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714 lines
22 KiB
C
714 lines
22 KiB
C
/* GStreamer OSS4 audio plugin
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* Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <sys/ioctl.h>
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#include <sys/types.h>
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#include <sys/stat.h>
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#include <fcntl.h>
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#include <unistd.h>
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#include <string.h>
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#include "gst/gst-i18n-plugin.h"
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#include <gst/audio/multichannel.h>
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#include "oss4-audio.h"
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#include "oss4-mixer.h"
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#include "oss4-property-probe.h"
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#include "oss4-sink.h"
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#include "oss4-source.h"
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#include "oss4-soundcard.h"
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GST_DEBUG_CATEGORY (oss4mixer_debug);
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GST_DEBUG_CATEGORY (oss4sink_debug);
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GST_DEBUG_CATEGORY (oss4src_debug);
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GST_DEBUG_CATEGORY (oss4_debug);
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#define GST_CAT_DEFAULT oss4_debug
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typedef struct
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{
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const GstBufferFormat gst_fmt;
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const gint oss_fmt;
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const gchar name[16];
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const gint depth;
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const gint width;
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const gint endianness;
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const gboolean signedness;
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} GstOss4AudioFormat;
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/* *INDENT-OFF* */
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static const GstOss4AudioFormat fmt_map[] = {
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/* note: keep sorted by preference, prefered formats first */
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{
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GST_MU_LAW, AFMT_MU_LAW, "audio/x-mulaw", 0, 0, 0, FALSE}, {
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GST_A_LAW, AFMT_A_LAW, "audio/x-alaw", 0, 0, 0, FALSE}, {
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GST_S32_LE, AFMT_S32_LE, "audio/x-raw-int", 32, 32, G_LITTLE_ENDIAN, TRUE}, {
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GST_S32_BE, AFMT_S32_BE, "audio/x-raw-int", 32, 32, G_BIG_ENDIAN, TRUE}, {
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GST_S24_LE, AFMT_S24_LE, "audio/x-raw-int", 24, 32, G_LITTLE_ENDIAN, TRUE}, {
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GST_S24_BE, AFMT_S24_BE, "audio/x-raw-int", 24, 32, G_BIG_ENDIAN, TRUE}, {
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GST_S24_3LE, AFMT_S24_PACKED, "audio/x-raw-int", 24, 24, G_LITTLE_ENDIAN,
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TRUE}, {
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GST_S16_LE, AFMT_S16_LE, "audio/x-raw-int", 16, 16, G_LITTLE_ENDIAN, TRUE}, {
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GST_S16_BE, AFMT_S16_BE, "audio/x-raw-int", 16, 16, G_BIG_ENDIAN, TRUE}, {
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GST_U16_LE, AFMT_U16_LE, "audio/x-raw-int", 16, 16, G_LITTLE_ENDIAN, FALSE}, {
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GST_U16_BE, AFMT_U16_BE, "audio/x-raw-int", 16, 16, G_BIG_ENDIAN, FALSE}, {
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GST_S8, AFMT_S8, "audio/x-raw-int", 8, 8, 0, TRUE}, {
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GST_U8, AFMT_U8, "audio/x-raw-int", 8, 8, 0, FALSE}
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};
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/* *INDENT-ON* */
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/* formats we assume the OSS4 layer can always handle and convert internally */
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#define CONVERTIBLE_FORMATS ( \
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AFMT_MU_LAW | AFMT_A_LAW | \
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AFMT_S32_LE | AFMT_S32_BE | \
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AFMT_S24_LE | AFMT_S24_BE | \
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AFMT_S24_PACKED | \
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AFMT_S16_LE | AFMT_S16_BE | \
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AFMT_U16_LE | AFMT_U16_BE | \
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AFMT_S8 | AFMT_U8 )
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static void
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gst_oss4_append_format_to_caps (const GstOss4AudioFormat * fmt, GstCaps * caps)
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{
