* <refsect2>
* <title>Example launch line</title>
* |[
- * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! audio/x-raw-int, rate=8000 ! alsasink
+ * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! audio/x-raw, rate=8000 ! alsasink
* ]| Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa.
* To create the Ogg/Vorbis file refer to the documentation of vorbisenc.
* </refsect2>
enum
{
PROP_0,
- PROP_QUALITY,
- PROP_FILTER_LENGTH
+ PROP_QUALITY
};
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define SUPPORTED_CAPS \
-GST_STATIC_CAPS ( \
- "audio/x-raw-float, " \
- "rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, MAX ], " \
- "endianness = (int) BYTE_ORDER, " \
- "width = (int) { 32, 64 }; " \
- "audio/x-raw-int, " \
- "rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, MAX ], " \
- "endianness = (int) BYTE_ORDER, " \
- "width = (int) 32, " \
- "depth = (int) 32, " \
- "signed = (boolean) true; " \
- "audio/x-raw-int, " \
- "rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, MAX ], " \
- "endianness = (int) BYTE_ORDER, " \
- "width = (int) 24, " \
- "depth = (int) 24, " \
- "signed = (boolean) true; " \
- "audio/x-raw-int, " \
- "rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, MAX ], " \
- "endianness = (int) BYTE_ORDER, " \
- "width = (int) 16, " \
- "depth = (int) 16, " \
- "signed = (boolean) true; " \
- "audio/x-raw-int, " \
- "rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, MAX ], " \
- "endianness = (int) BYTE_ORDER, " \
- "width = (int) 8, " \
- "depth = (int) 8, " \
- "signed = (boolean) true" \
-)
+ GST_AUDIO_CAPS_MAKE ("{ F32LE, F64LE, S32LE, S24LE, S16LE, S8 }")
+#else
+#define SUPPORTED_CAPS \
+ GST_AUDIO_CAPS_MAKE ("{ F32BE, F64BE, S32BE, S24BE, S16BE, S8 }")
+#endif
/* If TRUE integer arithmetic resampling is faster and will be used if appropiate */
#if defined AUDIORESAMPLE_FORMAT_INT
static GstStaticPadTemplate gst_audio_resample_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (SUPPORTED_CAPS));
static GstStaticPadTemplate gst_audio_resample_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (SUPPORTED_CAPS));
static void gst_audio_resample_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
/* vmethods */
static gboolean gst_audio_resample_get_unit_size (GstBaseTransform * base,
- GstCaps * caps, guint * size);
+ GstCaps * caps, gsize * size);
static GstCaps *gst_audio_resample_transform_caps (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps);
+ GstPadDirection direction, GstCaps * caps, GstCaps * filter);
static void gst_audio_resample_fixate_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * othercaps);
static gboolean gst_audio_resample_transform_size (GstBaseTransform * trans,
- GstPadDirection direction, GstCaps * incaps, guint insize,
- GstCaps * outcaps, guint * outsize);
+ GstPadDirection direction, GstCaps * incaps, gsize insize,
+ GstCaps * outcaps, gsize * outsize);
static gboolean gst_audio_resample_set_caps (GstBaseTransform * base,
GstCaps * incaps, GstCaps * outcaps);
static GstFlowReturn gst_audio_resample_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
-static gboolean gst_audio_resample_event (GstBaseTransform * base,
+static gboolean gst_audio_resample_sink_event (GstBaseTransform * base,
GstEvent * event);
static gboolean gst_audio_resample_start (GstBaseTransform * base);
static gboolean gst_audio_resample_stop (GstBaseTransform * base);
-static gboolean gst_audio_resample_query (GstPad * pad, GstQuery * query);
-static const GstQueryType *gst_audio_resample_query_type (GstPad * pad);
-
-GST_BOILERPLATE (GstAudioResample, gst_audio_resample, GstBaseTransform,
- GST_TYPE_BASE_TRANSFORM);
-
-static void
