+2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+
+ * ext/a52dec/gsta52dec.c: (gst_a52dec_channels), (gst_a52dec_push),
+ (gst_a52dec_reneg), (gst_a52dec_loop), (plugin_init):
+ * ext/alsa/gstalsa.c: (gst_alsa_get_caps):
+ * ext/alsa/gstalsaplugin.c: (plugin_init):
+ * ext/dts/gstdtsdec.c: (gst_dtsdec_channels),
+ (gst_dtsdec_renegotiate), (gst_dtsdec_loop), (plugin_init):
+ * ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_chanpos_from_gst),
+ (gst_faad_chanpos_to_gst), (gst_faad_sinkconnect),
+ (gst_faad_srcgetcaps), (gst_faad_srcconnect), (gst_faad_chain),
+ (gst_faad_change_state), (plugin_init):
+ * ext/faad/gstfaad.h:
+ * ext/vorbis/vorbis.c: (plugin_init):
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/audio/audio.c: (plugin_init):
+ * gst-libs/gst/audio/multichannel.c:
+ (gst_audio_check_channel_positions),
+ (gst_audio_get_channel_positions),
+ (gst_audio_set_channel_positions),
+ (gst_audio_set_structure_channel_positions_list),
+ (add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
+ (gst_audio_fixate_channel_positions):
+ * gst-libs/gst/audio/multichannel.h:
+ * gst-libs/gst/audio/testchannels.c: (main):
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_class_init), (gst_audio_convert_init),
+ (gst_audio_convert_dispose), (gst_audio_convert_getcaps),
+ (gst_audio_convert_parse_caps), (gst_audio_convert_link),
+ (gst_audio_convert_fixate), (gst_audio_convert_channels):
+ * gst/audioconvert/plugin.c: (plugin_init):
+ Surround sound support.
+
2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_push):
#include <stdlib.h>
#include <gst/gst.h>
+#include <gst/audio/multichannel.h>
+
#include <dts.h>
#include "gstdtsdec.h"
}
static gint
-gst_dtsdec_channels (uint32_t flags)
+gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition ** pos)
{
gint chans = 0;
switch (flags & DTS_CHANNEL_MASK) {
case DTS_MONO:
chans = 1;
+ if (pos) {
+ *pos = g_new (GstAudioChannelPosition, 2);
+ *pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
+ }
break;
- case DTS_CHANNEL:
+ /* case DTS_CHANNEL: */
case DTS_STEREO:
case DTS_STEREO_SUMDIFF:
case DTS_STEREO_TOTAL:
case DTS_DOLBY:
chans = 2;
+ if (pos) {
+ *pos = g_new (GstAudioChannelPosition, 3);
+ *pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ *pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ }
break;
case DTS_3F:
+ chans = 3;
+ if (pos) {
+ *pos = g_new (GstAudioChannelPosition, 4);
+ *pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
+ *pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ *pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ }
+ break;
case DTS_2F1R:
chans = 3;
+ if (pos) {
+ *pos = g_new (GstAudioChannelPosition, 4);
+ *pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ *pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ *pos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
+ }
break;
case DTS_3F1R:
+ chans = 4;
+ if (pos) {
+ *pos = g_new (GstAudioChannelPosition, 5);
+ *pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
+ *pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ *pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ *pos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
+ }
+ break;
case DTS_2F2R:
chans = 4;
+ if (pos) {
+ *pos = g_new (GstAudioChannelPosition, 5);
+ *pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ *pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ *pos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
+ *pos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
+ }
break;
case DTS_3F2R:
chans = 5;
+ if (pos) {
+ *pos = g_new (GstAudioChannelPosition, 6);
+ *pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
+ *pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ *pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ *pos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
+ *pos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
+ }
break;
case DTS_4F2R:
chans = 6;
+ if (pos) {
+ *pos = g_new (GstAudioChannelPosition, 7);
+ *pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
+ *pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
+ *pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ *pos[3] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ *pos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
+ *pos[5] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
+ }
break;
default:
/* error */
g_warning ("dtsdec: invalid flags 0x%x", flags);
return 0;
}
- if (flags & DTS_LFE)
+ if (flags & DTS_LFE) {
+ if (pos) {
+ *pos[chans] = GST_AUDIO_CHANNEL_POSITION_LFE;
+ }
chans += 1;
+ }
return chans;
}
static gboolean
gst_dtsdec_renegotiate (GstDtsDec * dts)
{
+ GstAudioChannelPosition *pos;
GstCaps *caps = gst_caps_from_string (DTS_CAPS);
- gint channels = gst_dtsdec_channels (dts->using_channels);
+ gint channels = gst_dtsdec_channels (dts->using_channels, &pos);
+
+ if (!channels)
+ return FALSE;
GST_INFO ("dtsdec renegotiate, channels=%d, rate=%d",
channels, dts->sample_rate);
gst_caps_set_simple (caps,
"channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, (gint) dts->sample_rate, NULL);
+ gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
+ g_free (pos);
return gst_pad_set_explicit_caps (dts->srcpad, caps);
}
}
samples = dts_samples (dts->state);
- num_c = gst_dtsdec_channels (dts->using_channels);
+ num_c = gst_dtsdec_channels (dts->using_channels, NULL);
out = gst_buffer_new_and_alloc ((SAMPLE_WIDTH / 8) * 256 * num_c);
GST_BUFFER_TIMESTAMP (out) = timestamp;
GST_BUFFER_DURATION (out) = GST_SECOND * 256 / dts->sample_rate;
static gboolean
plugin_init (GstPlugin * plugin)
{
- if (!gst_library_load ("gstbytestream"))
+ if (!gst_library_load ("gstbytestream") || !gst_library_load ("gstaudio"))
return FALSE;
if (!gst_element_register (plugin, "dtsdec", GST_RANK_PRIMARY,
#include <string.h>
+#include <gst/audio/multichannel.h>
+
#include "gstfaad.h"
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) { 2, 4 }")
);
-static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
+#define STATIC_INT_CAPS(bpp) \
+ "audio/x-raw-int, " \
+ "endianness = (int) BYTE_ORDER, " \
+ "signed = (bool) TRUE, " \
+ "width = (int) " G_STRINGIFY (bpp) ", " \
+ "depth = (int) " G_STRINGIFY (bpp) ", " \
+ "rate = (int) [ 8000, 96000 ], " \
+ "channels = (int) [ 1, 8 ]"
+
+#define STATIC_FLOAT_CAPS(bpp) \
+ "audio/x-raw-float, " \
+ "endianness = (int) BYTE_ORDER, " \
+ "depth = (int) " G_STRINGIFY (bpp) ", " \
+ "rate = (int) [ 8000, 96000 ], " \
+ "channels = (int) [ 1, 8 ]"
+
+/*
+ * All except 16-bit integer are disabled until someone fixes FAAD.
+ * FAAD allocates approximately 8*1024*2 bytes bytes, which is enough
+ * for 1 frame (1024 samples) of 6 channel (5.1) 16-bit integer 16bpp
+ * audio, but not for any other. You'll get random segfaults, crashes
+ * and even valgrind goes crazy.
