/* GStreamer
* Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) 2003,2004 David A. Schleef <ds@schleef.org>
+ * Copyright (C) 2007-2008 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
-/* Element-Checklist-Version: 5 */
/**
* SECTION:element-audioresample
*
- * Audioresample resamples raw audio buffers to different sample rates using
+ * audioresample resamples raw audio buffers to different sample rates using
* a configurable windowing function to enhance quality.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! audio/x-raw-int, rate=8000 ! alsasink
- * ]| Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa.
+ * ]| 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>
- *
- * Last reviewed on 2006-03-02 (0.10.4)
+ */
+
+/* TODO:
+ * - Enable SSE/ARM optimizations and select at runtime
*/
#ifdef HAVE_CONFIG_H
#include <string.h>
#include <math.h>
-/*#define DEBUG_ENABLED */
#include "gstaudioresample.h"
+#include <gst/gstutils.h>
#include <gst/audio/audio.h>
#include <gst/base/gstbasetransform.h>
-GST_DEBUG_CATEGORY_STATIC (audioresample_debug);
-#define GST_CAT_DEFAULT audioresample_debug
-
-/* elementfactory information */
-static const GstElementDetails gst_audioresample_details =
-GST_ELEMENT_DETAILS ("Audio scaler",
- "Filter/Converter/Audio",
- "Resample audio",
- "David Schleef <ds@schleef.org>");
+#ifndef DISABLE_ORC
+#include <orc/orc.h>
+#include <orc-test/orctest.h>
+#include <orc-test/orcprofile.h>
+#endif
-#define DEFAULT_FILTERLEN 16
+GST_DEBUG_CATEGORY (audio_resample_debug);
+#define GST_CAT_DEFAULT audio_resample_debug
enum
{
PROP_0,
- PROP_FILTERLEN
+ PROP_QUALITY,
+ PROP_FILTER_LENGTH
};
#define SUPPORTED_CAPS \
GST_STATIC_CAPS ( \
- "audio/x-raw-int, " \
- "rate = (int) [ 1, MAX ], " \
+ "audio/x-raw-float, " \
+ "rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
- "width = (int) 16, " \
- "depth = (int) 16, " \
- "signed = (boolean) true;" \
+ "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-float, " \
- "rate = (int) [ 1, MAX ], " \
+ "signed = (boolean) true; " \
+ "audio/x-raw-int, " \
+ "rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
- "width = (int) 32; " \
- "audio/x-raw-float, " \
- "rate = (int) [ 1, MAX ], " \
+ "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) 64" \
+ "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" \
)
-static GstStaticPadTemplate gst_audioresample_sink_template =
+/* If TRUE integer arithmetic resampling is faster and will be used if appropiate */
+#if defined AUDIORESAMPLE_FORMAT_INT
+static gboolean gst_audio_resample_use_int = TRUE;
+#elif defined AUDIORESAMPLE_FORMAT_FLOAT
+static gboolean gst_audio_resample_use_int = FALSE;
+#else
+static gboolean gst_audio_resample_use_int = FALSE;
+#endif
+
+static GstStaticPadTemplate gst_audio_resample_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
-static GstStaticPadTemplate gst_audioresample_src_template =
+static GstStaticPadTemplate gst_audio_resample_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
-static void gst_audioresample_set_property (GObject * object,
+static void gst_audio_resample_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
-static void gst_audioresample_get_property (GObject * object,
+static void gst_audio_resample_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
/* vmethods */
-static gboolean audioresample_get_unit_size (GstBaseTransform * base,
- GstCaps * caps, guint * size);
-static GstCaps *audioresample_transform_caps (GstBaseTransform * base,
+static gboolean gst_audio_resample_get_unit_size (GstBaseTransform * base,
+ GstCaps * caps, gsize * size);
+static GstCaps *gst_audio_resample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps);
-static void audioresample_fixate_caps (GstBaseTransform * base,
+static void gst_audio_resample_fixate_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * othercaps);
-static gboolean audioresample_transform_size (GstBaseTransform * trans,
- GstPadDirection direction, GstCaps * incaps, guint insize,
- GstCaps * outcaps, guint * outsize);
-static gboolean audioresample_set_caps (GstBaseTransform * base,
+static gboolean gst_audio_resample_transform_size (GstBaseTransform * trans,
+ 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 audioresample_pushthrough (GstAudioresample *
- audioresample);
-static GstFlowReturn audioresample_transform (GstBaseTransform * base,
+static GstFlowReturn gst_audio_resample_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
-static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event);
-static gboolean audioresample_start (GstBaseTransform * base);
-static gboolean audioresample_stop (GstBaseTransform * base);
-
-static gboolean audioresample_query (GstPad * pad, GstQuery * query);
-static const GstQueryType *audioresample_query_type (GstPad * pad);
+static gboolean gst_audio_resample_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);
-#define DEBUG_INIT(bla) \
- GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0, "audio resampling element");
-
-GST_BOILERPLATE_FULL (GstAudioresample, gst_audioresample, GstBaseTransform,
- GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
+GST_BOILERPLATE (GstAudioResample, gst_audio_resample, GstBaseTransform,
+ GST_TYPE_BASE_TRANSFORM);
static void