-gst_audio_resample_base_init (gpointer g_class)
-{
- GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
-
- gst_element_class_add_static_pad_template (gstelement_class,
- &gst_audio_resample_src_template);
- gst_element_class_add_static_pad_template (gstelement_class,
- &gst_audio_resample_sink_template);
+static gboolean gst_audio_resample_query (GstPad * pad, GstObject * parent,
+ GstQuery * query);
- gst_element_class_set_details_simple (gstelement_class, "Audio resampler",
- "Filter/Converter/Audio", "Resamples audio",
- "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
-}
+#define gst_audio_resample_parent_class parent_class
+G_DEFINE_TYPE (GstAudioResample, gst_audio_resample, GST_TYPE_BASE_TRANSFORM);
static void
gst_audio_resample_class_init (GstAudioResampleClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstElementClass *gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_audio_resample_set_property;
gobject_class->get_property = gst_audio_resample_get_property;
SPEEX_RESAMPLER_QUALITY_DEFAULT,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
- /* FIXME 0.11: Remove this property, it's just for compatibility
- * with old audioresample
- */
- /**
- * GstAudioResample:filter-length:
- *
- * Length of the resample filter
- *
- * Deprectated: Use #GstAudioResample:quality property instead
- */
- g_object_class_install_property (gobject_class, PROP_FILTER_LENGTH,
- g_param_spec_int ("filter-length", "Filter length",
- "Length of the resample filter", 0, G_MAXINT, 64,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_audio_resample_src_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_audio_resample_sink_template));
+
+ gst_element_class_set_details_simple (gstelement_class, "Audio resampler",
+ "Filter/Converter/Audio", "Resamples audio",
+ "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
GST_BASE_TRANSFORM_CLASS (klass)->start =
GST_DEBUG_FUNCPTR (gst_audio_resample_start);
GST_DEBUG_FUNCPTR (gst_audio_resample_set_caps);
GST_BASE_TRANSFORM_CLASS (klass)->transform =
GST_DEBUG_FUNCPTR (gst_audio_resample_transform);
- GST_BASE_TRANSFORM_CLASS (klass)->event =
- GST_DEBUG_FUNCPTR (gst_audio_resample_event);
+ GST_BASE_TRANSFORM_CLASS (klass)->sink_event =
+ GST_DEBUG_FUNCPTR (gst_audio_resample_sink_event);
GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE;
}
static void
-gst_audio_resample_init (GstAudioResample * resample,
- GstAudioResampleClass * klass)
+gst_audio_resample_init (GstAudioResample * resample)
{
GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
gst_base_transform_set_gap_aware (trans, TRUE);
gst_pad_set_query_function (trans->srcpad, gst_audio_resample_query);
- gst_pad_set_query_type_function (trans->srcpad,
- gst_audio_resample_query_type);
}
/* vmethods */
resample->tmp_out = NULL;
resample->tmp_out_size = 0;
- gst_caps_replace (&resample->sinkcaps, NULL);
- gst_caps_replace (&resample->srccaps, NULL);
-
return TRUE;
}
static gboolean
gst_audio_resample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
- guint * size)
+ gsize * size)
{
- gint width, channels;
- GstStructure *structure;
- gboolean ret;
+ GstAudioInfo info;
- g_return_val_if_fail (size != NULL, FALSE);
+ if (!gst_audio_info_from_caps (&info, caps))
+ goto invalid_caps;
- /* this works for both float and int */
- structure = gst_caps_get_structure (caps, 0);
- ret = gst_structure_get_int (structure, "width", &width);
- ret &= gst_structure_get_int (structure, "channels", &channels);
-
- if (G_UNLIKELY (!ret))
- return FALSE;
-
- *size = (width / 8) * channels;
+ *size = GST_AUDIO_INFO_BPF (&info);
return TRUE;
+
+ /* ERRORS */
+invalid_caps:
+ {
+ GST_ERROR_OBJECT (base, "invalid caps");
+ return FALSE;
+ }
}
static GstCaps *
