel_mixer_samples (GstAudioChannelMixer * mix, const gpointer in[], gpointer out[], gint samples) { g_return_if_fail (mix != NULL); g_return_if_fail (mix->matrix != NULL); mix->func (mix, in, out, samples); }