+ */
+
+#define STATIC_CAPS \
+ STATIC_INT_CAPS (16)
+#if 0
+"; "
+STATIC_INT_CAPS (24)
+ "; "
+STATIC_INT_CAPS (32)
+ "; "
+STATIC_FLOAT_CAPS (32)
+ "; "
+STATIC_FLOAT_CAPS (64)
+#endif
+ static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "endianness = (int) BYTE_ORDER, "
- "signed = (bool) TRUE, "
- "width = (int) { 16, 24, 32 }, "
- "depth = (int) { 16, 24, 32 }, "
- "rate = (int) [ 8000, 96000 ], "
- "channels = (int) [ 1, 6 ]; "
- "audio/x-raw-float, "
- "endianness = (int) BYTE_ORDER, "
- "depth = (int) { 32, 64 }, "
- "rate = (int) [ 8000, 96000 ], " "channels = (int) [ 1, 6 ]")
+ GST_STATIC_CAPS (STATIC_CAPS)
);
-static void gst_faad_base_init (GstFaadClass * klass);
-static void gst_faad_class_init (GstFaadClass * klass);
-static void gst_faad_init (GstFaad * faad);
+ static void gst_faad_base_init (GstFaadClass * klass);
+ static void gst_faad_class_init (GstFaadClass * klass);
+ static void gst_faad_init (GstFaad * faad);
-static GstPadLinkReturn
-gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps);
-static GstPadLinkReturn
-gst_faad_srcconnect (GstPad * pad, const GstCaps * caps);
-static GstCaps *gst_faad_srcgetcaps (GstPad * pad);
-static void gst_faad_chain (GstPad * pad, GstData * data);
-static GstElementStateReturn gst_faad_change_state (GstElement * element);
+ static GstPadLinkReturn
+ gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps);
+ static GstPadLinkReturn
+ gst_faad_srcconnect (GstPad * pad, const GstCaps * caps);
+ static GstCaps *gst_faad_srcgetcaps (GstPad * pad);
+ static void gst_faad_chain (GstPad * pad, GstData * data);
+ static GstElementStateReturn gst_faad_change_state (GstElement * element);
-static GstElementClass *parent_class = NULL;
+ static GstElementClass *parent_class = NULL;
/* static guint gst_faad_signals[LAST_SIGNAL] = { 0 }; */
faad->samplerate = -1;
faad->channels = -1;
faad->tempbuf = NULL;
+ faad->need_channel_setup = TRUE;
+ faad->channel_positions = NULL;
+ faad->init = FALSE;
GST_FLAG_SET (faad, GST_ELEMENT_EVENT_AWARE);
gst_pad_set_getcaps_function (faad->srcpad, gst_faad_srcgetcaps);
}
+/*
+ * Channel identifier conversion - caller g_free()s result!
+ */
+
+static guchar *
+gst_faad_chanpos_from_gst (GstAudioChannelPosition * pos, guint num)
+{
+ guchar *fpos = g_new (guchar, num);
+ guint n;
+
+ for (n = 0; n < num; n++) {
+ switch (pos[n]) {
+ case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT:
+ fpos[n] = FRONT_CHANNEL_LEFT;
+ break;
+ case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT:
+ fpos[n] = FRONT_CHANNEL_RIGHT;
+ break;
+ case GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER:
+ fpos[n] = FRONT_CHANNEL_CENTER;
+ break;
+ case GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT:
+ fpos[n] = SIDE_CHANNEL_LEFT;
+ break;
+ case GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT:
+ fpos[n] = SIDE_CHANNEL_RIGHT;
+ break;
+ case GST_AUDIO_CHANNEL_POSITION_REAR_LEFT:
+ fpos[n] = BACK_CHANNEL_LEFT;
+ break;
+ case GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT:
+ fpos[n] = BACK_CHANNEL_RIGHT;
+ break;
+ case GST_AUDIO_CHANNEL_POSITION_REAR_CENTER:
+ fpos[n] = BACK_CHANNEL_CENTER;
+ break;
+ case GST_AUDIO_CHANNEL_POSITION_LFE:
+ fpos[n] = LFE_CHANNEL;
+ break;
+ default:
+ GST_WARNING ("Unsupported GST channel position 0x%x encountered",
+ pos[n]);
+ g_free (fpos);
+ return NULL;
+ }
+ }
+
+ return fpos;
+}
+
+static GstAudioChannelPosition *
+gst_faad_chanpos_to_gst (guchar * fpos, guint num)
+{
+ GstAudioChannelPosition *pos = g_new (GstAudioChannelPosition, num);
+ guint n;
+
+ for (n = 0; n < num; n++) {
+ switch (fpos[n]) {
+ case FRONT_CHANNEL_LEFT:
+ pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ break;
+ case FRONT_CHANNEL_RIGHT:
+ pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ break;
+ case FRONT_CHANNEL_CENTER:
+ pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
+ break;
+ case SIDE_CHANNEL_LEFT:
+ pos[n] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
+ break;
+ case SIDE_CHANNEL_RIGHT:
+ pos[n] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
+ break;
+ case BACK_CHANNEL_LEFT:
+ pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
+ break;
+ case BACK_CHANNEL_RIGHT:
+ pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
+ break;
+ case BACK_CHANNEL_CENTER:
+ pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
+ break;
+ case LFE_CHANNEL:
+ pos[n] = GST_AUDIO_CHANNEL_POSITION_LFE;
+ break;
+ default:
+ GST_WARNING ("Unsupported FAAD channel position 0x%x encountered",
+ fpos[n]);
+ g_free (pos);
+ return NULL;
+ }
+ }
+
+ return pos;
+}
+
static GstPadLinkReturn
gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps)
{
GST_BUFFER_SIZE (buf), &samplerate, &channels) < 0)
return GST_PAD_LINK_REFUSED;
- faad->samplerate = samplerate;
- faad->channels = channels;
+ //faad->samplerate = samplerate;
+ //faad->channels = channels;
+ faad->init = TRUE;
if (faad->tempbuf) {
gst_buffer_unref (faad->tempbuf);
faad->tempbuf = NULL;
}
-
- return GST_PAD_LINK_OK;
+ } else {
+ faad->init = FALSE;
}
+ faad->need_channel_setup = TRUE;
+
/* if there's no decoderspecificdata, it's all fine. We cannot know
* much more at this point... */
return GST_PAD_LINK_OK;
gst_faad_srcgetcaps (GstPad * pad)
{
GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
+ static GstAudioChannelPosition *supported_positions = NULL;
+ static gint num_supported_positions = LFE_CHANNEL - FRONT_CHANNEL_CENTER;
+ GstCaps *templ;
+
+ if (!supported_positions) {
+ guchar *supported_fpos = g_new0 (guchar,
+ LFE_CHANNEL - FRONT_CHANNEL_CENTER);
+ gint n;
+
+ for (n = 0; n < LFE_CHANNEL - FRONT_CHANNEL_CENTER; n++) {
+ supported_fpos[n] = n + FRONT_CHANNEL_CENTER;
+ }
+ supported_positions = gst_faad_chanpos_to_gst (supported_fpos, n);
+ g_free (supported_fpos);
+ }
if (faad->handle != NULL && faad->channels != -1 && faad->samplerate != -1) {