gst_audio_resample_transform_caps (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps)
+ GstPadDirection direction, GstCaps * caps, GstCaps * filter)
{
const GValue *val;
GstStructure *s;
GstCaps *res;
+ gint i, n;
/* transform single caps into input_caps + input_caps with the rate
* field set to our supported range. This ensures that upstream knows
* about downstream's prefered rate(s) and can negotiate accordingly. */
- res = gst_caps_copy (caps);
-
- /* first, however, check if the caps contain a range for the rate field, in
- * which case that side isn't going to care much about the exact sample rate
- * chosen and we should just assume things will get fixated to something sane
- * and we may just as well offer our full range instead of the range in the
- * caps. If the rate is not an int range value, it's likely to express a
- * real preference or limitation and we should maintain that structure as
- * preference by putting it first into the transformed caps, and only add
- * our full rate range as second option */
- s = gst_caps_get_structure (res, 0);
- val = gst_structure_get_value (s, "rate");
- if (val == NULL || GST_VALUE_HOLDS_INT_RANGE (val)) {
- /* overwrite existing range, or add field if it doesn't exist yet */
- gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
- } else {
- /* append caps with full range to existing caps with non-range rate field */
+ res = gst_caps_new_empty ();
+ n = gst_caps_get_size (caps);
+ for (i = 0; i < n; i++) {
+ s = gst_caps_get_structure (caps, i);
+
+ /* If this is already expressed by the existing caps
+ * skip this structure */
+ if (i > 0 && gst_caps_is_subset_structure (res, s))
+ continue;
+
+ /* first, however, check if the caps contain a range for the rate field, in
+ * which case that side isn't going to care much about the exact sample rate
+ * chosen and we should just assume things will get fixated to something sane
+ * and we may just as well offer our full range instead of the range in the
+ * caps. If the rate is not an int range value, it's likely to express a
+ * real preference or limitation and we should maintain that structure as
+ * preference by putting it first into the transformed caps, and only add
+ * our full rate range as second option */
s = gst_structure_copy (s);
- gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
+ val = gst_structure_get_value (s, "rate");
+ if (val == NULL || GST_VALUE_HOLDS_INT_RANGE (val)) {
+ /* overwrite existing range, or add field if it doesn't exist yet */
+ gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
+ } else {
+ /* append caps with full range to existing caps with non-range rate field */
+ gst_caps_append_structure (res, gst_structure_copy (s));
+ gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
+ }
gst_caps_append_structure (res, s);
}
+ if (filter) {
+ GstCaps *intersection;
+
+ intersection =
+ gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
+ gst_caps_unref (res);
+ res = intersection;
+ }
+
return res;
}
resample->funcs->reset_mem (resample->state);
}
-static gboolean
-gst_audio_resample_parse_caps (GstCaps * incaps,
- GstCaps * outcaps, gint * width, gint * channels, gint * inrate,
- gint * outrate, gboolean * fp)
-{
- GstStructure *structure;
- gboolean ret;
- gint mywidth, myinrate, myoutrate, mychannels;
- gboolean myfp;
-
- GST_DEBUG ("incaps %" GST_PTR_FORMAT ", outcaps %"
- GST_PTR_FORMAT, incaps, outcaps);
-
- structure = gst_caps_get_structure (incaps, 0);
-
- if (gst_structure_has_name (structure, "audio/x-raw-float"))
- myfp = TRUE;
- else
- myfp = FALSE;
-
- ret = gst_structure_get_int (structure, "rate", &myinrate);
- ret &= gst_structure_get_int (structure, "channels", &mychannels);