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GstStructure *s;
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s = gst_structure_empty_new (fmt->name);
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if (fmt->width != 0 && fmt->depth != 0) {
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gst_structure_set (s, "width", G_TYPE_INT, fmt->width, "depth", G_TYPE_INT,
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fmt->depth, "signed", G_TYPE_BOOLEAN, fmt->signedness, NULL);
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}
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if (fmt->endianness != 0) {
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gst_structure_set (s, "endianness", G_TYPE_INT, fmt->endianness, NULL);
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}
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gst_caps_append_structure (caps, s);
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}
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static gint
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gst_oss4_audio_get_oss_format (GstBufferFormat fmt)
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{
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guint i;
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for (i = 0; i < G_N_ELEMENTS (fmt_map); ++i) {
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if (fmt_map[i].gst_fmt == fmt)
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return fmt_map[i].oss_fmt;
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}
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return 0;
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}
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/* These are pretty random */
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#define GST_OSS4_MIN_SAMPLE_RATE 1
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#define GST_OSS4_MAX_SAMPLE_RATE 192000
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static gboolean
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gst_oss4_audio_detect_rates (GstObject * obj, oss_audioinfo * ai,
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GstCaps * caps)
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{
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GValue val = { 0, };
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int minrate, maxrate, i;
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minrate = ai->min_rate;
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maxrate = ai->max_rate;
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/* sanity check */
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if (minrate > maxrate) {
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GST_WARNING_OBJECT (obj, "min_rate %d > max_rate %d (buggy driver?)",
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minrate, maxrate);
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maxrate = ai->min_rate; /* swap */
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minrate = ai->max_rate;
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}
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/* limit to something sensible */
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if (minrate < GST_OSS4_MIN_SAMPLE_RATE)
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minrate = GST_OSS4_MIN_SAMPLE_RATE;
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if (maxrate > GST_OSS4_MAX_SAMPLE_RATE)
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maxrate = GST_OSS4_MAX_SAMPLE_RATE;
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if (maxrate < GST_OSS4_MIN_SAMPLE_RATE) {
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GST_WARNING_OBJECT (obj, "max_rate < %d, which makes no sense",
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GST_OSS4_MIN_SAMPLE_RATE);
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return FALSE;
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}
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GST_LOG_OBJECT (obj, "min_rate %d, max_rate %d (originally: %d, %d)",
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minrate, maxrate, ai->min_rate, ai->max_rate);
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if ((ai->caps & PCM_CAP_FREERATE)) {
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GST_LOG_OBJECT (obj, "device supports any sample rate between min and max");
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if (minrate == maxrate) {
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g_value_init (&val, G_TYPE_INT);
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g_value_set_int (&val, maxrate);
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} else {
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g_value_init (&val, GST_TYPE_INT_RANGE);
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gst_value_set_int_range (&val, minrate, maxrate);
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}
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} else {
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GST_LOG_OBJECT (obj, "%d sample rates:", ai->nrates);
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g_value_init (&val, GST_TYPE_LIST);
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for (i = 0; i < ai->nrates; ++i) {
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GST_LOG_OBJECT (obj, " rate: %d", ai->rates[i]);
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if (ai->rates[i] >= minrate && ai->rates[i] <= maxrate) {