-gst_audioresample_base_init (gpointer g_class)
+gst_audio_resample_base_init (gpointer g_class)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_audioresample_src_template));
+ gst_static_pad_template_get (&gst_audio_resample_src_template));
gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_audioresample_sink_template));
+ gst_static_pad_template_get (&gst_audio_resample_sink_template));
- gst_element_class_set_details (gstelement_class, &gst_audioresample_details);
+ gst_element_class_set_details_simple (gstelement_class, "Audio resampler",
+ "Filter/Converter/Audio", "Resamples audio",
+ "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
}
static void
-gst_audioresample_class_init (GstAudioresampleClass * klass)
+gst_audio_resample_class_init (GstAudioResampleClass * klass)
{
- GObjectClass *gobject_class;
-
- gobject_class = (GObjectClass *) klass;
+ GObjectClass *gobject_class = (GObjectClass *) klass;
- gobject_class->set_property = gst_audioresample_set_property;
- gobject_class->get_property = gst_audioresample_get_property;
+ gobject_class->set_property = gst_audio_resample_set_property;
+ gobject_class->get_property = gst_audio_resample_get_property;
- g_object_class_install_property (gobject_class, PROP_FILTERLEN,
- g_param_spec_int ("filter-length", "filter length",
- "Length of the resample filter", 0, G_MAXINT, DEFAULT_FILTERLEN,
+ g_object_class_install_property (gobject_class, PROP_QUALITY,
+ g_param_spec_int ("quality", "Quality", "Resample quality with 0 being "
+ "the lowest and 10 being the best",
+ SPEEX_RESAMPLER_QUALITY_MIN, SPEEX_RESAMPLER_QUALITY_MAX,
+ 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_BASE_TRANSFORM_CLASS (klass)->start =
- GST_DEBUG_FUNCPTR (audioresample_start);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_start);
GST_BASE_TRANSFORM_CLASS (klass)->stop =
- GST_DEBUG_FUNCPTR (audioresample_stop);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_stop);
GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
- GST_DEBUG_FUNCPTR (audioresample_transform_size);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_transform_size);
GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
- GST_DEBUG_FUNCPTR (audioresample_get_unit_size);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_get_unit_size);
GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
- GST_DEBUG_FUNCPTR (audioresample_transform_caps);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_transform_caps);
GST_BASE_TRANSFORM_CLASS (klass)->fixate_caps =
- GST_DEBUG_FUNCPTR (audioresample_fixate_caps);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_fixate_caps);
GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
- GST_DEBUG_FUNCPTR (audioresample_set_caps);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_set_caps);
GST_BASE_TRANSFORM_CLASS (klass)->transform =
- GST_DEBUG_FUNCPTR (audioresample_transform);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_transform);
GST_BASE_TRANSFORM_CLASS (klass)->event =
- GST_DEBUG_FUNCPTR (audioresample_event);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_event);
GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE;
}
static void
-gst_audioresample_init (GstAudioresample * audioresample,
- GstAudioresampleClass * klass)
+gst_audio_resample_init (GstAudioResample * resample,
+ GstAudioResampleClass * klass)
{
- GstBaseTransform *trans;
-
- trans = GST_BASE_TRANSFORM (audioresample);
-
- /* buffer alloc passthrough is too impossible. FIXME, it
- * is trivial in the passthrough case. */
- gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
+ GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
- audioresample->filter_length = DEFAULT_FILTERLEN;
+ resample->quality = SPEEX_RESAMPLER_QUALITY_DEFAULT;
- audioresample->need_discont = FALSE;
-
- gst_pad_set_query_function (trans->srcpad, audioresample_query);
- gst_pad_set_query_type_function (trans->srcpad, audioresample_query_type);
+ 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 */
static gboolean
-audioresample_start (GstBaseTransform * base)
+gst_audio_resample_start (GstBaseTransform * base)
{
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
+ GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
+
+ resample->need_discont = TRUE;
- audioresample->resample = resample_new ();
- audioresample->ts_offset = -1;
- audioresample->offset = -1;
- audioresample->next_ts = -1;
+ resample->num_gap_samples = 0;
+ resample->num_nongap_samples = 0;
+ resample->t0 = GST_CLOCK_TIME_NONE;
+ resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
+ resample->out_offset0 = GST_BUFFER_OFFSET_NONE;
+ resample->samples_in = 0;
+ resample->samples_out = 0;
- resample_set_filter_length (audioresample->resample,
- audioresample->filter_length);
+ resample->tmp_in = NULL;
+ resample->tmp_in_size = 0;
+ resample->tmp_out = NULL;
+ resample->tmp_out_size = 0;
return TRUE;
}
static gboolean
-audioresample_stop (GstBaseTransform * base)
+gst_audio_resample_stop (GstBaseTransform * base)
{
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
+ GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
- if (audioresample->resample) {
- resample_free (audioresample->resample);
- audioresample->resample = NULL;
+ if (resample->state) {