- ret &= gst_structure_get_int (structure, "width", &mywidth);
- if (G_UNLIKELY (!ret))
- goto no_in_rate_channels;
-
- structure = gst_caps_get_structure (outcaps, 0);
- ret = gst_structure_get_int (structure, "rate", &myoutrate);
- if (G_UNLIKELY (!ret))
- goto no_out_rate;
-
- if (channels)
- *channels = mychannels;
- if (inrate)
- *inrate = myinrate;
- if (outrate)
- *outrate = myoutrate;
- if (width)
- *width = mywidth;
- if (fp)
- *fp = myfp;
-
- return TRUE;
-
- /* ERRORS */
-no_in_rate_channels:
- {
- GST_DEBUG ("could not get input rate and channels");
- return FALSE;
- }
-no_out_rate:
- {
- GST_DEBUG ("could not get output rate");
- return FALSE;
- }
-}
-
static gint
_gcd (gint a, gint b)
{
static gboolean
gst_audio_resample_transform_size (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
- guint * othersize)
+ GstPadDirection direction, GstCaps * caps, gsize size, GstCaps * othercaps,
+ gsize * othersize)
{
gboolean ret = TRUE;
+ GstAudioInfo in, out;
guint32 ratio_den, ratio_num;
gint inrate, outrate, gcd;
- gint bytes_per_samp, channels;
+ gint bpf;
- GST_LOG_OBJECT (base, "asked to transform size %d in direction %s",
- size, direction == GST_PAD_SINK ? "SINK" : "SRC");
+ GST_LOG_OBJECT (base, "asked to transform size %" G_GSIZE_FORMAT
+ " in direction %s", size, direction == GST_PAD_SINK ? "SINK" : "SRC");
/* Get sample width -> bytes_per_samp, channels, inrate, outrate */
- ret =
- gst_audio_resample_parse_caps (caps, othercaps, &bytes_per_samp,
- &channels, &inrate, &outrate, NULL);
+ ret = gst_audio_info_from_caps (&in, caps);
+ ret &= gst_audio_info_from_caps (&out, othercaps);
if (G_UNLIKELY (!ret)) {
GST_ERROR_OBJECT (base, "Wrong caps");
return FALSE;
}
/* Number of samples in either buffer is size / (width*channels) ->
* calculate the factor */
- bytes_per_samp = bytes_per_samp * channels / 8;
+ bpf = GST_AUDIO_INFO_BPF (&in);
+ inrate = GST_AUDIO_INFO_RATE (&in);
+ outrate = GST_AUDIO_INFO_RATE (&out);
+
/* Convert source buffer size to samples */
- size /= bytes_per_samp;
+ size /= bpf;
/* Simplify the conversion ratio factors */
gcd = _gcd (inrate, outrate);
if (direction == GST_PAD_SINK) {
/* asked to convert size of an incoming buffer. Round up the output size */
*othersize = gst_util_uint64_scale_int_ceil (size, ratio_den, ratio_num);
- *othersize *= bytes_per_samp;
+ *othersize *= bpf;
} else {
/* asked to convert size of an outgoing buffer. Round down the input size */
*othersize = gst_util_uint64_scale_int (size, ratio_num, ratio_den);
- *othersize *= bytes_per_samp;
+ *othersize *= bpf;
}
- GST_LOG_OBJECT (base, "transformed size %d to %d", size * bytes_per_samp,
- *othersize);
+ GST_LOG_OBJECT (base,
+ "transformed size %" G_GSIZE_FORMAT " to %" G_GSIZE_FORMAT,
+ size * bpf, *othersize);
return ret;
}
GstCaps * outcaps)
{
gboolean ret;
- gint width = 0, inrate = 0, outrate = 0, channels = 0;
+ gint width, inrate, outrate, channels;
gboolean fp;
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
+ GstAudioInfo in, out;
GST_LOG ("incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
- ret = gst_audio_resample_parse_caps (incaps, outcaps,
- &width, &channels, &inrate, &outrate, &fp);
+ if (!gst_audio_info_from_caps (&in, incaps))
+ goto invalid_incaps;
+ if (!gst_audio_info_from_caps (&out, outcaps))
+ goto invalid_outcaps;
- if (G_UNLIKELY (!ret))
- return FALSE;
+ /* FIXME do some checks */
+
+ /* take new values */
+ width = GST_AUDIO_FORMAT_INFO_WIDTH (in.finfo);
+ channels = GST_AUDIO_INFO_CHANNELS (&in);
+ inrate = GST_AUDIO_INFO_RATE (&in);
+ outrate = GST_AUDIO_INFO_RATE (&out);
+ fp = GST_AUDIO_FORMAT_INFO_IS_FLOAT (in.finfo);
ret =
gst_audio_resample_update_state (resample, width, channels, inrate,