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GValue rate_val = { 0, };
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g_value_init (&rate_val, G_TYPE_INT);
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g_value_set_int (&rate_val, ai->rates[i]);
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gst_value_list_append_value (&val, &rate_val);
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g_value_unset (&rate_val);
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}
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}
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if (gst_value_list_get_size (&val) == 0) {
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g_value_unset (&val);
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return FALSE;
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}
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}
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for (i = 0; i < gst_caps_get_size (caps); ++i) {
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GstStructure *s;
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s = gst_caps_get_structure (caps, i);
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gst_structure_set_value (s, "rate", &val);
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}
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g_value_unset (&val);
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return TRUE;
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}
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static void
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gst_oss4_audio_add_channel_layout (GstObject * obj, guint64 layout,
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guint num_channels, GstStructure * s)
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{
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const GstAudioChannelPosition pos_map[16] = {
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GST_AUDIO_CHANNEL_POSITION_NONE, /* 0 = dunno */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, /* 1 = left */
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, /* 2 = right */
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, /* 3 = center */
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GST_AUDIO_CHANNEL_POSITION_LFE, /* 4 = lfe */
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GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, /* 5 = left surround */
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GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT, /* 6 = right surround */
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, /* 7 = left rear */
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, /* 8 = right rear */
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GST_AUDIO_CHANNEL_POSITION_NONE,
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GST_AUDIO_CHANNEL_POSITION_NONE,
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GST_AUDIO_CHANNEL_POSITION_NONE,
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GST_AUDIO_CHANNEL_POSITION_NONE,
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GST_AUDIO_CHANNEL_POSITION_NONE,
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GST_AUDIO_CHANNEL_POSITION_NONE,
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GST_AUDIO_CHANNEL_POSITION_NONE
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};
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GstAudioChannelPosition ch_layout[8] = { 0, };
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guint speaker_pos; /* speaker position as defined by OSS */
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guint i;
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g_return_if_fail (num_channels <= G_N_ELEMENTS (ch_layout));
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for (i = 0; i < num_channels; ++i) {
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/* layout contains up to 16 speaker positions, with each taking up 4 bits */
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speaker_pos = (guint) ((layout >> (i * 4)) & 0x0f);
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/* if it's a channel position that's unknown to us, set all to NONE and
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* bail out */
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if (G_UNLIKELY (pos_map[speaker_pos] == GST_AUDIO_CHANNEL_POSITION_NONE))
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goto no_layout;
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ch_layout[i] = pos_map[speaker_pos];
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}
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gst_audio_set_channel_positions (s, ch_layout);
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return;
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no_layout:
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{
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/* only warn if it's really unknown, position 0 is ok and represents NONE
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* (in which case we also just set all others to NONE ignoring the other
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* positions in the OSS-given layout, because that's what we currently
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* require in GStreamer) */
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if (speaker_pos != 0) {
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GST_WARNING_OBJECT (obj, "unknown OSS channel position %x", ch_layout[i]);