+ resample->funcs->destroy (resample->state);
+ resample->state = NULL;
}
- gst_caps_replace (&audioresample->sinkcaps, NULL);
- gst_caps_replace (&audioresample->srccaps, NULL);
+ resample->funcs = NULL;
+
+ g_free (resample->tmp_in);
+ resample->tmp_in = NULL;
+ resample->tmp_in_size = 0;
+
+ g_free (resample->tmp_out);
+ 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
-audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
- guint * size)
+gst_audio_resample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
+ gsize * size)
{
gint width, channels;
GstStructure *structure;
gboolean ret;
- g_assert (size);
+ g_return_val_if_fail (size != NULL, FALSE);
/* 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);
- g_return_val_if_fail (ret, FALSE);
- *size = width * channels / 8;
+ if (G_UNLIKELY (!ret))
+ return FALSE;
+
+ *size = (width / 8) * channels;
return TRUE;
}
static GstCaps *
-audioresample_transform_caps (GstBaseTransform * base,
+gst_audio_resample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps)
{
+ const GValue *val;
+ GstStructure *s;
GstCaps *res;
- GstStructure *structure;
- /* transform caps gives one single caps so we can just replace
- * the rate property with our range. */
+ /* 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);
- structure = gst_caps_get_structure (res, 0);
- gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
+
+ /* 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 */
+ s = gst_structure_copy (s);
+ gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
+ gst_caps_append_structure (res, s);
+ }
return res;
}
/* Fixate rate to the allowed rate that has the smallest difference */
static void
-audioresample_fixate_caps (GstBaseTransform * base,
+gst_audio_resample_fixate_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * othercaps)
{
GstStructure *s;
gint rate;
s = gst_caps_get_structure (caps, 0);
- if (!gst_structure_get_int (s, "rate", &rate))
+ if (G_UNLIKELY (!gst_structure_get_int (s, "rate", &rate)))
return;
s = gst_caps_get_structure (othercaps, 0);
gst_structure_fixate_field_nearest_int (s, "rate", rate);
}
+static const SpeexResampleFuncs *
+gst_audio_resample_get_funcs (gint width, gboolean fp)
+{
+ const SpeexResampleFuncs *funcs = NULL;
+
+ if (gst_audio_resample_use_int && (width == 8 || width == 16) && !fp)
+ funcs = &int_funcs;
+ else if ((!gst_audio_resample_use_int && (width == 8 || width == 16) && !fp)
+ || (width == 32 && fp))
+ funcs = &float_funcs;
+ else if ((width == 64 && fp) || ((width == 32 || width == 24) && !fp))
+ funcs = &double_funcs;
+ else
+ g_assert_not_reached ();
+
+ return funcs;
+}
+
+static SpeexResamplerState *
+gst_audio_resample_init_state (GstAudioResample * resample, gint width,
+ gint channels, gint inrate, gint outrate, gint quality, gboolean fp)
+{
+ SpeexResamplerState *ret = NULL;
+ gint err = RESAMPLER_ERR_SUCCESS;
+ const SpeexResampleFuncs *funcs = gst_audio_resample_get_funcs (width, fp);
+
+ ret = funcs->init (channels, inrate, outrate, quality, &err);
+
+ if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS)) {
+ GST_ERROR_OBJECT (resample, "Failed to create resampler state: %s",
+ funcs->strerror (err));
+ return NULL;
+ }
+
+ funcs->skip_zeros (ret);
+
+ return ret;
+}
+
+static gboolean
+gst_audio_resample_update_state (GstAudioResample * resample, gint width,
+ gint channels, gint inrate, gint outrate, gint quality, gboolean fp)
+{
+ gboolean ret = TRUE;
+ gboolean updated_latency = FALSE;
+
+ updated_latency = (resample->inrate != inrate
+ || quality != resample->quality) && resample->state != NULL;
+
+ if (resample->state == NULL) {
+ ret = TRUE;
+ } else if (resample->channels != channels || fp != resample->fp
+ || width != resample->width) {
+ resample->funcs->destroy (resample->state);
+ resample->state =
+ gst_audio_resample_init_state (resample, width, channels, inrate,
+ outrate, quality, fp);
+
+ resample->funcs = gst_audio_resample_get_funcs (width, fp);
+ ret = (resample->state != NULL);
+ } else if (resample->inrate != inrate || resample->outrate != outrate) {
+ gint err = RESAMPLER_ERR_SUCCESS;
+
+ err = resample->funcs->set_rate (resample->state, inrate, outrate);
+
+ if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS))
+ GST_ERROR_OBJECT (resample, "Failed to update rate: %s",
+ resample->funcs->strerror (err));
+
+ ret = (err == RESAMPLER_ERR_SUCCESS);
+ } else if (quality != resample->quality) {
+ gint err = RESAMPLER_ERR_SUCCESS;
+
+ err = resample->funcs->set_quality (resample->state, quality);
+
+ if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS))
+ GST_ERROR_OBJECT (resample, "Failed to update quality: %s",
+ resample->funcs->strerror (err));
+
+ ret = (err == RESAMPLER_ERR_SUCCESS);
+ }
+
+ resample->width = width;
+ resample->channels = channels;
+ resample->fp = fp;
+ resample->quality = quality;
+ resample->inrate = inrate;
+ resample->outrate = outrate;
+
+ if (updated_latency)
+ gst_element_post_message (GST_ELEMENT (resample),
+ gst_message_new_latency (GST_OBJECT (resample)));
+
+ return ret;
+}
+
+static void
+gst_audio_resample_reset_state (GstAudioResample * resample)
+{
+ if (resample->state)
+ resample->funcs->reset_mem (resample->state);
+}
+
static gboolean
