--- a/gst_plugins_base/gst/audioconvert/gstaudioconvert.c Wed Mar 24 17:58:42 2010 -0500
+++ b/gst_plugins_base/gst/audioconvert/gstaudioconvert.c Wed Mar 24 18:04:17 2010 -0500
@@ -24,31 +24,20 @@
/**
* SECTION:element-audioconvert
*
- * <refsect2>
- * <para>
* Audioconvert converts raw audio buffers between various possible formats.
* It supports integer to float conversion, width/depth conversion,
- * signedness and endianness conversion.
- * </para>
- * <para>
- * Some format conversion are not carried out in an optimal way right now.
- * E.g. converting from double to float would cause a loss of precision.
- * </para>
+ * signedness and endianness conversion and channel transformations.
+ *
+ * <refsect2>
* <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * |[
* gst-launch -v -m audiotestsrc ! audioconvert ! audio/x-raw-int,channels=2,width=8,depth=8 ! level ! fakesink silent=TRUE
- * </programlisting>
- * This pipeline converts audio to 8-bit. The level element shows that
+ * ]| This pipeline converts audio to 8-bit. The level element shows that
* the output levels still match the one for a sine wave.
- * </para>
- * <para>
- * <programlisting>
+ * |[
* gst-launch -v -m audiotestsrc ! audioconvert ! vorbisenc ! fakesink silent=TRUE
- * </programlisting>
- * The vorbis encoder takes float audio data instead of the integer data
+ * ]| The vorbis encoder takes float audio data instead of the integer data
* generated by audiotestsrc.
- * </para>
* </refsect2>
*
* Last reviewed on 2006-03-02 (0.10.4)
@@ -107,7 +96,8 @@
const GValue * value, GParamSpec * pspec);
static void gst_audio_convert_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
-
+static gboolean structure_has_fixed_channel_positions (GstStructure * s,
+ gboolean * unpositioned_layout);
/* AudioConvert signals and args */
enum
@@ -135,37 +125,37 @@
GST_STATIC_CAPS ( \
"audio/x-raw-float, " \
"rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, 8 ], " \
+ "channels = (int) [ 1, MAX ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 64;" \
"audio/x-raw-float, " \
"rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, 8 ], " \
+ "channels = (int) [ 1, MAX ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 32;" \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, 8 ], " \
+ "channels = (int) [ 1, MAX ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 32, " \
"depth = (int) [ 1, 32 ], " \
"signed = (boolean) { true, false }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, 8 ], " \
+ "channels = (int) [ 1, MAX ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 24, " \
"depth = (int) [ 1, 24 ], " "signed = (boolean) { true, false }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, 8 ], " \
+ "channels = (int) [ 1, MAX ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 16, " \
"depth = (int) [ 1, 16 ], " \
"signed = (boolean) { true, false }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, 8 ], " \
+ "channels = (int) [ 1, MAX ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 8, " \
"depth = (int) [ 1, 8 ], " \
@@ -263,13 +253,14 @@
g_object_class_install_property (gobject_class, ARG_DITHERING,
g_param_spec_enum ("dithering", "Dithering",
"Selects between different dithering methods.",
- GST_TYPE_AUDIO_CONVERT_DITHERING, DITHER_TPDF, G_PARAM_READWRITE));
+ GST_TYPE_AUDIO_CONVERT_DITHERING, DITHER_TPDF,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, ARG_NOISE_SHAPING,
g_param_spec_enum ("noise-shaping", "Noise shaping",
"Selects between different noise shaping methods.",