GstCaps *caps = gst_caps_new_empty ();
GstStructure *str;
gint fmt[] = {
FAAD_FMT_16BIT,
+#if 0
FAAD_FMT_24BIT,
FAAD_FMT_32BIT,
FAAD_FMT_FLOAT,
FAAD_FMT_DOUBLE,
+#endif
-1
}
, n;
for (n = 0; fmt[n] != -1; n++) {
- switch (n) {
+ switch (fmt[n]) {
case FAAD_FMT_16BIT:
str = gst_structure_new ("audio/x-raw-int",
"signed", G_TYPE_BOOLEAN, TRUE,
"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
break;
+#if 0
case FAAD_FMT_24BIT:
str = gst_structure_new ("audio/x-raw-int",
"signed", G_TYPE_BOOLEAN, TRUE,
str = gst_structure_new ("audio/x-raw-float",
"depth", G_TYPE_INT, 64, NULL);
break;
+#endif
default:
str = NULL;
break;
if (faad->channels != -1) {
gst_structure_set (str, "channels", G_TYPE_INT, faad->channels, NULL);
+
+ /* put channel information here */
+ if (faad->channel_positions) {
+ GstAudioChannelPosition *pos;
+
+ pos = gst_faad_chanpos_to_gst (faad->channel_positions,
+ faad->channels);
+ if (!pos) {
+ gst_structure_free (str);
+ continue;
+ }
+ gst_audio_set_channel_positions (str, pos);
+ g_free (pos);
+ } else {
+ gst_audio_set_structure_channel_positions_list (str,
+ supported_positions, num_supported_positions);
+ }
} else {
- gst_structure_set (str, "channels", GST_TYPE_INT_RANGE, 1, 6, NULL);
+ gst_structure_set (str, "channels", GST_TYPE_INT_RANGE, 1, 8, NULL);
+ /* we set channel positions later */
}
gst_structure_set (str, "endianness", G_TYPE_INT, G_BYTE_ORDER, NULL);
gst_caps_append_structure (caps, str);
}
+ if (faad->channels == -1) {
+ gst_audio_set_caps_channel_positions_list (caps,
+ supported_positions, num_supported_positions);
+ }
+
return caps;
}
- return gst_caps_copy (GST_PAD_TEMPLATE_CAPS (GST_PAD_PAD_TEMPLATE (pad)));
+ /* template with channel positions */
+ templ = gst_caps_copy (GST_PAD_TEMPLATE_CAPS (GST_PAD_PAD_TEMPLATE (pad)));
+ gst_audio_set_caps_channel_positions_list (templ,
+ supported_positions, num_supported_positions);
+
+ return templ;
}
static GstPadLinkReturn
gint depth, rate, channels;
GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
- if (!faad->handle || (faad->samplerate == -1 || faad->channels == -1)) {
+ structure = gst_caps_get_structure (caps, 0);
+
+ if (!faad->handle || (faad->samplerate == -1 || faad->channels == -1) ||
+ !faad->channel_positions) {
return GST_PAD_LINK_DELAYED;
}
- structure = gst_caps_get_structure (caps, 0);
mimetype = gst_structure_get_name (structure);
/* Samplerate and channels are normally provided through
return GST_PAD_LINK_REFUSED;
}
+ /* Another internal checkup. */
+ if (faad->need_channel_setup) {
+ GstAudioChannelPosition *pos;
+ guchar *fpos;
+ guint i;
+
+ pos = gst_audio_get_channel_positions (structure);
+ if (!pos) {
+ return GST_PAD_LINK_DELAYED;
+ }
+ fpos = gst_faad_chanpos_from_gst (pos, faad->channels);
+ g_free (pos);
+ if (!fpos)
+ return GST_PAD_LINK_REFUSED;
+
+ for (i = 0; i < faad->channels; i++) {
+ if (fpos[i] != faad->channel_positions[i]) {
+ g_free (fpos);
+ return GST_PAD_LINK_REFUSED;
+ }
+ }
+ g_free (fpos);
+ }
+
if (!strcmp (mimetype, "audio/x-raw-int")) {
gint width;
case 16:
fmt = FAAD_FMT_16BIT;
break;
+#if 0
case 24:
fmt = FAAD_FMT_24BIT;
break;
case 32:
fmt = FAAD_FMT_32BIT;
break;
+#endif
}
} else {
if (!gst_structure_get_int (structure, "depth", &depth))
return GST_PAD_LINK_REFUSED;
switch (depth) {
+#if 0
case 32:
fmt = FAAD_FMT_FLOAT;
break;
case 64:
fmt = FAAD_FMT_DOUBLE;
break;
+#endif
}
}
if (fmt != -1) {
faacDecConfiguration *conf;
+ g_print ("Set format %d\n", fmt);
conf = faacDecGetCurrentConfiguration (faad->handle);
conf->outputFormat = fmt;
- faacDecSetConfiguration (faad->handle, conf);
+ g_print ("Trying to conf\n");
+ if (faacDecSetConfiguration (faad->handle, conf) == 0)
+ return GST_PAD_LINK_REFUSED;
+ g_print ("Done\n");
/* FIXME: handle return value, how? */
faad->bps = depth / 8;
return GST_PAD_LINK_OK;
}
-
+ g_print ("Format not recognized\n");
return GST_PAD_LINK_REFUSED;
}
guchar *input_data;
GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
GstBuffer *buf, *outbuf;
- faacDecFrameInfo info;
+ faacDecFrameInfo *info;
void *out;
if (GST_IS_EVENT (data)) {
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
if (faad->tempbuf != NULL) {
- /* Try to decode the remaining data */
- out = faacDecDecode (faad->handle, &info,
- GST_BUFFER_DATA (faad->tempbuf), GST_BUFFER_SIZE (faad->tempbuf));
gst_buffer_unref (faad->tempbuf);
faad->tempbuf = NULL;
- if (out && !info.error && info.samples > 0) {
- outbuf = gst_buffer_new_and_alloc (info.samples * faad->bps);
- /* ugh */
- memcpy (GST_BUFFER_DATA (outbuf), out, GST_BUFFER_SIZE (outbuf));
-
- gst_pad_push (faad->srcpad, GST_DATA (outbuf));
- }
}
gst_element_set_eos (GST_ELEMENT (faad));
gst_pad_push (faad->srcpad, data);
}
}
+ info = g_new0 (faacDecFrameInfo, 1);
+
+ /* buffer + remaining data */
buf = GST_BUFFER (data);
+ if (faad->tempbuf) {
+ buf = gst_buffer_join (faad->tempbuf, buf);
+ faad->tempbuf = NULL;
+ }
- if (faad->samplerate == -1 || faad->channels == -1) {
- GstPadLinkReturn ret;
+ /* init if not already done during capsnego */
+ if (!faad->init) {
gulong samplerate;
guchar channels;
faacDecInit (faad->handle,
GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf), &samplerate, &channels);
- faad->samplerate = samplerate;
- faad->channels = channels;
-
- ret = gst_pad_renegotiate (faad->srcpad);
- if (GST_PAD_LINK_FAILED (ret)) {
- GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL), (NULL));
- gst_buffer_unref (buf);
- return;
- }
- }
+ faad->init = TRUE;
- /* Use the leftovers */
- if (faad->tempbuf) {
- buf = gst_buffer_join (faad->tempbuf, buf);
- faad->tempbuf = NULL;
+ /* store for renegotiation later on */
+ info->samplerate = samplerate;
+ info->channels = channels;
+ } else {
+ info->samplerate = 0;
+ info->channels = 0;
}
+ /* decode cycle */
input_data = GST_BUFFER_DATA (buf);
input_size = GST_BUFFER_SIZE (buf);
- info.bytesconsumed = input_size;