-resample_set_state_from_caps (ResampleState * state, GstCaps * incaps,
- GstCaps * outcaps, gint * channels, gint * inrate, gint * outrate)
+gst_audio_resample_parse_caps (GstCaps * incaps,
+ GstCaps * outcaps, gint * width, gint * channels, gint * inrate,
+ gint * outrate, gboolean * fp)
{
GstStructure *structure;
gboolean ret;
- gint myinrate, myoutrate;
- int mychannels;
- gint width, depth;
- ResampleFormat format;
+ 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);
- /* get width */
- ret = gst_structure_get_int (structure, "width", &width);
- if (!ret)
- goto no_width;
-
- /* figure out the format */
- if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float")) {
- if (width == 32)
- format = RESAMPLE_FORMAT_F32;
- else if (width == 64)
- format = RESAMPLE_FORMAT_F64;
- else
- goto wrong_depth;
- } else {
- /* for int, depth and width must be the same */
- ret = gst_structure_get_int (structure, "depth", &depth);
- if (!ret || width != depth)
- goto not_equal;
-
- if (width == 16)
- format = RESAMPLE_FORMAT_S16;
- else if (width == 32)
- format = RESAMPLE_FORMAT_S32;
- else
- goto wrong_depth;
- }
+ if (g_str_equal (gst_structure_get_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);
- if (!ret)
+ 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 (!ret)
+ if (G_UNLIKELY (!ret))
goto no_out_rate;
if (channels)
*inrate = myinrate;
if (outrate)
*outrate = myoutrate;
-
- resample_set_format (state, format);
- resample_set_n_channels (state, mychannels);
- resample_set_input_rate (state, myinrate);
- resample_set_output_rate (state, myoutrate);
+ if (width)
+ *width = mywidth;
+ if (fp)
+ *fp = myfp;
return TRUE;
/* ERRORS */
-no_width:
- {
- GST_DEBUG ("failed to get width from caps");
- return FALSE;
- }
-not_equal:
- {
- GST_DEBUG ("width %d and depth %d must be the same", width, depth);
- return FALSE;
- }
-wrong_depth:
- {
- GST_DEBUG ("unknown depth %d found", depth);
- return FALSE;
- }
no_in_rate_channels:
{
GST_DEBUG ("could not get input rate and channels");
}
}
+static gint
+_gcd (gint a, gint b)
+{
+ while (b != 0) {
+ int temp = a;
+
+ a = b;
+ b = temp % b;
+ }
+
+ return ABS (a);
+}
+
static gboolean
-audioresample_transform_size (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
- guint * othersize)
-{
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
- ResampleState *state;
- GstCaps *srccaps, *sinkcaps;
- gboolean use_internal = FALSE; /* whether we use the internal state */
+gst_audio_resample_transform_size (GstBaseTransform * base,
+ GstPadDirection direction, GstCaps * caps, gsize size, GstCaps * othercaps,
+ gsize * othersize)
+{
gboolean ret = TRUE;
+ guint32 ratio_den, ratio_num;
+ gint inrate, outrate, gcd;
+ gint bytes_per_samp, channels;
GST_LOG_OBJECT (base, "asked to transform size %d in direction %s",
size, direction == GST_PAD_SINK ? "SINK" : "SRC");
- if (direction == GST_PAD_SINK) {
- sinkcaps = caps;
- srccaps = othercaps;
- } else {
- sinkcaps = othercaps;
- srccaps = caps;
- }
- /* if the caps are the ones that _set_caps got called with; we can use
- * our own state; otherwise we'll have to create a state */
- if (gst_caps_is_equal (sinkcaps, audioresample->sinkcaps) &&
- gst_caps_is_equal (srccaps, audioresample->srccaps)) {
- use_internal = TRUE;
- state = audioresample->resample;
- } else {
- GST_DEBUG_OBJECT (audioresample,
- "caps are not the set caps, creating state");
- state = resample_new ();
- resample_set_filter_length (state, audioresample->filter_length);
- resample_set_state_from_caps (state, sinkcaps, srccaps, NULL, NULL, NULL);
+ /* Get sample width -> bytes_per_samp, channels, inrate, outrate */
+ ret =
+ gst_audio_resample_parse_caps (caps, othercaps, &bytes_per_samp,
+ &channels, &inrate, &outrate, NULL);
+ 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;
+ /* Convert source buffer size to samples */
+ size /= bytes_per_samp;
+
+ /* Simplify the conversion ratio factors */
+ gcd = _gcd (inrate, outrate);
+ ratio_num = inrate / gcd;
+ ratio_den = outrate / gcd;
if (direction == GST_PAD_SINK) {
- /* asked to convert size of an incoming buffer */
- *othersize = resample_get_output_size_for_input (state, size);
+ /* 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;
} else {
- /* asked to convert size of an outgoing buffer */
- *othersize = resample_get_input_size_for_output (state, size);
+ /* 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;
}
- g_assert (*othersize % state->sample_size == 0);
- /* we make room for one extra sample, given that the resampling filter
- * can output an extra one for non-integral i_rate/o_rate */
- GST_LOG_OBJECT (base, "transformed size %d to %d", size, *othersize);
-
- if (!use_internal) {
- resample_free (state);
- }
+ GST_LOG_OBJECT (base, "transformed size %d to %d", size * bytes_per_samp,
+ *othersize);
return ret;
}
static gboolean
-audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
+gst_audio_resample_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps)
{
gboolean ret;
- gint inrate, outrate;
- int channels;
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
+ gint width = 0, inrate = 0, outrate = 0, channels = 0;
+ gboolean fp;
+ GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
- GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %"