if (G_UNLIKELY (!ret))
return FALSE;
- /* save caps so we can short-circuit in the size_transform if the caps
- * are the same */
- gst_caps_replace (&resample->sinkcaps, incaps);
- gst_caps_replace (&resample->srccaps, outcaps);
-
return TRUE;
+
+ /* ERROR */
+invalid_incaps:
+ {
+ GST_ERROR_OBJECT (base, "invalid incaps");
+ return FALSE;
+ }
+invalid_outcaps:
+ {
+ GST_ERROR_OBJECT (base, "invalid outcaps");
+ return FALSE;
+ }
}
#define GST_MAXINT24 (8388607)
guint out_len, out_processed;
gint err;
guint num, den;
+ guint8 *data;
g_assert (resample->state != NULL);
if (out_len == 0)
return;
- res =
- gst_pad_alloc_buffer_and_set_caps (GST_BASE_TRANSFORM_SRC_PAD (resample),
- GST_BUFFER_OFFSET_NONE, outsize,
- GST_PAD_CAPS (GST_BASE_TRANSFORM_SRC_PAD (resample)), &outbuf);
- if (G_UNLIKELY (res != GST_FLOW_OK)) {
- GST_WARNING_OBJECT (resample, "failed allocating buffer of %d bytes",
- outsize);
- return;
- }
+ outbuf = gst_buffer_new_and_alloc (outsize);
+
+ data = gst_buffer_map (outbuf, NULL, NULL, GST_MAP_WRITE);
if (resample->funcs->width != resample->width) {
/* need to convert data format; allocate workspace */
/* convert output format */
gst_audio_resample_convert_buffer (resample, resample->tmp_out,
- GST_BUFFER_DATA (outbuf), out_processed, TRUE);
+ data, out_processed, TRUE);
} else {
/* don't need to convert data format; process */
err = resample->funcs->process (resample->state, NULL, &in_processed,
- GST_BUFFER_DATA (outbuf), &out_processed);
+ data, &out_processed);
}
/* If we wrote more than allocated something is really wrong now
* and we should better abort immediately */
g_assert (out_len >= out_processed);
+ outsize = out_processed * resample->channels * (resample->width / 8);
+ gst_buffer_unmap (outbuf, data, outsize);
+
if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS)) {
GST_WARNING_OBJECT (resample, "Failed to process drain: %s",
resample->funcs->strerror (err));
return;
}
- GST_BUFFER_SIZE (outbuf) =
- out_processed * resample->channels * (resample->width / 8);
-
GST_LOG_OBJECT (resample,
"Pushing drain buffer of %u bytes with timestamp %" GST_TIME_FORMAT
" duration %" GST_TIME_FORMAT " offset %" G_GUINT64_FORMAT " offset_end %"
- G_GUINT64_FORMAT, GST_BUFFER_SIZE (outbuf),
+ G_GUINT64_FORMAT, outsize,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
GST_BUFFER_OFFSET_END (outbuf));
}
static gboolean
-gst_audio_resample_event (GstBaseTransform * base, GstEvent * event)
+gst_audio_resample_sink_event (GstBaseTransform * base, GstEvent * event)
{
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
resample->samples_out = 0;
resample->need_discont = TRUE;
break;
- case GST_EVENT_NEWSEGMENT:
+ case GST_EVENT_SEGMENT:
if (resample->state) {
guint latency = resample->funcs->get_input_latency (resample->state);
gst_audio_resample_push_drain (resample, latency);
break;
}
- return parent_class->event (base, event);
+ return GST_BASE_TRANSFORM_CLASS (parent_class)->sink_event (base, event);
}
static gboolean
gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf,
GstBuffer * outbuf)
{
+ gsize in_size, out_size;
+ guint8 *in_data, *out_data;
guint32 in_len, in_processed;
guint32 out_len, out_processed;
guint filt_len = resample->funcs->get_filt_len (resample->state);
- in_len = GST_BUFFER_SIZE (inbuf) / resample->channels;
- out_len = GST_BUFFER_SIZE (outbuf) / resample->channels;
+ in_data = gst_buffer_map (inbuf, &in_size, NULL, GST_MAP_READ);
+ out_data = gst_buffer_map (outbuf, &out_size, NULL, GST_MAP_WRITE);
+
+ in_len = in_size / resample->channels;
+ out_len = out_size / resample->channels;
in_len /= (resample->width / 8);
out_len /= (resample->width / 8);
else