- while (input_size > (faad->channels * FAAD_MIN_STREAMSIZE)
- && info.bytesconsumed > 0) {
- out = faacDecDecode (faad->handle, &info, input_data, input_size);
- if (info.error) {
+ info->bytesconsumed = input_size;
+ while (input_size >= FAAD_MIN_STREAMSIZE && info->bytesconsumed > 0) {
+ g_print ("Decoding %d bytes of data\n", input_size);
+ out = faacDecDecode (faad->handle, info, input_data, input_size);
+ g_print ("done, rec. %p\n", out);
+ if (info->error) {
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
- ("Failed to decode buffer: %s", faacDecGetErrorMessage (info.error)));
+ ("Failed to decode buffer: %s",
+ faacDecGetErrorMessage (info->error)));
break;
}
- input_size -= info.bytesconsumed;
- input_data += info.bytesconsumed;
+ if (info->bytesconsumed > input_size)
+ info->bytesconsumed = input_size;
+ input_size -= info->bytesconsumed;
+ input_data += info->bytesconsumed;
+
+ if (out && info->samples > 0) {
+ gboolean fmt_change = FALSE;
- if (out) {
+ /* see if we need to renegotiate */
+ if (info->samplerate != faad->samplerate ||
+ info->channels != faad->channels || !faad->channel_positions) {
+ fmt_change = TRUE;
+ } else {
+ gint i;
- if (info.samplerate != faad->samplerate
- || info.channels != faad->channels) {
+ for (i = 0; i < info->channels; i++) {
+ if (info->channel_position[i] != faad->channel_positions[i])
+ fmt_change = TRUE;
+ }
+ }
+
+ if (fmt_change) {
GstPadLinkReturn ret;
- faad->samplerate = info.samplerate;
- faad->channels = info.channels;
+ g_print ("Format change\n");
+ g_print ("To %ld Hz, %d chans, %d/%d/%d/%d/%d/%d\n",
+ info->samplerate, info->channels,
+ info->channel_position[0],
+ info->channel_position[1],
+ info->channel_position[2],
+ info->channel_position[3],
+ info->channel_position[4], info->channel_position[5]);
+ /* store new negotiation information */
+ faad->samplerate = info->samplerate;
+ faad->channels = info->channels;
+ if (faad->channel_positions)
+ g_free (faad->channel_positions);
+ faad->channel_positions = g_new (guint8, faad->channels);
+ memcpy (faad->channel_positions, info->channel_position,
+ faad->channels);
+
+ /* and negotiate */
ret = gst_pad_renegotiate (faad->srcpad);
if (GST_PAD_LINK_FAILED (ret)) {
GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL), (NULL));
}
}
- if (info.samples > 0) {
- outbuf = gst_buffer_new_and_alloc (info.samples * faad->bps);
+ /* play decoded data */
+ if (info->samples > 0) {
+ g_print ("Playing %ld samples from buf %p\n", info->samples, out);
+ outbuf = gst_buffer_new_and_alloc (info->samples * faad->bps);
/* ugh */
memcpy (GST_BUFFER_DATA (outbuf), out, GST_BUFFER_SIZE (outbuf));
+ g_print ("done, to %p\n", GST_BUFFER_DATA (outbuf));
GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (buf);
gst_pad_push (faad->srcpad, GST_DATA (outbuf));
}
}
- };
+ }
/* Keep the leftovers */
if (input_size > 0) {
- if (input_size < GST_BUFFER_SIZE (buf))
+ if (input_size < GST_BUFFER_SIZE (buf)) {
faad->tempbuf = gst_buffer_create_sub (buf,
GST_BUFFER_SIZE (buf) - input_size, input_size);
- else {
+ } else {
faad->tempbuf = buf;
gst_buffer_ref (buf);
}
}
gst_buffer_unref (buf);
+
+ g_free (info);
}
static GstElementStateReturn
case GST_STATE_PAUSED_TO_READY:
faad->samplerate = -1;
faad->channels = -1;
+ faad->need_channel_setup = TRUE;
+ faad->init = FALSE;
+ g_free (faad->channel_positions);
+ faad->channel_positions = NULL;
break;
case GST_STATE_READY_TO_NULL:
faacDecClose (faad->handle);
static gboolean
plugin_init (GstPlugin * plugin)
{
- return gst_element_register (plugin, "faad", GST_RANK_PRIMARY, GST_TYPE_FAAD);
+ return gst_library_load ("gstaudio") &&
+ gst_element_register (plugin, "faad", GST_RANK_PRIMARY, GST_TYPE_FAAD);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
/* FAAD object */
faacDecHandle handle;
+ gboolean init;
+
+ /* FAAD channel setup */
+ guchar *channel_positions;
+ gboolean need_channel_setup;
} GstFaad;
typedef struct _GstFaadClass {
+# variables used for enum/marshal generation
+glib_enum_headers=multichannel.h
+glib_enum_define=GST_AUDIO
+glib_enum_prefix=gst_audio
+
+built_sources = multichannel-enumtypes.c
+built_headers = multichannel-enumtypes.h
+BUILT_SOURCES = $(built_sources) $(built_headers)
+
librarydir = $(libdir)/gstreamer-@GST_MAJORMINOR@
library_LTLIBRARIES = libgstaudio.la libgstaudiofilter.la
noinst_LTLIBRARIES = libgstaudiofilterexample.la
EXTRA_DIST = gstaudiofiltertemplate.c make_filter
-CLEANFILES = gstaudiofilterexample.c
+CLEANFILES = gstaudiofilterexample.c \
+ $(BUILT_SOURCES)
-libgstaudio_la_SOURCES = audio.c audioclock.c
+libgstaudio_la_SOURCES = audio.c audioclock.c \
+ multichannel.c
+nodist_libgstaudio_la_SOURCES = $(built_sources)
libgstaudioincludedir = $(includedir)/gstreamer-@GST_MAJORMINOR@/gst/audio
-libgstaudioinclude_HEADERS = audio.h audioclock.h gstaudiofilter.h
+libgstaudioinclude_HEADERS = \
+ audio.h \
+ audioclock.h \
+ gstaudiofilter.h \
+ multichannel.h \
+ multichannel-enumtypes.h
libgstaudio_la_LIBADD =
libgstaudio_la_CFLAGS = $(GST_CFLAGS)
gstaudiofilterexample.c: $(srcdir)/make_filter $(srcdir)/gstaudiofiltertemplate.c
$(srcdir)/make_filter AudiofilterExample $(srcdir)/gstaudiofiltertemplate.c
+
+noinst_PROGRAMS = testchannels
+testchannels_SOURCES = testchannels.c
+testchannels_CFLAGS = $(GST_CFLAGS)
+testchannels_LDFLAGS = $(GST_LIBS)
+
+include $(top_srcdir)/common/glib-gen.mak
#endif
#include "audio.h"
+#include "multichannel-enumtypes.h"
#include <gst/gststructure.h>
static gboolean
plugin_init (GstPlugin * plugin)
{
+ gst_audio_channel_position_get_type ();
+
return TRUE;
}
--- /dev/null
+/* GStreamer Multichannel-Audio helper functions
+ * (c) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include "multichannel.h"
+
+#define GST_AUDIO_CHANNEL_POSITIONS_PROPERTY_NAME "channel-positions"
+
+/*
+ * This function checks if basic assumptions apply:
+ * - does each position occur at most once?