+ GST_LOG ("incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
- ret = resample_set_state_from_caps (audioresample->resample, incaps, outcaps,
- &channels, &inrate, &outrate);
+ ret = gst_audio_resample_parse_caps (incaps, outcaps,
+ &width, &channels, &inrate, &outrate, &fp);
- g_return_val_if_fail (ret, FALSE);
+ if (G_UNLIKELY (!ret))
+ return FALSE;
- audioresample->channels = channels;
- GST_DEBUG_OBJECT (audioresample, "set channels to %d", channels);
- audioresample->i_rate = inrate;
- GST_DEBUG_OBJECT (audioresample, "set i_rate to %d", inrate);
- audioresample->o_rate = outrate;
- GST_DEBUG_OBJECT (audioresample, "set o_rate to %d", outrate);
+ ret =
+ gst_audio_resample_update_state (resample, width, channels, inrate,
+ outrate, resample->quality, fp);
+
+ 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 (&audioresample->sinkcaps, incaps);
- gst_caps_replace (&audioresample->srccaps, outcaps);
+ gst_caps_replace (&resample->sinkcaps, incaps);
+ gst_caps_replace (&resample->srccaps, outcaps);
return TRUE;
}
-static gboolean
-audioresample_event (GstBaseTransform * base, GstEvent * event)
+#define GST_MAXINT24 (8388607)
+#define GST_MININT24 (-8388608)
+
+#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
+#define GST_READ_UINT24 GST_READ_UINT24_LE
+#define GST_WRITE_UINT24 GST_WRITE_UINT24_LE
+#else
+#define GST_READ_UINT24 GST_READ_UINT24_BE
+#define GST_WRITE_UINT24 GST_WRITE_UINT24_BE
+#endif
+
+static void
+gst_audio_resample_convert_buffer (GstAudioResample * resample,
+ const guint8 * in, guint8 * out, guint len, gboolean inverse)
{
- GstAudioresample *audioresample;
+ len *= resample->channels;
+
+ if (inverse) {
+ if (gst_audio_resample_use_int && resample->width == 8 && !resample->fp) {
+ gint8 *o = (gint8 *) out;
+ gint16 *i = (gint16 *) in;
+ gint32 tmp;
+
+ while (len) {
+ tmp = *i + (G_MAXINT8 >> 1);
+ *o = CLAMP (tmp >> 8, G_MININT8, G_MAXINT8);
+ o++;
+ i++;
+ len--;
+ }
+ } else if (!gst_audio_resample_use_int && resample->width == 8
+ && !resample->fp) {
+ gint8 *o = (gint8 *) out;
+ gfloat *i = (gfloat *) in;
+ gfloat tmp;
+
+ while (len) {
+ tmp = *i;
+ *o = (gint8) CLAMP (tmp * G_MAXINT8 + 0.5, G_MININT8, G_MAXINT8);
+ o++;
+ i++;
+ len--;
+ }
+ } else if (!gst_audio_resample_use_int && resample->width == 16
+ && !resample->fp) {
+ gint16 *o = (gint16 *) out;
+ gfloat *i = (gfloat *) in;
+ gfloat tmp;
+
+ while (len) {
+ tmp = *i;
+ *o = (gint16) CLAMP (tmp * G_MAXINT16 + 0.5, G_MININT16, G_MAXINT16);
+ o++;
+ i++;
+ len--;
+ }
+ } else if (resample->width == 24 && !resample->fp) {
+ guint8 *o = (guint8 *) out;
+ gdouble *i = (gdouble *) in;
+ gdouble tmp;
+
+ while (len) {
+ tmp = *i;
+ GST_WRITE_UINT24 (o, (gint32) CLAMP (tmp * GST_MAXINT24 + 0.5,
+ GST_MININT24, GST_MAXINT24));
+ o += 3;
+ i++;
+ len--;
+ }
+ } else if (resample->width == 32 && !resample->fp) {
+ gint32 *o = (gint32 *) out;
+ gdouble *i = (gdouble *) in;
+ gdouble tmp;
+
+ while (len) {
+ tmp = *i;
+ *o = (gint32) CLAMP (tmp * G_MAXINT32 + 0.5, G_MININT32, G_MAXINT32);
+ o++;
+ i++;
+ len--;
+ }
+ } else {
+ g_assert_not_reached ();
+ }
+ } else {
+ if (gst_audio_resample_use_int && resample->width == 8 && !resample->fp) {
+ gint8 *i = (gint8 *) in;
+ gint16 *o = (gint16 *) out;
+ gint32 tmp;
+
+ while (len) {
+ tmp = *i;
+ *o = tmp << 8;
+ o++;
+ i++;
+ len--;
+ }
+ } else if (!gst_audio_resample_use_int && resample->width == 8
+ && !resample->fp) {
+ gint8 *i = (gint8 *) in;
+ gfloat *o = (gfloat *) out;
+ gfloat tmp;
+
+ while (len) {
+ tmp = *i;
+ *o = tmp / G_MAXINT8;
+ o++;
+ i++;
+ len--;
+ }
+ } else if (!gst_audio_resample_use_int && resample->width == 16
+ && !resample->fp) {
+ gint16 *i = (gint16 *) in;
+ gfloat *o = (gfloat *) out;
+ gfloat tmp;
+
+ while (len) {
+ tmp = *i;
+ *o = tmp / G_MAXINT16;
+ o++;
+ i++;
+ len--;
+ }
+ } else if (resample->width == 24 && !resample->fp) {
+ guint8 *i = (guint8 *) in;
+ gdouble *o = (gdouble *) out;
+ gdouble tmp;
+ guint32 tmp2;
+
+ while (len) {
+ tmp2 = GST_READ_UINT24 (i);
+ if (tmp2 & 0x00800000)
+ tmp2 |= 0xff000000;
+ tmp = (gint32) tmp2;
+ *o = tmp / GST_MAXINT24;
+ o++;
+ i += 3;
+ len--;
+ }
+ } else if (resample->width == 32 && !resample->fp) {
+ gint32 *i = (gint32 *) in;
+ gdouble *o = (gdouble *) out;
+ gdouble tmp;
+
+ while (len) {
+ tmp = *i;
+ *o = tmp / G_MAXINT32;
+ o++;
+ i++;
+ len--;
+ }
+ } else {
+ g_assert_not_reached ();
+ }
+ }
+}
- audioresample = GST_AUDIORESAMPLE (base);
+static guint8 *
+gst_audio_resample_workspace_realloc (guint8 ** workspace, guint * size,
+ guint new_size)
+{
+ guint8 *new;
+ if (new_size <= *size)
+ /* no need to resize */
+ return *workspace;
+ new = g_realloc (*workspace, new_size);
+ if (!new)
+ /* failure (re)allocating memeory */
+ return NULL;
+ /* success */
+ *workspace = new;
+ *size = new_size;
+ return *workspace;
+}
+
+/* Push history_len zeros into the filter, but discard the output. */
+static void
+gst_audio_resample_dump_drain (GstAudioResample * resample, guint history_len)
+{
+ gint outsize;
+ guint in_len, in_processed;
+ guint out_len, out_processed;
+ guint num, den;
+ gpointer buf;
+
+ g_assert (resample->state != NULL);
+
+ resample->funcs->get_ratio (resample->state, &num, &den);
+
+ in_len = in_processed = history_len;
+ out_processed = out_len =