out_processed = 0;
- memset (GST_BUFFER_DATA (outbuf), 0, GST_BUFFER_SIZE (outbuf));
+ memset (out_data, 0, out_size);
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
resample->num_gap_samples += in_len;
in_processed = in_len;
&resample->tmp_out_size, out_len * resample->channels *
(resample->funcs->width / 8))) {
GST_ERROR_OBJECT (resample, "failed to allocate workspace");
+ gst_buffer_unmap (inbuf, in_data, in_size);
+ gst_buffer_unmap (outbuf, out_data, out_size);
return GST_FLOW_ERROR;
}
/* convert input */
- gst_audio_resample_convert_buffer (resample, GST_BUFFER_DATA (inbuf),
+ gst_audio_resample_convert_buffer (resample, in_data,
resample->tmp_in, in_len, FALSE);
/* process */
/* convert output */
gst_audio_resample_convert_buffer (resample, resample->tmp_out,
- GST_BUFFER_DATA (outbuf), out_processed, TRUE);
+ out_data, out_processed, TRUE);
} else {
/* no format conversion required; process */
err = resample->funcs->process (resample->state,
- GST_BUFFER_DATA (inbuf), &in_processed,
- GST_BUFFER_DATA (outbuf), &out_processed);
+ in_data, &in_processed, out_data, &out_processed);
}
if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS)) {
GST_ERROR_OBJECT (resample, "Failed to convert data: %s",
resample->funcs->strerror (err));
+ gst_buffer_unmap (inbuf, in_data, in_size);
+ gst_buffer_unmap (outbuf, out_data, out_size);
return GST_FLOW_ERROR;
}
}
resample->samples_out += out_processed;
resample->samples_in += in_len;
- GST_BUFFER_SIZE (outbuf) =
- out_processed * resample->channels * (resample->width / 8);
+ out_size = out_processed * resample->channels * (resample->width / 8);
+ gst_buffer_unmap (inbuf, in_data, in_size);
+ gst_buffer_unmap (outbuf, out_data, out_size);
GST_LOG_OBJECT (resample,
"Converted to buffer of %" G_GUINT32_FORMAT
- " samples (%u bytes) with timestamp %" GST_TIME_FORMAT ", duration %"
- GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT ", offset_end %"
- G_GUINT64_FORMAT, out_processed, GST_BUFFER_SIZE (outbuf),
+ " samples (%" G_GSIZE_FORMAT " bytes) with timestamp %" GST_TIME_FORMAT
+ ", duration %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT
+ ", offset_end %" G_GUINT64_FORMAT, out_processed, out_size,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
GstBuffer * outbuf)
{
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
- gulong size;
GstFlowReturn ret;
if (resample->state == NULL) {
gst_audio_resample_get_funcs (resample->width, resample->fp);
}
- size = GST_BUFFER_SIZE (inbuf);
-
GST_LOG_OBJECT (resample, "transforming buffer of %ld bytes, ts %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
- size, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)),
+ gst_buffer_get_size (inbuf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)),
GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
}
static gboolean
-gst_audio_resample_query (GstPad * pad, GstQuery * query)
+gst_audio_resample_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
- GstAudioResample *resample = GST_AUDIO_RESAMPLE (gst_pad_get_parent (pad));
+ GstAudioResample *resample = GST_AUDIO_RESAMPLE (parent);
GstBaseTransform *trans;
gboolean res = TRUE;
- if (G_UNLIKELY (resample == NULL))
- return FALSE;
trans = GST_BASE_TRANSFORM (resample);
GstClockTime min, max;
gboolean live;
guint64 latency;
- GstPad *peer;
gint rate = resample->inrate;
gint resampler_latency;
if (gst_base_transform_is_passthrough (trans))
resampler_latency = 0;
- if ((peer = gst_pad_get_peer (GST_BASE_TRANSFORM_SINK_PAD (trans)))) {
- if ((res = gst_pad_query (peer, query))) {
- gst_query_parse_latency (query, &live, &min, &max);
+ if ((res =
+ gst_pad_peer_query (GST_BASE_TRANSFORM_SINK_PAD (trans),
+ query))) {