GST_TYPE_AUDIO_CONVERT_NOISE_SHAPING, NOISE_SHAPING_NONE,
- G_PARAM_READWRITE));
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
basetransform_class->get_unit_size =
GST_DEBUG_FUNCPTR (gst_audio_convert_get_unit_size);
@@ -293,6 +284,8 @@
this->dither = DITHER_TPDF;
this->ns = NOISE_SHAPING_NONE;
memset (&this->ctx, 0, sizeof (AudioConvertCtx));
+
+ gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (this), TRUE);
}
static void
@@ -330,6 +323,10 @@
goto no_values;
if (!(fmt->pos = gst_audio_get_channel_positions (structure)))
goto no_values;
+
+ fmt->unpositioned_layout = FALSE;
+ structure_has_fixed_channel_positions (structure, &fmt->unpositioned_layout);
+
if (!gst_structure_get_int (structure, "width", &fmt->width))
goto no_values;
if (!gst_structure_get_int (structure, "rate", &fmt->rate))
@@ -529,6 +526,40 @@
}
}
+static gboolean
+structure_has_fixed_channel_positions (GstStructure * s,
+ gboolean * unpositioned_layout)
+{
+ GstAudioChannelPosition *pos;
+ const GValue *val;
+ gint channels = 0;
+
+ if (!gst_structure_get_int (s, "channels", &channels))
+ return FALSE; /* probably a range */
+
+ val = gst_structure_get_value (s, "channel-positions");
+ if ((val == NULL || !gst_value_is_fixed (val)) && channels <= 8) {
+ GST_LOG ("no or unfixed channel-positions in %" GST_PTR_FORMAT, s);
+ return FALSE;
+ } else if (val == NULL || !gst_value_is_fixed (val)) {
+ GST_LOG ("implicit undefined channel-positions");
+ *unpositioned_layout = TRUE;
+ return TRUE;
+ }
+
+ pos = gst_audio_get_channel_positions (s);
+ if (pos && pos[0] == GST_AUDIO_CHANNEL_POSITION_NONE) {
+ GST_LOG ("fixed undefined channel-positions in %" GST_PTR_FORMAT, s);
+ *unpositioned_layout = TRUE;
+ } else {
+ GST_LOG ("fixed defined channel-positions in %" GST_PTR_FORMAT, s);
+ *unpositioned_layout = FALSE;
+ }
+ g_free (pos);
+
+ return TRUE;
+}
+
/* Audioconvert can perform all conversions on audio except for resampling.
* However, there are some conversions we _prefer_ not to do. For example, it's
* better to convert format (float<->int, endianness, etc) than the number of
@@ -546,8 +577,8 @@
{
GstCaps *ret;
GstStructure *s, *structure;
- gboolean isfloat;
- gint width, depth, channels;
+ gboolean isfloat, allow_mixing;
+ gint width, depth, channels = 0;
const gchar *fields_used[] = {
"width", "depth", "rate", "channels", "endianness", "signed"
};
@@ -603,11 +634,28 @@
}
}
+ allow_mixing = TRUE;
if (gst_structure_get_int (structure, "channels", &channels)) {
- if (channels == 8)
- gst_structure_set (s, "channels", G_TYPE_INT, 8, NULL);
+ gboolean unpositioned;
+
+ /* we don't support mixing for channels without channel positions */
+ if (structure_has_fixed_channel_positions (structure, &unpositioned))
+ allow_mixing = (unpositioned == FALSE);
+ }
+
+ if (!allow_mixing) {
+ gst_structure_set (s, "channels", G_TYPE_INT, channels, NULL);
+ if (gst_structure_has_field (structure, "channel-positions"))
+ gst_structure_set_value (s, "channel-positions",
+ gst_structure_get_value (structure, "channel-positions"));
+ } else {
+ if (channels == 0)
+ gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, 11, NULL);
+ else if (channels == 11)
+ gst_structure_set (s, "channels", G_TYPE_INT, 11, NULL);
else
- gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, channels, 8, NULL);
+ gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, channels, 11, NULL);
+ gst_structure_remove_field (s, "channel-positions");
}
gst_caps_append_structure (ret, s);
@@ -636,7 +684,16 @@
* it's very bad to drop channels entirely.