+ gst_util_uint64_scale_int_ceil (history_len, den, num);
+ outsize = out_len * resample->channels * (resample->funcs->width / 8);
+
+ if (out_len == 0)
+ return;
+
+ buf = g_malloc (outsize);
+ resample->funcs->process (resample->state, NULL, &in_processed, buf,
+ &out_processed);
+ g_free (buf);
+
+ g_assert (in_len == in_processed);
+}
+
+static void
+gst_audio_resample_push_drain (GstAudioResample * resample, guint history_len)
+{
+ GstBuffer *outbuf;
+ GstFlowReturn res;
+ gint outsize;
+ guint in_len, in_processed;
+ guint out_len, out_processed;
+ gint err;
+ guint num, den;
+ guint8 *data;
+
+ g_assert (resample->state != NULL);
+
+ /* Don't drain samples if we were reset. */
+ if (!GST_CLOCK_TIME_IS_VALID (resample->t0))
+ return;
+
+ resample->funcs->get_ratio (resample->state, &num, &den);
+
+ in_len = in_processed = history_len;
+ out_len = out_processed =
+ gst_util_uint64_scale_int_ceil (history_len, den, num);
+ outsize = out_len * resample->channels * (resample->width / 8);
+
+ 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;
+ }
+
+ data = gst_buffer_map (outbuf, NULL, NULL, GST_MAP_WRITE);
+
+ if (resample->funcs->width != resample->width) {
+ /* need to convert data format; allocate workspace */
+ if (!gst_audio_resample_workspace_realloc (&resample->tmp_out,
+ &resample->tmp_out_size, (resample->funcs->width / 8) * out_len *
+ resample->channels)) {
+ GST_ERROR_OBJECT (resample, "failed to allocate workspace");
+ return;
+ }
+
+ /* process */
+ err = resample->funcs->process (resample->state, NULL, &in_processed,
+ resample->tmp_out, &out_processed);
+
+ /* convert output format */
+ gst_audio_resample_convert_buffer (resample, resample->tmp_out,
+ data, out_processed, TRUE);
+ } else {
+ /* don't need to convert data format; process */
+ err = resample->funcs->process (resample->state, NULL, &in_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));
+ gst_buffer_unref (outbuf);
+ return;
+ }
+
+ /* time */
+ if (GST_CLOCK_TIME_IS_VALID (resample->t0)) {
+ GST_BUFFER_TIMESTAMP (outbuf) = resample->t0 +
+ gst_util_uint64_scale_int_round (resample->samples_out, GST_SECOND,
+ resample->outrate);
+ GST_BUFFER_DURATION (outbuf) = resample->t0 +
+ gst_util_uint64_scale_int_round (resample->samples_out + out_processed,
+ GST_SECOND, resample->outrate) - GST_BUFFER_TIMESTAMP (outbuf);
+ } else {
+ GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE;
+ GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
+ }
+ /* offset */
+ if (resample->out_offset0 != GST_BUFFER_OFFSET_NONE) {
+ GST_BUFFER_OFFSET (outbuf) = resample->out_offset0 + resample->samples_out;
+ GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET (outbuf) + out_processed;
+ } else {
+ GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
+ GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
+ }
+ /* move along */
+ resample->samples_out += out_processed;
+ resample->samples_in += history_len;
+
+ if (G_UNLIKELY (out_processed == 0 && in_len * den > num)) {
+ GST_WARNING_OBJECT (resample, "Failed to get drain, dropping buffer");
+ gst_buffer_unref (outbuf);
+ return;
+ }
+
+ 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, outsize,
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
+ GST_BUFFER_OFFSET_END (outbuf));
+
+ res = gst_pad_push (GST_BASE_TRANSFORM_SRC_PAD (resample), outbuf);
+
+ if (G_UNLIKELY (res != GST_FLOW_OK))
+ GST_WARNING_OBJECT (resample, "Failed to push drain: %s",
+ gst_flow_get_name (res));
+
+ return;
+}
+
+static gboolean
+gst_audio_resample_event (GstBaseTransform * base, GstEvent * event)
+{
+ GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_FLUSH_START:
- break;
case GST_EVENT_FLUSH_STOP:
- resample_input_flush (audioresample->resample);
- audioresample->ts_offset = -1;
- audioresample->next_ts = -1;
- audioresample->offset = -1;
+ gst_audio_resample_reset_state (resample);
+ if (resample->state)
+ resample->funcs->skip_zeros (resample->state);
+ resample->num_gap_samples = 0;
+ resample->num_nongap_samples = 0;
+ resample->t0 = GST_CLOCK_TIME_NONE;
+ resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
+ resample->out_offset0 = GST_BUFFER_OFFSET_NONE;
+ resample->samples_in = 0;
+ resample->samples_out = 0;
+ resample->need_discont = TRUE;
break;
case GST_EVENT_NEWSEGMENT:
- resample_input_pushthrough (audioresample->resample);
- audioresample_pushthrough (audioresample);
- audioresample->ts_offset = -1;
- audioresample->next_ts = -1;
- audioresample->offset = -1;
+ if (resample->state) {
+ guint latency = resample->funcs->get_input_latency (resample->state);
+ gst_audio_resample_push_drain (resample, latency);
+ }
+ gst_audio_resample_reset_state (resample);
+ if (resample->state)
+ resample->funcs->skip_zeros (resample->state);
+ resample->num_gap_samples = 0;
+ resample->num_nongap_samples = 0;
+ resample->t0 = GST_CLOCK_TIME_NONE;
+ resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
+ resample->out_offset0 = GST_BUFFER_OFFSET_NONE;
+ resample->samples_in = 0;
+ resample->samples_out = 0;
+ resample->need_discont = TRUE;
break;
case GST_EVENT_EOS:
- resample_input_eos (audioresample->resample);
- audioresample_pushthrough (audioresample);
+ if (resample->state) {
+ guint latency = resample->funcs->get_input_latency (resample->state);