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}
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for (i = 0; i < num_channels; ++i) {
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ch_layout[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
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}
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gst_audio_set_channel_positions (s, ch_layout);
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return;
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}
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}
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/* arbitrary max. limit */
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#define GST_OSS4_MIN_CHANNELS 1
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#define GST_OSS4_MAX_CHANNELS 4096
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/* takes ownership of the input caps */
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static GstCaps *
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gst_oss4_audio_detect_channels (GstObject * obj, int fd, oss_audioinfo * ai,
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GstCaps * in_caps)
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{
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const gchar *forced_layout;
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GstStructure *s = NULL;
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guint64 layout = 0;
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GstCaps *chan_caps = NULL;
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GstCaps *out_caps = NULL;
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int minchans, maxchans;
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int c, i, j;
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/* GST_OSS4_CHANNEL_LAYOUT environment variable: may be used to force a
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* particular channel layout (if it contains an odd number of channel
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* positions it will also make us advertise a channel layout for that
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* channel count, even if we'd usually skip it; this is especially useful
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* for folks with 2.1 speakers, I guess) */
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forced_layout = g_getenv ("GST_OSS4_CHANNEL_LAYOUT");
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minchans = ai->min_channels;
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maxchans = ai->max_channels;
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/* sanity check */
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if (minchans > maxchans) {
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GST_WARNING_OBJECT (obj, "min_chans %d > max_chans %d (buggy driver?)",
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minchans, maxchans);
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maxchans = ai->min_channels; /* swap */
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minchans = ai->max_channels;
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}
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/* limit to something sensible */
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if (minchans < GST_OSS4_MIN_CHANNELS)
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minchans = GST_OSS4_MIN_CHANNELS;
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if (maxchans > GST_OSS4_MAX_CHANNELS)
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maxchans = GST_OSS4_MAX_CHANNELS;
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if (maxchans < GST_OSS4_MIN_CHANNELS) {
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GST_WARNING_OBJECT (obj, "max_chans < %d, which makes no sense",
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GST_OSS4_MIN_CHANNELS);
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gst_caps_unref (in_caps);
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return NULL;
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}
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GST_LOG_OBJECT (obj, "min_channels %d, max_channels %d (originally: %d, %d)",
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minchans, maxchans, ai->min_channels, ai->max_channels);
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chan_caps = gst_caps_new_empty ();
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/* first do the simple cases: mono + stereo (channel layout implied) */
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if (minchans == 1 && maxchans == 1)
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s = gst_structure_new ("x", "channels", G_TYPE_INT, 1, NULL);
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else if (minchans == 2 && maxchans >= 2)
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s = gst_structure_new ("x", "channels", G_TYPE_INT, 2, NULL);
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else if (minchans == 1 && maxchans >= 2)
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s = gst_structure_new ("x", "channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
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gst_caps_append_structure (chan_caps, s);
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s = NULL;
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/* TODO: we assume all drivers use a left/right layout for stereo here */