*/
s = gst_structure_copy (s);
- gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, 8, NULL);
+ if (allow_mixing) {
+ gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, 11, NULL);
+ gst_structure_remove_field (s, "channel-positions");
+ } else {
+ /* allow_mixing can only be FALSE if we got a fixed number of channels */
+ gst_structure_set (s, "channels", G_TYPE_INT, channels, NULL);
+ if (gst_structure_has_field (structure, "channel-positions"))
+ gst_structure_set_value (s, "channel-positions",
+ gst_structure_get_value (structure, "channel-positions"));
+ }
gst_caps_append_structure (ret, s);
/* Same, plus a float<->int conversion */
@@ -751,7 +808,7 @@
gst_audio_convert_fixate_channels (GstBaseTransform * base, GstStructure * ins,
GstStructure * outs)
{
- const GValue *out_layout;
+ const GValue *in_layout, *out_layout;
gint in_chans, out_chans;
if (!gst_structure_get_int (ins, "channels", &in_chans))
@@ -776,19 +833,18 @@
/* check if the output has a channel layout (or a list of layouts) */
out_layout = gst_structure_get_value (outs, "channel-positions");
+ /* get the channel layout of the input if any */
+ in_layout = gst_structure_get_value (ins, "channel-positions");
+
if (out_layout == NULL) {
- if (out_chans <= 2)
+ if (out_chans <= 2 && (in_chans != out_chans || in_layout == NULL))
return; /* nothing to do, default layout will be assumed */
GST_WARNING_OBJECT (base, "downstream caps contain no channel layout");
}
- if (in_chans == out_chans) {
- const GValue *in_layout;
+ if (in_chans == out_chans && in_layout != NULL) {
GValue res = { 0, };
- in_layout = gst_structure_get_value (ins, "channel-positions");
- g_return_if_fail (in_layout != NULL);
-
/* same number of channels and no output layout: just use input layout */
if (out_layout == NULL) {
gst_structure_set_value (outs, "channel-positions", in_layout);
@@ -846,7 +902,7 @@
* and try to add/remove channels from the input layout, or pick a default
* layout based on LFE-presence in input layout, but let's save that for
* another day) */
- if (out_chans > 0 && out_chans < G_N_ELEMENTS (default_positions[0])) {
+ if (out_chans > 0 && out_chans <= G_N_ELEMENTS (default_positions[0])) {
GST_DEBUG_OBJECT (base, "using default channel layout as fallback");
gst_audio_set_channel_positions (outs, default_positions[out_chans - 1]);
}
@@ -946,6 +1002,79 @@
return GST_FLOW_OK;
}
+static void
+gst_audio_convert_create_silence_buffer (GstAudioConvert * this, gpointer dst,
+ gint size)
+{
+ if (this->ctx.out.is_int && !this->ctx.out.sign) {
+ gint i;
+
+ switch (this->ctx.out.width) {
+ case 8:{
+ guint8 zero = 0x80 >> (8 - this->ctx.out.depth);
+
+ memset (dst, zero, size);
+ break;
+ }
+ case 16:{
+ guint16 *data = (guint16 *) dst;
+ guint16 zero = 0x8000 >> (16 - this->ctx.out.depth);
+
+ if (this->ctx.out.endianness == G_LITTLE_ENDIAN)
+ zero = GUINT16_TO_LE (zero);
+ else
+ zero = GUINT16_TO_BE (zero);
+
+ size /= 2;
+
+ for (i = 0; i < size; i++)
+ data[i] = zero;
+ break;
+ }
+ case 24:{
+ guint32 zero = 0x800000 >> (24 - this->ctx.out.depth);
+ guint8 *data = (guint8 *) dst;
+
+ if (this->ctx.out.endianness == G_LITTLE_ENDIAN) {
+ for (i = 0; i < size; i += 3) {
+ data[i] = zero & 0xff;
+ data[i + 1] = (zero >> 8) & 0xff;
+ data[i + 2] = (zero >> 16) & 0xff;
+ }
+ } else {
+ for (i = 0; i < size; i += 3) {
+ data[i + 2] = zero & 0xff;
+ data[i + 1] = (zero >> 8) & 0xff;
+ data[i] = (zero >> 16) & 0xff;
+ }
+ }
+ break;
+ }
+ case 32:{
+ guint32 *data = (guint32 *) dst;
+ guint32 zero = (0x80000000 >> (32 - this->ctx.out.depth));
+
+ if (this->ctx.out.endianness == G_LITTLE_ENDIAN)
+ zero = GUINT32_TO_LE (zero);
+ else
+ zero = GUINT32_TO_BE (zero);
+
+ size /= 4;
+
+ for (i = 0; i < size; i++)
+ data[i] = zero;
+ break;
+ }
+ default:
+ memset (dst, 0, size);
+ g_return_if_reached ();
+ break;
+ }
+ } else {
+ memset (dst, 0, size);
+ }
+}
+
static GstFlowReturn
gst_audio_convert_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
@@ -978,9 +1107,14 @@
dst = GST_BUFFER_DATA (outbuf);
/* and convert the samples */
- if (!(res = audio_convert_convert (&this->ctx, src, dst,
- samples, gst_buffer_is_writable (inbuf))))
- goto convert_error;
+ if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
+ if (!(res = audio_convert_convert (&this->ctx, src, dst,
+ samples, gst_buffer_is_writable (inbuf))))
+ goto convert_error;
+ } else {
+ /* Create silence buffer */
+ gst_audio_convert_create_silence_buffer (this, dst, outsize);
+ }
GST_BUFFER_SIZE (outbuf) = outsize;