+ gst_audio_resample_push_drain (resample, latency);
+ }
+ gst_audio_resample_reset_state (resample);
break;
default:
break;
}
+
return parent_class->event (base, event);
}
-static GstFlowReturn
-audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
-{
- int outsize;
- int outsamples;
- ResampleState *r;
-
- r = audioresample->resample;
-
- outsize = resample_get_output_size (r);
- GST_LOG_OBJECT (audioresample, "audioresample can give me %d bytes", outsize);
-
- /* protect against mem corruption */
- if (outsize > GST_BUFFER_SIZE (outbuf)) {
- GST_WARNING_OBJECT (audioresample,
- "overriding audioresample's outsize %d with outbuffer's size %d",
- outsize, GST_BUFFER_SIZE (outbuf));
- outsize = GST_BUFFER_SIZE (outbuf);
- }
- /* catch possibly wrong size differences */
- if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) {
- GST_WARNING_OBJECT (audioresample,
- "audioresample's outsize %d too far from outbuffer's size %d",
- outsize, GST_BUFFER_SIZE (outbuf));
- }
-
- outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize);
- outsamples = outsize / r->sample_size;
- GST_LOG_OBJECT (audioresample, "resample gave me %d bytes or %d samples",
- outsize, outsamples);
-
- GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
- GST_BUFFER_TIMESTAMP (outbuf) = audioresample->next_ts;
-
- if (audioresample->ts_offset != -1) {
- audioresample->offset += outsamples;
- audioresample->ts_offset += outsamples;
- audioresample->next_ts =
- gst_util_uint64_scale_int (audioresample->ts_offset, GST_SECOND,
- audioresample->o_rate);
- GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset;
-
- /* we calculate DURATION as the difference between "next" timestamp
- * and current timestamp so we ensure a contiguous stream, instead of
- * having rounding errors. */
- GST_BUFFER_DURATION (outbuf) = audioresample->next_ts -
- GST_BUFFER_TIMESTAMP (outbuf);
- } else {
- /* no valid offset know, we can still sortof calculate the duration though */
- GST_BUFFER_DURATION (outbuf) =
- gst_util_uint64_scale_int (outsamples, GST_SECOND,
- audioresample->o_rate);
- }
-
- /* check for possible mem corruption */
- if (outsize > GST_BUFFER_SIZE (outbuf)) {
- /* this is an error that when it happens, would need fixing in the
- * resample library; we told it we wanted only GST_BUFFER_SIZE (outbuf),
- * and it gave us more ! */
- GST_WARNING_OBJECT (audioresample,
- "audioresample, you memory corrupting bastard. "
- "you gave me outsize %d while my buffer was size %d",
- outsize, GST_BUFFER_SIZE (outbuf));
- return GST_FLOW_ERROR;
- }
- /* catch possibly wrong size differences */
- if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) {
- GST_WARNING_OBJECT (audioresample,
- "audioresample's written outsize %d too far from outbuffer's size %d",
- outsize, GST_BUFFER_SIZE (outbuf));
- }
- GST_BUFFER_SIZE (outbuf) = outsize;
-
- if (G_UNLIKELY (audioresample->need_discont)) {
- GST_DEBUG_OBJECT (audioresample,
- "marking this buffer with the DISCONT flag");
- GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
- audioresample->need_discont = FALSE;
- }
-
- GST_LOG_OBJECT (audioresample, "transformed to buffer of %d bytes, ts %"
- GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
- G_GINT64_FORMAT ", offset_end %" G_GINT64_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_check_discont (GstAudioResample * resample, GstBuffer * buf)
+{
+ guint64 offset;
+ guint64 delta;
+ /* is the incoming buffer a discontinuity? */
+ if (G_UNLIKELY (GST_BUFFER_IS_DISCONT (buf)))
+ return TRUE;
- return GST_FLOW_OK;
-}
+ /* no valid timestamps or offsets to compare --> no discontinuity */
+ if (G_UNLIKELY (!(GST_BUFFER_TIMESTAMP_IS_VALID (buf) &&
+ GST_CLOCK_TIME_IS_VALID (resample->t0))))
+ return FALSE;
-static gboolean
-audioresample_check_discont (GstAudioresample * audioresample,
- GstClockTime timestamp)
-{
- if (timestamp != GST_CLOCK_TIME_NONE &&
- audioresample->prev_ts != GST_CLOCK_TIME_NONE &&
- audioresample->prev_duration != GST_CLOCK_TIME_NONE &&
- timestamp != audioresample->prev_ts + audioresample->prev_duration) {
- /* Potentially a discontinuous buffer. However, it turns out that many
- * elements generate imperfect streams due to rounding errors, so we permit
- * a small error (up to one sample) without triggering a filter
- * flush/restart (if triggered incorrectly, this will be audible) */
- GstClockTimeDiff diff = timestamp -
- (audioresample->prev_ts + audioresample->prev_duration);
-
- if (ABS (diff) > GST_SECOND / audioresample->i_rate) {
- GST_WARNING_OBJECT (audioresample,
- "encountered timestamp discontinuity of %" G_GINT64_FORMAT, diff);
- return TRUE;
- }
- }
+ /* convert the inbound timestamp to an offset. */
+ offset =
+ gst_util_uint64_scale_int_round (GST_BUFFER_TIMESTAMP (buf) -
+ resample->t0, resample->inrate, GST_SECOND);
+
+ /* many elements generate imperfect streams due to rounding errors, so we
+ * permit a small error (up to one sample) without triggering a filter
+ * flush/restart (if triggered incorrectly, this will be audible) */
+ /* allow even up to more samples, since sink is not so strict anyway,
+ * so give that one a chance to handle this as configured */
+ delta = ABS ((gint64) (offset - resample->samples_in));