+ * - do conflicting positions occur?
+ * + front_mono vs. front_left/right
+ * + front_center vs. front_left/right_of_center
+ * + rear_center vs. rear_left/right
+ * It also adds some hacks that 0.8.x needs for compatibility:
+ * - if channels == 1, are we really mono?
+ * - if channels == 2, are we really stereo?
+ */
+
+static gboolean
+gst_audio_check_channel_positions (const GstAudioChannelPosition * pos,
+ gint channels)
+{
+ gint i, n;
+ struct
+ {
+ GstAudioChannelPosition pos1[2];
+ GstAudioChannelPosition pos2[1];
+ } conf[] = {
+ /* front: mono <-> stereo */
+ { {
+ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
+ GST_AUDIO_CHANNEL_POSITION_FRONT_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}}
+ };
+
+ /* check for invalid channel positions */
+ for (n = 0; n < channels; n++) {
+ if (pos[n] == GST_AUDIO_CHANNEL_POSITION_INVALID) {
+ g_warning ("Position %d is invalid, not allowed", n);
+ return FALSE;
+ }
+ }
+
+ /* check for multiple position occurrences */
+ for (i = GST_AUDIO_CHANNEL_POSITION_INVALID + 1;
+ i < GST_AUDIO_CHANNEL_POSITION_NUM; i++) {
+ gint count = 0;
+
+ for (n = 0; n < channels; n++) {
+ if (pos[n] == i)
+ count++;
+ }
+
+ if (count > 1) {
+ g_warning ("Channel position %d occurred %d times, not allowed",
+ i, count);
+ return FALSE;
+ }
+ }
+
+ /* check for position conflicts */
+ for (i = 0; conf[i].pos1[0] != GST_AUDIO_CHANNEL_POSITION_INVALID; i++) {
+ gboolean found1 = FALSE, found2 = FALSE;
+
+ for (n = 0; n < channels; n++) {
+ if (pos[n] == conf[i].pos1[0] || pos[n] == conf[i].pos1[1])
+ found1 = TRUE;
+ else if (pos[n] == conf[i].pos2[0])
+ found2 = TRUE;
+ }
+
+ if (found1 && found2) {
+ g_warning ("Found conflicting channel positions %d/%d and %d",
+ conf[i].pos1[0], conf[i].pos1[1], conf[i].pos2[0]);
+ return FALSE;
+ }
+ }
+
+ /* 0.8.x evilry */
+ if ((channels == 1 && pos[0] != GST_AUDIO_CHANNEL_POSITION_FRONT_MONO) ||
+ (channels == 2 && (pos[0] != GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT ||
+ pos[1] != GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT))) {
+ g_warning ("0.8.x: channels=1 implies mono; channels=2 implies stereo");
+ return FALSE;
+ }
+
+ return TRUE;
+}
+
+/**
+ * gst_audio_get_channel_positions:
+ * @str: A #GstStructure to retrieve channel positions from.
+ *
+ * Retrieves a number of (fixed!) audio channel positions from
+ * the provided #GstStructure and returns it as a newly allocated
+ * array. The caller should g_free () this array. The caller
+ * should also check that the members in this #GstStructure are
+ * indeed "fixed" before calling this function.
+ *
+ * Returns: a newly allocated array containing the channel
+ * positions as provided in the given #GstStructure. Returns
+ * NULL on error.
+ */
+
+GstAudioChannelPosition *
+gst_audio_get_channel_positions (GstStructure * str)
+{
+ GstAudioChannelPosition *pos;
+ gint channels, n;
+ const GValue *pos_val_arr, *pos_val_entry;
+ gboolean res;
+ GType t;
+
+ /* get number of channels, general type checkups */
+ g_return_val_if_fail (str != NULL, NULL);
+ res = gst_structure_get_int (str, "channels", &channels);
+ g_return_val_if_fail (res, NULL);
+ g_return_val_if_fail (channels > 0, NULL);
+ pos_val_arr = gst_structure_get_value (str,
+ GST_AUDIO_CHANNEL_POSITIONS_PROPERTY_NAME);
+
+ /* The following checks are here to retain compatibility for plugins not
+ * implementing this property. They expect that channels=1 implies mono
+ * and channels=2 implies stereo, so we follow that.
+ * This might be removed during 0.9.x. */
+ if (!pos_val_arr && (channels == 1 || channels == 2)) {
+ pos = g_new (GstAudioChannelPosition, channels);
+ if (channels == 1) {
+ pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
+ } else {
+ pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ }
+ return pos;
+ }
+ g_return_val_if_fail (pos_val_arr != NULL, NULL);
+ g_return_val_if_fail (gst_value_list_get_size (pos_val_arr) == channels,
+ NULL);
+ for (n = 0; n < channels; n++) {
+ t = G_VALUE_TYPE (gst_value_list_get_value (pos_val_arr, n));
+ g_return_val_if_fail (t == GST_TYPE_AUDIO_CHANNEL_POSITION, NULL);
+ }
+
+ /* ... and fill array */
+ pos = g_new (GstAudioChannelPosition, channels);
+ for (n = 0; n < channels; n++) {
+ pos_val_entry = gst_value_list_get_value (pos_val_arr, n);
+ pos[n] = g_value_get_enum (pos_val_entry);
+ }
+
+ if (!gst_audio_check_channel_positions (pos, channels)) {
+ g_free (pos);
+ return NULL;
+ }
+
+ return pos;
+}
+
+/**
+ * gst_audio_set_channel_positions:
+ * @str: A #GstStructure to retrieve channel positions from.
+ * @pos: an array of channel positions. The number of members
+ * in this array should be equal to the (fixed!) number
+ * of the "channels" property in the given #GstStructure.
+ *
+ * Adds a "channel-positions" property to the given #GstStructure,
+ * which will represent the channel positions as given in the
+ * provided #GstAudioChannelPosition array.