+ gst_query_parse_latency (query, &live, &min, &max);
- GST_DEBUG_OBJECT (resample, "Peer latency: min %"
- GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
- GST_TIME_ARGS (min), GST_TIME_ARGS (max));
+ GST_DEBUG_OBJECT (resample, "Peer latency: min %"
+ GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (min), GST_TIME_ARGS (max));
- /* add our own latency */
- if (rate != 0 && resampler_latency != 0)
- latency = gst_util_uint64_scale_round (resampler_latency,
- GST_SECOND, rate);
- else
- latency = 0;
+ /* add our own latency */
+ if (rate != 0 && resampler_latency != 0)
+ latency = gst_util_uint64_scale_round (resampler_latency,
+ GST_SECOND, rate);
+ else
+ latency = 0;
- GST_DEBUG_OBJECT (resample, "Our latency: %" GST_TIME_FORMAT,
- GST_TIME_ARGS (latency));
+ GST_DEBUG_OBJECT (resample, "Our latency: %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (latency));
- min += latency;
- if (GST_CLOCK_TIME_IS_VALID (max))
- max += latency;
+ min += latency;
+ if (GST_CLOCK_TIME_IS_VALID (max))
+ max += latency;
- GST_DEBUG_OBJECT (resample, "Calculated total latency : min %"
- GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
- GST_TIME_ARGS (min), GST_TIME_ARGS (max));
+ GST_DEBUG_OBJECT (resample, "Calculated total latency : min %"
+ GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (min), GST_TIME_ARGS (max));
- gst_query_set_latency (query, live, min, max);
- }
- gst_object_unref (peer);
+ gst_query_set_latency (query, live, min, max);
}
break;
}
default:
- res = gst_pad_query_default (pad, query);
+ res = gst_pad_query_default (pad, parent, query);
break;
}
- gst_object_unref (resample);
return res;
}
-static const GstQueryType *
-gst_audio_resample_query_type (GstPad * pad)
-{
- static const GstQueryType types[] = {
- GST_QUERY_LATENCY,
- 0
- };
-
- return types;
-}
-
static void
gst_audio_resample_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
quality, resample->fp);
GST_BASE_TRANSFORM_UNLOCK (resample);
break;
- case PROP_FILTER_LENGTH:{
- gint filter_length = g_value_get_int (value);
-
- GST_BASE_TRANSFORM_LOCK (resample);
- if (filter_length <= 8)
- quality = 0;
- else if (filter_length <= 16)
- quality = 1;
- else if (filter_length <= 32)
- quality = 2;
- else if (filter_length <= 48)
- quality = 3;
- else if (filter_length <= 64)
- quality = 4;
- else if (filter_length <= 80)
- quality = 5;
- else if (filter_length <= 96)
- quality = 6;
- else if (filter_length <= 128)
- quality = 7;
- else if (filter_length <= 160)
- quality = 8;
- else if (filter_length <= 192)
- quality = 9;
- else
- quality = 10;
-
- GST_DEBUG_OBJECT (resample, "new quality %d", quality);
-
- gst_audio_resample_update_state (resample, resample->width,
- resample->channels, resample->inrate, resample->outrate,
- quality, resample->fp);
- GST_BASE_TRANSFORM_UNLOCK (resample);
- break;
- }
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
case PROP_QUALITY:
g_value_set_int (value, resample->quality);
break;
- case PROP_FILTER_LENGTH:
- switch (resample->quality) {
- case 0:
- g_value_set_int (value, 8);
- break;
- case 1:
- g_value_set_int (value, 16);
- break;
- case 2:
- g_value_set_int (value, 32);
- break;
- case 3:
- g_value_set_int (value, 48);
- break;
- case 4:
- g_value_set_int (value, 64);
- break;
- case 5:
- g_value_set_int (value, 80);
- break;
- case 6:
- g_value_set_int (value, 96);
- break;
- case 7:
- g_value_set_int (value, 128);
- break;
- case 8:
- g_value_set_int (value, 160);
- break;
- case 9:
- g_value_set_int (value, 192);
- break;
- case 10:
- g_value_set_int (value, 256);
- break;
- }
- break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
projects / platform / upstream / gstreamer.git / blobdiff