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if (maxchans <= 2)
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goto done;
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if (ioctl (fd, SNDCTL_DSP_GET_CHNORDER, &layout) == -1) {
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GST_WARNING_OBJECT (obj, "couldn't query channel layout, assuming default");
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layout = CHNORDER_NORMAL;
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}
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GST_DEBUG_OBJECT (obj, "channel layout: %08" G_GINT64_MODIFIER "x", layout);
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/* e.g. forced 2.1 layout would be GST_OSS4_CHANNEL_LAYOUT=421 */
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if (forced_layout != NULL && *forced_layout != '\0') {
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guint layout_len;
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layout_len = strlen (forced_layout);
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if (layout_len >= minchans && layout_len <= maxchans) {
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layout = g_ascii_strtoull (forced_layout, NULL, 16);
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maxchans = layout_len;
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GST_DEBUG_OBJECT (obj, "forced channel layout: %08" G_GINT64_MODIFIER "x"
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" ('%s'), maxchans now %d", layout, forced_layout, maxchans);
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} else {
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GST_WARNING_OBJECT (obj, "ignoring forced channel layout: layout has %d "
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"channel positions but maxchans is %d", layout_len, maxchans);
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}
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}
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/* need to advertise channel layouts for anything >2 and <=8 channels */
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for (c = MAX (3, minchans); c <= MIN (maxchans, 8); c++) {
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/* "The min_channels and max_channels fields define the limits for the
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* number of channels. However some devices don't support all channels
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* within this range. It's possible that the odd values (3, 5, 7, 9, etc).
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* are not supported. There is currently no way to check for this other
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* than checking if SNDCTL_DSP_CHANNELS accepts the requested value.
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* Another approach is trying to avoid using odd number of channels."
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*
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* So, we don't know for sure if these odd values are supported:
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*/
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if ((c == 3 || c == 5 || c == 7) && (c != maxchans)) {
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GST_LOG_OBJECT (obj, "not adding layout with %d channels", c);
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continue;
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}
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s = gst_structure_new ("x", "channels", G_TYPE_INT, c, NULL);
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gst_oss4_audio_add_channel_layout (obj, layout, c, s);
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GST_LOG_OBJECT (obj, "c=%u, appending struct %" GST_PTR_FORMAT, c, s);
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gst_caps_append_structure (chan_caps, s);
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s = NULL;
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}
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if (maxchans <= 8)
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goto done;
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/* for everything >8 channels, CHANNEL_POSITION_NONE is implied. */
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if (minchans == maxchans || maxchans == 9) {
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s = gst_structure_new ("x", "channels", G_TYPE_INT, maxchans, NULL);
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} else {
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s = gst_structure_new ("x", "channels", GST_TYPE_INT_RANGE,
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MAX (9, minchans), maxchans, NULL);
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}
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gst_caps_append_structure (chan_caps, s);
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s = NULL;
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done:
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GST_LOG_OBJECT (obj, "channel structures: %" GST_PTR_FORMAT, chan_caps);
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out_caps = gst_caps_new_empty ();
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/* combine each structure in the input caps with each channel caps struct */
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for (i = 0; i < gst_caps_get_size (in_caps); ++i) {
