mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-18 15:51:11 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst-libs/gst/audio/audio-channel-mixer.c

1076 lines
35 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
* Copyright (C) 2008 Sebastian Dröge <slomo@circular-chaos.org>
*
* 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 <math.h>
#include <string.h>
#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_slice_free (GstAudioChannelMixer, 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.
*/
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;
gst_audio_channel_mixer_fill_identical (matrix, in_channels, in_position,
out_channels, out_position, flags);
if (!(flags & GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_IN)) {
gst_audio_channel_mixer_fill_compatible (matrix, in_channels, in_position,
out_channels, out_position);
gst_audio_channel_mixer_fill_others (matrix, in_channels, in_position,
out_channels, out_position);
gst_audio_channel_mixer_fill_normalize (matrix, in_channels, out_channels);
}
}
/* 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, or %NULL if @format isn't supported,
* @matrix is invalid, or @matrix is %NULL and @in_channels != @out_channels.
* 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);
g_return_val_if_fail (in_channels > 0 && in_channels < 64, NULL);
g_return_val_if_fail (out_channels > 0 && out_channels < 64, NULL);
mix = g_slice_new0 (GstAudioChannelMixer);
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, or %NULL if @format isn't supported.
* 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);
g_return_val_if_fail (in_channels > 0 && in_channels < 64, NULL);
g_return_val_if_fail (out_channels > 0 && out_channels < 64, 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);
}
Reference in a new issue View git blame Copy permalink