+ */
+
+void
+gst_audio_set_channel_positions (GstStructure * str,
+ const GstAudioChannelPosition * pos)
+{
+ GValue pos_val_arr = { 0 }, pos_val_entry = {
+ 0};
+ gint channels, n;
+ gboolean res;
+
+ /* get number of channels, checkups */
+ g_return_if_fail (str != NULL);
+ g_return_if_fail (pos != NULL);
+ res = gst_structure_get_int (str, "channels", &channels);
+ g_return_if_fail (res);
+ g_return_if_fail (channels > 0);
+ if (!gst_audio_check_channel_positions (pos, channels))
+ return;
+
+ /* build gvaluearray from positions */
+ g_value_init (&pos_val_entry, GST_TYPE_AUDIO_CHANNEL_POSITION);
+ g_value_init (&pos_val_arr, GST_TYPE_FIXED_LIST);
+ for (n = 0; n < channels; n++) {
+ g_value_set_enum (&pos_val_entry, pos[n]);
+ gst_value_list_append_value (&pos_val_arr, &pos_val_entry);
+ }
+ g_value_unset (&pos_val_entry);
+
+ /* add to structure */
+ gst_structure_set_value (str,
+ GST_AUDIO_CHANNEL_POSITIONS_PROPERTY_NAME, &pos_val_arr);
+ g_value_unset (&pos_val_arr);
+}
+
+/**
+ * gst_audio_set_structure_channel_positions_list:
+ * @str: #GstStructure to set the list of channel positions
+ * on.
+ * @pos: the array containing one or more possible audio
+ * channel positions that we should add in each value
+ * of the array in the given structure.
+ * @num_positions: the number of values in pos.
+ *
+ * Sets a (possibly non-fixed) list of possible audio channel
+ * positions (given in pos) on the given structure. The
+ * structure, after this function has been called, will contain
+ * a "channel-positions" property with an array of the size of
+ * the "channels" property value in the given structure (note
+ * that this means that the channels property in the provided
+ * structure should be fixed!). Each value in the array will
+ * contain each of the values given in the pos array.
+ */
+
+void
+gst_audio_set_structure_channel_positions_list (GstStructure * str,
+ const GstAudioChannelPosition * pos, gint num_positions)
+{
+ gint channels, n, c;
+ GValue pos_val_arr = { 0 }, pos_val_list = {
+ 0}, pos_val_entry = {
+ 0};
+ gboolean res;
+
+ /* get number of channels, general type checkups */
+ g_return_if_fail (str != NULL);
+ g_return_if_fail (num_positions > 0);
+ g_return_if_fail (pos != NULL);
+ res = gst_structure_get_int (str, "channels", &channels);
+ g_return_if_fail (res);
+ g_return_if_fail (channels > 0);
+
+ /* 0.8.x: channels=1 or channels=2 is mono/stereo, no positions needed
+ * there (we discard them anyway) */
+ if (channels == 1 || channels == 2)
+ return;
+
+ /* create the array of lists */
+ g_value_init (&pos_val_arr, GST_TYPE_FIXED_LIST);
+ g_value_init (&pos_val_entry, GST_TYPE_AUDIO_CHANNEL_POSITION);
+ for (n = 0; n < channels; n++) {
+ g_value_init (&pos_val_list, GST_TYPE_LIST);
+ for (c = 0; c < num_positions; c++) {
+ g_value_set_enum (&pos_val_entry, pos[c]);
+ gst_value_list_append_value (&pos_val_list, &pos_val_entry);
+ }
+ gst_value_list_append_value (&pos_val_arr, &pos_val_list);
+ g_value_unset (&pos_val_list);
+ }
+ g_value_unset (&pos_val_entry);
+ gst_structure_set_value (str, GST_AUDIO_CHANNEL_POSITIONS_PROPERTY_NAME,
+ &pos_val_arr);
+ g_value_unset (&pos_val_arr);
+}
+
+/*
+ * Helper function for below. The structure will be conserved,
+ * but might be cut down. Any additional structures that were
+ * created will be stored in the returned caps.
+ */
+
+static GstCaps *
+add_list_to_struct (GstStructure * str,
+ const GstAudioChannelPosition * pos, gint num_positions)
+{
+ GstCaps *caps = gst_caps_new_empty ();
+ const GValue *chan_val;
+
+ chan_val = gst_structure_get_value (str, "channels");
+ if (G_VALUE_TYPE (chan_val) == G_TYPE_INT) {
+ gst_audio_set_structure_channel_positions_list (str, pos, num_positions);
+ } else if (G_VALUE_TYPE (chan_val) == GST_TYPE_LIST) {
+ gint size;
+ const GValue *sub_val;
+
+ size = gst_value_list_get_size (chan_val);
+ sub_val = gst_value_list_get_value (chan_val, 0);
+ gst_structure_set_value (str, "channels", sub_val);
+ gst_caps_append (caps, add_list_to_struct (str, pos, num_positions));
+ while (--size > 0) {
+ str = gst_structure_copy (str);
+ sub_val = gst_value_list_get_value (chan_val, size);
+ gst_structure_set_value (str, "channels", sub_val);
+ gst_caps_append (caps, add_list_to_struct (str, pos, num_positions));
+ gst_caps_append_structure (caps, str);
+ }
+ } else if (G_VALUE_TYPE (chan_val) == GST_TYPE_INT_RANGE) {
+ gint min, max;
+
+ min = gst_value_get_int_range_min (chan_val);
+ max = gst_value_get_int_range_max (chan_val);
+
+ gst_structure_set (str, "channels", G_TYPE_INT, min, NULL);
+ gst_audio_set_structure_channel_positions_list (str, pos, num_positions);
+ for (++min; min < max; min++) {
+ str = gst_structure_copy (str);
+ gst_structure_set (str, "channels", G_TYPE_INT, min, NULL);
+ gst_audio_set_structure_channel_positions_list (str, pos, num_positions);
+ gst_caps_append_structure (caps, str);
+ }
+ } else {
+ g_warning ("Unknown value type for channels property");
+ }
+
+ return caps;
+}
+
+/**
+ * gst_audio_set_caps_channel_positions_list:
+ * @caps: #GstCaps to set the list of channel positions on.
+ * @pos: the array containing one or more possible audio
+ * channel positions that we should add in each value
+ * of the array in the given structure.
+ * @num_positions: the number of values in pos.
+ *
+ * Sets a (possibly non-fixed) list of possible audio channel
+ * positions (given in pos) on the given caps. Each of the
+ * structures of the caps, after this function has been called,
+ * will contain a "channel-positions" property with an array.
+ * Each value in the array will contain each of the values given
+ * in the pos array. Note that the size of the caps might be
+ * increased by this, since each structure with a "channel-
+ * positions" property needs to have a fixed "channels" property.
+ * The input caps is not required to have this.
+ */
+
+void
+gst_audio_set_caps_channel_positions_list (GstCaps * caps,
+ const GstAudioChannelPosition * pos, gint num_positions)
+{
+ gint size, n;
+
+ /* get number of channels, general type checkups */
+ g_return_if_fail (caps != NULL);
+ g_return_if_fail (num_positions > 0);
+ g_return_if_fail (pos != NULL);
+
+ size = gst_caps_get_size (caps);
+ for (n = 0; n < size; n++) {
+ gst_caps_append (caps, add_list_to_struct (gst_caps_get_structure (caps,
+ n), pos, num_positions));
+ }
+}
+
+/**
+ * gst_audio_fixate_channel_positions:
+ * @str: a #GstStructure containing a (possibly unfixed)
+ * "channel-positions" property.