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const GstStructure *in_s;
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in_s = gst_caps_get_structure (in_caps, i);
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for (j = 0; j < gst_caps_get_size (chan_caps); ++j) {
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const GstStructure *chan_s;
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const GValue *val;
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s = gst_structure_copy (in_s);
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chan_s = gst_caps_get_structure (chan_caps, j);
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if ((val = gst_structure_get_value (chan_s, "channels")))
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gst_structure_set_value (s, "channels", val);
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if ((val = gst_structure_get_value (chan_s, "channel-positions")))
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gst_structure_set_value (s, "channel-positions", val);
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gst_caps_append_structure (out_caps, s);
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s = NULL;
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}
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}
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gst_caps_unref (in_caps);
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gst_caps_unref (chan_caps);
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return out_caps;
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}
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GstCaps *
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gst_oss4_audio_probe_caps (GstObject * obj, int fd)
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{
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oss_audioinfo ai = { 0, };
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gboolean output;
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GstCaps *caps;
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int nonnative_formats = 0;
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int formats, i;
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output = GST_IS_OSS4_SINK (obj);
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/* -1 = get info for currently open device (fd). This will fail with
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* OSS build <= 1013 because of a bug in OSS */
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ai.dev = -1;
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if (ioctl (fd, SNDCTL_ENGINEINFO, &ai) == -1)
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goto engineinfo_failed;
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formats = (output) ? ai.oformats : ai.iformats;
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|
|
GST_LOG_OBJECT (obj, "%s formats : 0x%08x", (output) ? "out" : "in", formats);
|
|
|
|
caps = gst_caps_new_empty ();
|
|
|
|
/* first list all the formats natively supported */
|
|
for (i = 0; i < G_N_ELEMENTS (fmt_map); ++i) {
|
|
if ((formats & fmt_map[i].oss_fmt)) {
|
|
gst_oss4_append_format_to_caps (&fmt_map[i], caps);
|
|
} else if ((fmt_map[i].oss_fmt & CONVERTIBLE_FORMATS)) {
|
|
nonnative_formats |= fmt_map[i].oss_fmt;
|
|
}
|
|
}
|
|
|
|
GST_LOG_OBJECT (obj, "adding non-native %s formats : 0x%08x",
|
|
(output) ? "out" : "in", nonnative_formats);
|
|
|
|
/* now append non-native formats for which conversion would be needed */
|
|
for (i = 0; i < G_N_ELEMENTS (fmt_map); ++i) {
|
|
if ((nonnative_formats & fmt_map[i].oss_fmt)) {
|
|
gst_oss4_append_format_to_caps (&fmt_map[i], caps);
|
|
}
|
|
}
|
|
|
|
gst_caps_do_simplify (caps);
|
|
GST_LOG_OBJECT (obj, "formats: %" GST_PTR_FORMAT, caps);
|
|
|
|
if (!gst_oss4_audio_detect_rates (obj, &ai, caps))
|
|
goto detect_rates_failed;
|
|
|
|
caps = gst_oss4_audio_detect_channels (obj, fd, &ai, caps);
|
|
if (caps == NULL)
|
|
goto detect_channels_failed;
|
|
|
|
GST_LOG_OBJECT (obj, "probed caps: %" GST_PTR_FORMAT, caps);
|
|
|
|
return caps;
|
|
|
|
/* ERRORS */
|
|
engineinfo_failed:
|
|
{
|
|
GST_WARNING ("ENGINEINFO supported formats probe failed: %s",
|
|
g_strerror (errno));
|
|
return NULL;
|
|
}
|
|
detect_rates_failed:
|
|
{
|
|
GST_WARNING_OBJECT (obj, "failed to detect supported sample rates");
|
|
gst_caps_unref (caps);
|
|
return NULL;
|
|
}
|
|
detect_channels_failed:
|
|
{
|
|
GST_WARNING_OBJECT (obj, "failed to detect supported channels");
|
|
gst_caps_unref (caps);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
GstCaps *
|
|
gst_oss4_audio_get_template_caps (void)
|
|
{
|
|
GstCaps *caps;
|
|
gint i;
|
|
|
|
caps = gst_caps_new_empty ();
|
|
|
|
for (i = 0; i < G_N_ELEMENTS (fmt_map); ++i) {
|
|
gst_oss4_append_format_to_caps (&fmt_map[i], caps);
|
|
}
|
|
|
|
gst_caps_do_simplify (caps);
|
|
|
|
for (i = 0; i < gst_caps_get_size (caps); ++i) {
|
|
GstStructure *s;
|
|
|
|
s = gst_caps_get_structure (caps, i);
|
|
gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, GST_OSS4_MIN_SAMPLE_RATE,
|
|
GST_OSS4_MAX_SAMPLE_RATE, "channels", GST_TYPE_INT_RANGE,
|
|
GST_OSS4_MIN_CHANNELS, GST_OSS4_MAX_CHANNELS, NULL);
|
|
}
|
|
|
|
return caps;
|
|
}
|
|
|
|
/* called by gst_oss4_sink_prepare() and gst_oss4_source_prepare() */
|
|
gboolean
|
|
gst_oss4_audio_set_format (GstObject * obj, int fd, GstRingBufferSpec * spec)
|
|
{
|
|
struct audio_buf_info info = { 0, };
|
|
int fmt, chans, rate;
|
|
|
|
fmt = gst_oss4_audio_get_oss_format (spec->format);
