FCL/sf/mw/gstreamer: comparison gst_plugins_base/gst/audioconvert/gstaudioconvert.c

equal deleted inserted replaced

-:567bb019e3e3
+:7e817e7e631c
 */
 /**
 * 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 and channel transformations.
+* signedness and endianness conversion.
-*
+* </para>
-* <refsect2>
+* <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>
 * <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
-* ]| This pipeline converts audio to 8-bit.  The level element shows that
+* </programlisting>
+* 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
-* ]| The vorbis encoder takes float audio data instead of the integer data
+* </programlisting>
+* 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)
 */
 GstBuffer * buf);
 static void gst_audio_convert_set_property (GObject * object, guint prop_id,
 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
 {
 /* FILL ME */
 #define STATIC_CAPS \
 GST_STATIC_CAPS ( \
 "audio/x-raw-float, " \
 "rate = (int) [ 1, MAX ], " \
-"channels = (int) [ 1, MAX ], " \
+"channels = (int) [ 1, 8 ], " \
 "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
 "width = (int) 64;" \
 "audio/x-raw-float, " \
 "rate = (int) [ 1, MAX ], " \
-"channels = (int) [ 1, MAX ], " \
+"channels = (int) [ 1, 8 ], " \
 "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
 "width = (int) 32;" \
 "audio/x-raw-int, " \
 "rate = (int) [ 1, MAX ], " \
-"channels = (int) [ 1, MAX ], " \
+"channels = (int) [ 1, 8 ], " \
 "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, MAX ], "       \
+"channels = (int) [ 1, 8 ], "       \
 "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, MAX ], " \
+"channels = (int) [ 1, 8 ], " \
 "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, MAX ], " \
+"channels = (int) [ 1, 8 ], " \
 "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
 "width = (int) 8, " \
 "depth = (int) [ 1, 8 ], " \
 "signed = (boolean) { true, false } " \
 )
 supported_positions[i] = i;
 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,
+GST_TYPE_AUDIO_CONVERT_DITHERING, DITHER_TPDF, G_PARAM_READWRITE));
-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_STATIC_STRINGS));
+G_PARAM_READWRITE));
 basetransform_class->get_unit_size =
 GST_DEBUG_FUNCPTR (gst_audio_convert_get_unit_size);
 basetransform_class->transform_caps =
 GST_DEBUG_FUNCPTR (gst_audio_convert_transform_caps);
 gst_audio_convert_init (GstAudioConvert * this, GstAudioConvertClass * g_class)
 {
 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
 gst_audio_convert_dispose (GObject * obj)
 {
 /* parse common fields */
 if (!gst_structure_get_int (structure, "channels", &fmt->channels))
 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))
 goto no_values;
 /* width != 8 needs an endianness field */
 s2 = gst_structure_copy (s);
 gst_structure_set_name (s2, "audio/x-raw-float");
 s = make_lossless_changes (s2, TRUE);
 gst_caps_append_structure (caps, s2);
 }
-}
-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
 gst_audio_convert_transform_caps (GstBaseTransform * base,
 GstPadDirection direction, GstCaps * caps)
 {
 GstCaps *ret;
 GstStructure *s, *structure;
-gboolean isfloat, allow_mixing;
+gboolean isfloat;
-gint width, depth, channels = 0;
+gint width, depth, channels;
 const gchar *fields_used[] = {
 "width", "depth", "rate", "channels", "endianness", "signed"
 };
 const gchar *structure_name;
 int i;
 else
 gst_structure_set (s, "depth", GST_TYPE_INT_RANGE, depth, 32, NULL);
 }
 }
-allow_mixing = TRUE;
 if (gst_structure_get_int (structure, "channels", &channels)) {
-gboolean unpositioned;
+if (channels == 8)
+gst_structure_set (s, "channels", G_TYPE_INT, 8, NULL);
-/* 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, 11, NULL);
+gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, channels, 8, NULL);
-gst_structure_remove_field (s, "channel-positions");
 }
 gst_caps_append_structure (ret, s);
 /* Same, plus a float<->int conversion */
 append_with_other_format (ret, s, isfloat);
 /* Channel conversions to fewer channels is only done if needed - generally
 * it's very bad to drop channels entirely.
 */
 s = gst_structure_copy (s);
-if (allow_mixing) {
+gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, 8, NULL);
-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 */
 append_with_other_format (ret, s, isfloat);
 static void
 gst_audio_convert_fixate_channels (GstBaseTransform * base, GstStructure * ins,
 GstStructure * outs)
 {
-const GValue *in_layout, *out_layout;
+const GValue *out_layout;
 gint in_chans, out_chans;
 if (!gst_structure_get_int (ins, "channels", &in_chans))
 return;                     /* this shouldn't really happen, should it? */
 }
 /* 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 && (in_chans != out_chans || in_layout == NULL))
+if (out_chans <= 2)
 return;                   /* nothing to do, default layout will be assumed */
 GST_WARNING_OBJECT (base, "downstream caps contain no channel layout");
 }
-if (in_chans == out_chans && in_layout != NULL) {
+if (in_chans == out_chans) {
+const GValue *in_layout;
 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);
 return;
 /* missing or invalid output layout and we can't use the input layout for
 * one reason or another, so just pick a default layout (we could be smarter
 * 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]);
 }
 }
 static GstFlowReturn
 gst_audio_convert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
 {
 /* nothing to do here */
 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)
 /* get src and dst data */
 src = GST_BUFFER_DATA (inbuf);
 dst = GST_BUFFER_DATA (outbuf);
 /* and convert the samples */
-if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
+if (!(res = audio_convert_convert (&this->ctx, src, dst,
-if (!(res = audio_convert_convert (&this->ctx, src, dst,
+samples, gst_buffer_is_writable (inbuf))))
-samples, gst_buffer_is_writable (inbuf))))
+goto convert_error;
-goto convert_error;
-} else {
-/* Create silence buffer */
-gst_audio_convert_create_silence_buffer (this, dst, outsize);
-}
 GST_BUFFER_SIZE (outbuf) = outsize;
 return GST_FLOW_OK;

branch	RCL_3
changeset 30	7e817e7e631c
parent 29	567bb019e3e3