+ *
+ * Custom fixate function. Elements that implement some sort of
+ * channel conversion algorhithm should use this function for
+ * fixating on GstAudioChannelPosition properties. It will take
+ * care of equal channel positioning (left/right). Caller g_free()s
+ * the return value. The input properties may be (and are supposed
+ * to be) unfixed.
+ * Note that this function is mostly a hack because we currently
+ * have no way to add default fixation functions for new GTypes.
+ *
+ * Returns: fixed values that the caller could use as a fixed
+ * set of #GstAudioChannelPosition values.
+ */
+
+GstAudioChannelPosition *
+gst_audio_fixate_channel_positions (GstStructure * str)
+{
+ GstAudioChannelPosition *pos;
+ gint channels, n, num_unfixed = 0, i, c;
+ const GValue *pos_val_arr, *pos_val_entry, *pos_val;
+ gboolean res, is_stereo = TRUE;
+ GType t;
+
+ /*
+ * We're going to do this cluelessly. We'll make an array of values that
+ * conflict with each other and, for each iteration in this array, pick
+ * either one until all unknown values are filled. This might not work in
+ * corner cases but should work OK for the general case.
+ */
+ struct
+ {
+ GstAudioChannelPosition pos1[2];
+ GstAudioChannelPosition pos2[1];
+ } conf[] = {
+ /* front: mono <-> stereo */
+ { {
+ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
+ GST_AUDIO_CHANNEL_POSITION_FRONT_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, GST_AUDIO_CHANNEL_POSITION_INVALID}, {
+ GST_AUDIO_CHANNEL_POSITION_LFE}}, { {
+ GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, {
+ GST_AUDIO_CHANNEL_POSITION_INVALID}}, { {
+ GST_AUDIO_CHANNEL_POSITION_INVALID, GST_AUDIO_CHANNEL_POSITION_INVALID}, {
+ GST_AUDIO_CHANNEL_POSITION_INVALID}}
+ };
+ struct
+ {
+ gint num_opt[3];
+ guint num_opts[3];
+ gboolean is_fixed[3];
+ gint choice; /* -1 is none, 0 is the two, 1 is the one */
+ } opt;
+
+ /* get number of channels, general type checkups */
+ g_return_val_if_fail (str != NULL, NULL);
+ res = gst_structure_get_int (str, "channels", &channels);
+ g_return_val_if_fail (res, NULL);
+ g_return_val_if_fail (channels > 0, NULL);
+
+ /* 0.8.x mono/stereo checks */
+ pos_val_arr = gst_structure_get_value (str,
+ GST_AUDIO_CHANNEL_POSITIONS_PROPERTY_NAME);
+ if (!pos_val_arr && (channels == 1 || channels == 2)) {
+ pos = g_new (GstAudioChannelPosition, channels);
+ if (channels == 1) {
+ pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
+ } else {
+ pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ }
+ return pos;
+ }
+ g_return_val_if_fail (pos_val_arr != NULL, NULL);
+ g_return_val_if_fail (gst_value_list_get_size (pos_val_arr) == channels,
+ NULL);
+ for (n = 0; n < channels; n++) {
+ t = G_VALUE_TYPE (gst_value_list_get_value (pos_val_arr, n));
+ g_return_val_if_fail (t == GST_TYPE_LIST ||
+ t == GST_TYPE_AUDIO_CHANNEL_POSITION, NULL);
+ }
+
+ /* all unknown, to start with */
+ pos = g_new (GstAudioChannelPosition, channels);
+ for (n = 0; n < channels; n++)
+ pos[n] = GST_AUDIO_CHANNEL_POSITION_INVALID;
+ num_unfixed = channels;
+
+ /* Iterate the array of conflicting values */
+ for (i = 0; conf[i].pos1[0] != GST_AUDIO_CHANNEL_POSITION_INVALID ||
+ conf[i].pos2[0] != GST_AUDIO_CHANNEL_POSITION_INVALID; i++) {
+ /* front/center only important if not mono (obviously) */
+ if (conf[i].pos1[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER &&
+ !is_stereo)
+ continue;
+
+ /* init values */
+ for (n = 0; n < 3; n++) {
+ opt.num_opt[n] = -1;
+ opt.num_opts[n] = -1;
+ opt.is_fixed[n] = FALSE;
+ }
+
+ /* Now, we'll see for each channel if it allows for any of the values in
+ * the set of conflicting audio channel positions and keep scores. */
+ for (n = 0; n < channels; n++) {
+ /* if the channel is already taken, don't bother */
+ if (pos[n] != GST_AUDIO_CHANNEL_POSITION_INVALID)
+ continue;
+
+ pos_val_entry = gst_value_list_get_value (pos_val_arr, n);
+ t = G_VALUE_TYPE (pos_val_entry);
+ if (t == GST_TYPE_LIST) {
+ /* This algorhythm is suboptimal. */
+ for (c = 0; c < gst_value_list_get_size (pos_val_entry); c++) {
+ pos_val = gst_value_list_get_value (pos_val_entry, c);
+ if (g_value_get_enum (pos_val) == conf[i].pos1[0] &&
+ opt.num_opts[0] > gst_value_list_get_size (pos_val_entry) &&
+ !opt.is_fixed[0]) {
+ /* Now test if the old position of num_opt[0] also allows for
+ * the other channel (which was skipped previously). If so,
+ * keep score. */
+ if (opt.num_opt[0] != -1) {
+ gint c1;
+
+ pos_val_entry = gst_value_list_get_value (pos_val_arr,
+ opt.num_opt[0]);
+ if (G_VALUE_TYPE (pos_val_entry) == GST_TYPE_LIST) {
+ for (c1 = 0; c1 < gst_value_list_get_size (pos_val_entry); c1++) {
+ pos_val = gst_value_list_get_value (pos_val_entry, c1);
+ if (g_value_get_enum (pos_val) == conf[i].pos1[1] &&
+ opt.num_opts[1] > opt.num_opts[0] && !opt.is_fixed[1]) {
+ opt.num_opts[1] = opt.num_opts[0];
+ opt.num_opt[1] = opt.num_opt[0];
+ }
+ }
+ pos_val = gst_value_list_get_value (pos_val_entry, c);
+ }
+ pos_val_entry = gst_value_list_get_value (pos_val_arr, n);
+ }
+
+ /* and save values */
+ opt.num_opts[0] = gst_value_list_get_size (pos_val_entry);
+ opt.num_opt[0] = n;
+ } else if (g_value_get_enum (pos_val) == conf[i].pos1[1] &&
+ opt.num_opts[1] > gst_value_list_get_size (pos_val_entry) &&
+ !opt.is_fixed[1] && n != opt.num_opt[0]) {
+ opt.num_opts[1] = gst_value_list_get_size (pos_val_entry);
+ opt.num_opt[1] = n;
+ }
+
+ /* 2 goes separately, because 0/1 vs. 2 are separate */
+ if (g_value_get_enum (pos_val) == conf[i].pos2[0] &&
+ opt.num_opts[2] > gst_value_list_get_size (pos_val_entry) &&
+ !opt.is_fixed[2]) {
+ opt.num_opts[2] = gst_value_list_get_size (pos_val_entry);