|
|
if (fmt == 0)
|
|
goto wrong_format;
|
|
|
|
if (spec->type == GST_BUFTYPE_LINEAR && spec->width != 32 &&
|
|
spec->width != 24 && spec->width != 16 && spec->width != 8) {
|
|
goto dodgy_width;
|
|
}
|
|
|
|
/* format */
|
|
GST_LOG_OBJECT (obj, "setting format: %d", fmt);
|
|
if (ioctl (fd, SNDCTL_DSP_SETFMT, &fmt) == -1)
|
|
goto set_format_failed;
|
|
|
|
/* channels */
|
|
GST_LOG_OBJECT (obj, "setting channels: %d", spec->channels);
|
|
chans = spec->channels;
|
|
if (ioctl (fd, SNDCTL_DSP_CHANNELS, &chans) == -1)
|
|
goto set_channels_failed;
|
|
|
|
/* rate */
|
|
GST_LOG_OBJECT (obj, "setting rate: %d", spec->rate);
|
|
rate = spec->rate;
|
|
if (ioctl (fd, SNDCTL_DSP_SPEED, &rate) == -1)
|
|
goto set_rate_failed;
|
|
|
|
GST_DEBUG_OBJECT (obj, "effective format : %d", fmt);
|
|
GST_DEBUG_OBJECT (obj, "effective channels : %d", chans);
|
|
GST_DEBUG_OBJECT (obj, "effective rate : %d", rate);
|
|
|
|
/* make sure format, channels, and rate are the ones we requested */
|
|
if (fmt != gst_oss4_audio_get_oss_format (spec->format) ||
|
|
chans != spec->channels || rate != spec->rate) {
|
|
/* This shouldn't happen, but hey */
|
|
goto format_not_what_was_requested;
|
|
}
|
|
|
|
if (GST_IS_OSS4_SOURCE (obj)) {
|
|
if (ioctl (fd, SNDCTL_DSP_GETISPACE, &info) == -1)
|
|
goto get_ispace_failed;
|
|
} else {
|
|
if (ioctl (fd, SNDCTL_DSP_GETOSPACE, &info) == -1)
|
|
goto get_ospace_failed;
|
|
}
|
|
|
|
spec->segsize = info.fragsize;
|
|
|
|
/* we add some extra fragments -- this helps us account for delays due to
|
|
* conversion buffer, streams queueing, etc. It is important that these
|
|
* be taken into account because otherwise the delay counter can wind up
|
|
* being too large, and the buffer will wrap. */
|
|
spec->segtotal = info.fragstotal + 4;
|
|
|
|
spec->bytes_per_sample = (spec->width / 8) * spec->channels;
|
|
|
|
GST_DEBUG_OBJECT (obj, "got segsize: %d, segtotal: %d, value: %08x",
|
|
spec->segsize, spec->segtotal, info.fragsize);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
wrong_format:
|
|
{
|
|
GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to get format %d", spec->format));
|
|
return FALSE;
|
|
}
|
|
dodgy_width:
|
|
{
|
|
GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
|
|
("unexpected width %d", spec->width));
|
|
return FALSE;
|
|
}
|
|
set_format_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
|
|
("DSP_SETFMT(%d) failed: %s", fmt, g_strerror (errno)));
|
|
return FALSE;
|
|
}
|
|
set_channels_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
|
|
("DSP_CHANNELS(%d) failed: %s", chans, g_strerror (errno)));
|
|
return FALSE;
|
|
}
|
|
set_rate_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
|
|
("DSP_SPEED(%d) failed: %s", rate, g_strerror (errno)));
|
|
return FALSE;
|
|
}
|
|
get_ospace_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
|
|
("DSP_GETOSPACE failed: %s", g_strerror (errno)));
|
|
return FALSE;
|
|
}
|
|
get_ispace_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
|
|
("DSP_GETISPACE failed: %s", g_strerror (errno)));
|
|
return FALSE;
|
|
}
|
|
format_not_what_was_requested:
|
|
{
|
|
GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
|
|
("Format actually configured wasn't the one we requested. This is "
|
|
"probably either a bug in the driver or in the format probing code."));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
int
|
|
gst_oss4_audio_get_version (GstObject * obj, int fd)
|
|
{
|
|
gint ver = 0;
|
|
|
|
/* we use the old ioctl here on purpose instead of SNDCTL_SYSINFO */
|
|
if (ioctl (fd, OSS_GETVERSION, &ver) < 0) {
|
|
GST_LOG_OBJECT (obj, "OSS_GETVERSION failed: %s", g_strerror (errno));
|
|
return -1;
|
|
}
|
|
GST_LOG_OBJECT (obj, "OSS version: 0x%08x", ver);
|
|
return ver;
|
|
}
|
|
|
|
gboolean
|
|
gst_oss4_audio_check_version (GstObject * obj, int fd)
|
|
{
|
|
return (gst_oss4_audio_get_version (obj, fd) >= GST_MIN_OSS4_VERSION);
|
|
}
|
|
|
|
gchar *
|
|
gst_oss4_audio_find_device (GstObject * oss)
|
|
{
|
|
GValueArray *arr;
|
|
gchar *ret = NULL;
|
|
|
|
arr = gst_property_probe_probe_and_get_values_name (GST_PROPERTY_PROBE (oss),
|
|
"device");
|
|
|
|
if (arr != NULL) {
|
|
if (arr->n_values > 0) {
|
|
const GValue *val;
|
|
|
|
val = g_value_array_get_nth (arr, 0);
|
|
ret = g_value_dup_string (val);
|
|
}
|
|
g_value_array_free (arr);
|
|
}
|
|
|
|
GST_LOG_OBJECT (oss, "first device found: %s", GST_STR_NULL (ret));
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
gint rank;
|
|
|
|
GST_DEBUG_CATEGORY_INIT (oss4sink_debug, "oss4sink", 0, "OSS4 audio sink");
|
|
GST_DEBUG_CATEGORY_INIT (oss4src_debug, "oss4src", 0, "OSS4 audio src");
|
|
GST_DEBUG_CATEGORY_INIT (oss4mixer_debug, "oss4mixer", 0, "OSS4 mixer");
|
|
GST_DEBUG_CATEGORY_INIT (oss4_debug, "oss4", 0, "OSS4 plugin");
|
|
|
|
#ifdef ENABLE_NLS
|
|
GST_DEBUG ("binding text domain %s to locale dir %s", GETTEXT_PACKAGE,
|
|
LOCALEDIR);
|
|
bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
|
|
bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
|
|
#endif
|
|
|
|
/* we want a higher rank than the legacy OSS elements have now */
|
|
rank = GST_RANK_SECONDARY + 1;
|
|
|
|
if (!gst_element_register (plugin, "oss4sink", rank, GST_TYPE_OSS4_SINK) ||
|
|
!gst_element_register (plugin, "oss4src", rank, GST_TYPE_OSS4_SOURCE) ||
|
|
!gst_element_register (plugin, "oss4mixer", rank, GST_TYPE_OSS4_MIXER)) {
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
"oss4",
|
|
"Open Sound System (OSS) version 4 support for GStreamer",
|
|
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
|