+ opt.num_opt[2] = n;
+ }
+ }
+ } else {
+ if (g_value_get_enum (pos_val_entry) == conf[i].pos1[0]) {
+ opt.num_opt[0] = n;
+ opt.is_fixed[0] = TRUE;
+ } else if (g_value_get_enum (pos_val_entry) == conf[i].pos1[1]) {
+ opt.num_opt[1] = n;
+ opt.is_fixed[1] = TRUE;
+ } else if (g_value_get_enum (pos_val_entry) == conf[i].pos2[0]) {
+ opt.num_opt[2] = n;
+ opt.is_fixed[2] = TRUE;
+ }
+ }
+ }
+
+ /* check our results and choose either one */
+ if ((opt.is_fixed[0] || opt.is_fixed[1]) && opt.is_fixed[2]) {
+ g_warning ("Pre-fixated on both %d/%d and %d - conflict!",
+ conf[i].pos1[0], conf[i].pos1[1], conf[i].pos2[0]);
+ g_free (pos);
+ return NULL;
+ } else if ((opt.is_fixed[0] && opt.num_opt[1] == -1) ||
+ (opt.is_fixed[1] && opt.num_opt[0] == -1)) {
+ g_warning ("Pre-fixated one side, but other side n/a of %d/%d",
+ conf[i].pos1[0], conf[i].pos1[1]);
+ g_free (pos);
+ return NULL;
+ } else if (opt.is_fixed[0] || opt.is_fixed[1]) {
+ opt.choice = 0;
+ } else if (opt.is_fixed[2]) {
+ opt.choice = 1;
+ } else if (opt.num_opt[0] != -1 && opt.num_opt[1] != -1) {
+ opt.choice = 0;
+ } else if (opt.num_opt[2] != -1) {
+ opt.choice = 1;
+ } else {
+ opt.choice = -1;
+ }
+
+ /* stereo? Note that we keep is_stereo to TRUE if we didn't decide on
+ * any arrangement. The mono/stereo channels might be handled elsewhere
+ * which is clearly outside the scope of this element, so we cannot
+ * know and expect the application to handle that then. */
+ if (conf[i].pos2[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_MONO &&
+ opt.choice == 1) {
+ is_stereo = FALSE;
+ }
+
+ /* now actually decide what we'll do and fixate on that */
+ if (opt.choice == 0) {
+ g_assert (conf[i].pos1[0] != GST_AUDIO_CHANNEL_POSITION_INVALID &&
+ conf[i].pos1[1] != GST_AUDIO_CHANNEL_POSITION_INVALID);
+ pos[opt.num_opt[0]] = conf[i].pos1[0];
+ pos[opt.num_opt[1]] = conf[i].pos1[1];
+ num_unfixed -= 2;
+ } else if (opt.choice == 1) {
+ g_assert (conf[i].pos2[0] != GST_AUDIO_CHANNEL_POSITION_INVALID);
+ pos[opt.num_opt[2]] = conf[i].pos2[0];
+ num_unfixed--;
+ }
+ }
+
+ /* safety check */
+ if (num_unfixed > 0) {
+ g_warning ("%d unfixed channel positions left after fixation!",
+ num_unfixed);
+ g_free (pos);
+ return NULL;
+ }
+
+ if (!gst_audio_check_channel_positions (pos, channels)) {
+ g_free (pos);
+ return NULL;
+ }
+
+ return pos;
+}
--- /dev/null
+/* GStreamer Multichannel-Audio helper functions
+ * (c) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __GST_AUDIO_MULTICHANNEL_H__
+#define __GST_AUDIO_MULTICHANNEL_H__
+
+#include <gst/audio/audio.h>
+#include <gst/audio/multichannel-enumtypes.h>
+
+typedef enum {
+ GST_AUDIO_CHANNEL_POSITION_INVALID = -1,
+
+ /* Main front speakers. Mono and left/right are mututally exclusive! */
+ GST_AUDIO_CHANNEL_POSITION_FRONT_MONO,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+
+ /* rear. Left/right and center are mututally exclusive! */
+ GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
+ GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
+
+ /* subwoofer/low-frequency */
+ GST_AUDIO_CHANNEL_POSITION_LFE,
+
+ /* Center front speakers. Center and left/right_of_center cannot be
+ * used together! */
+ GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER,
+
+ /* sides */
+ GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
+
+ /* don't use - counter */
+ GST_AUDIO_CHANNEL_POSITION_NUM
+} GstAudioChannelPosition;
+
+/* Retrieves or sets the positions from/to a GstStructure. Only
+ * works with fixed caps, caller should check for that! Caller
+ * g_free()s result of the getter. */
+GstAudioChannelPosition *
+ gst_audio_get_channel_positions (GstStructure *str);
+void gst_audio_set_channel_positions (GstStructure *str,
+ const GstAudioChannelPosition *pos);
+
+/* Sets a (non-fixed) list of possible audio channel positions
+ * on a structure (this requires the "channels" property to
+ * be fixed!) or on a caps (here, the "channels" property may be
+ * unfixed and the caps may even contain multiple structures). */
+void gst_audio_set_structure_channel_positions_list
+ (GstStructure *str,
+ const GstAudioChannelPosition *pos,
+ gint num_positions);
+void gst_audio_set_caps_channel_positions_list
+ (GstCaps *caps,
+ const GstAudioChannelPosition *pos,
+ gint num_positions);
+
+/* Custom fixate function. Elements that implement some sort of
+ * channel conversion algorhithm should use this function for
+ * fixating on GstAudioChannelPosition properties. It will take
+ * care of equal channel positioning (left/right). Caller g_free()s
+ * the return value. The input properties may be (and are supposed
+ * to be) unfixed. */
+GstAudioChannelPosition *
+ gst_audio_fixate_channel_positions (GstStructure *str);
+
+#endif /* __GST_AUDIO_MULTICHANNEL_H__ */
--- /dev/null
+/* GStreamer Multichannel Test
+ * (c) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gst/gst.h>
+
+#include <multichannel.c>
+#include <multichannel-enumtypes.c>
+
+gint
+main (gint argc, gchar * argv[])
+{
+ gchar *str;
+ GstCaps *caps;
+ GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
+ };
+
+ /* register multichannel type */
+ gst_init (&argc, &argv);
+ gst_audio_channel_position_get_type ();
+
+ /* test some caps-string conversions */
+ caps = gst_caps_new_simple ("audio/x-raw-int",
+ "channels", G_TYPE_INT, 2, NULL);
+ str = gst_caps_to_string (caps);
+ g_print ("Test caps #1: %s\n", str);
+ g_free (str);
+ gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
+ str = gst_caps_to_string (caps);
+ g_print ("Test caps #2: %s\n", str);
+ g_free (str);
+ gst_caps_free (caps);
+
+ return 0;
+}
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