+ if (delta <= (resample->inrate >> 5))
+ return FALSE;
- return FALSE;
+ GST_WARNING_OBJECT (resample,
+ "encountered timestamp discontinuity of %" G_GUINT64_FORMAT " samples = %"
+ GST_TIME_FORMAT, delta,
+ GST_TIME_ARGS (gst_util_uint64_scale_int_round (delta, GST_SECOND,
+ resample->inrate)));
+ return TRUE;
}
static GstFlowReturn
-audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
+gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf,
GstBuffer * outbuf)
{
- GstAudioresample *audioresample;
- ResampleState *r;
- guchar *data, *datacopy;
- gulong size;
- GstClockTime timestamp;
+ 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_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);
+
+ in_processed = in_len;
+ out_processed = out_len;
+
+ if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
+ resample->num_nongap_samples = 0;
+ if (resample->num_gap_samples < filt_len) {
+ guint zeros_to_push;
+ if (in_len >= filt_len - resample->num_gap_samples)
+ zeros_to_push = filt_len - resample->num_gap_samples;
+ else
+ zeros_to_push = in_len;
+
+ gst_audio_resample_push_drain (resample, zeros_to_push);
+ in_len -= zeros_to_push;
+ resample->num_gap_samples += zeros_to_push;
+ }
- audioresample = GST_AUDIORESAMPLE (base);
- r = audioresample->resample;
+ {
+ guint num, den;
+ resample->funcs->get_ratio (resample->state, &num, &den);
+ if (resample->samples_in + in_len >= filt_len / 2)
+ out_processed =
+ gst_util_uint64_scale_int_ceil (resample->samples_in + in_len -
+ filt_len / 2, den, num) - resample->samples_out;
+ else
+ out_processed = 0;
+
+ 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;
+ }
+ } else { /* not a gap */
- data = GST_BUFFER_DATA (inbuf);
- size = GST_BUFFER_SIZE (inbuf);
- timestamp = GST_BUFFER_TIMESTAMP (inbuf);
+ gint err;
- GST_LOG_OBJECT (audioresample, "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 (timestamp),
- GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)),
- GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
+ if (resample->num_gap_samples > filt_len) {
+ /* push in enough zeros to restore the filter to the right offset */
+ guint num, den;
+ resample->funcs->get_ratio (resample->state, &num, &den);
+ gst_audio_resample_dump_drain (resample,
+ (resample->num_gap_samples - filt_len) % num);
+ }
+ resample->num_gap_samples = 0;
+ if (resample->num_nongap_samples < filt_len) {
+ resample->num_nongap_samples += in_len;
+ if (resample->num_nongap_samples > filt_len)
+ resample->num_nongap_samples = filt_len;
+ }
+
+ if (resample->funcs->width != resample->width) {
+ /* need to convert data format for processing; ensure we have enough
+ * workspace available */
+ if (!gst_audio_resample_workspace_realloc (&resample->tmp_in,
+ &resample->tmp_in_size, in_len * resample->channels *
+ (resample->funcs->width / 8)) ||
+ !gst_audio_resample_workspace_realloc (&resample->tmp_out,
+ &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, in_data,
+ resample->tmp_in, in_len, FALSE);
+
+ /* process */
+ err = resample->funcs->process (resample->state,
+ resample->tmp_in, &in_processed, resample->tmp_out, &out_processed);
+
+ /* convert output */
+ gst_audio_resample_convert_buffer (resample, resample->tmp_out,
+ out_data, out_processed, TRUE);
+ } else {
+ /* no format conversion required; process */
+ err = resample->funcs->process (resample->state,
+ in_data, &in_processed, out_data, &out_processed);
+ }
- /* check for timestamp discontinuities and flush/reset if needed */
- if (G_UNLIKELY (audioresample_check_discont (audioresample, timestamp))) {
- /* Flush internal samples */
- audioresample_pushthrough (audioresample);
- /* Inform downstream element about discontinuity */
- audioresample->need_discont = TRUE;
- /* We want to recalculate the offset */
- audioresample->ts_offset = -1;
- }
-
- if (audioresample->ts_offset == -1) {
- /* if we don't know the initial offset yet, calculate it based on the
- * input timestamp. */
- if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
- GstClockTime stime;
-
- /* offset used to calculate the timestamps. We use the sample offset for
- * this to make it more accurate. We want the first buffer to have the
- * same timestamp as the incoming timestamp. */
- audioresample->next_ts = timestamp;
- audioresample->ts_offset =
- gst_util_uint64_scale_int (timestamp, r->o_rate, GST_SECOND);
- /* offset used to set as the buffer offset, this offset is always
- * relative to the stream time, note that timestamp is not... */
- stime = (timestamp - base->segment.start) + base->segment.time;
- audioresample->offset =
- gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND);
+ 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;
}
}
- audioresample->prev_ts = timestamp;
- audioresample->prev_duration = GST_BUFFER_DURATION (inbuf);
- /* need to memdup, resample takes ownership. */
- datacopy = g_memdup (data, size);
- resample_add_input_data (r, datacopy, size, g_free, datacopy);
+ /* If we wrote more than allocated something is really wrong now and we
+ * should better abort immediately */
+ g_assert (out_len >= out_processed);
+
+ if (G_UNLIKELY (in_len != in_processed)) {
+ GST_WARNING_OBJECT (resample, "converted %d of %d input samples",
+ in_processed, in_len);
+ }
+
+ /* time */
+ if (GST_CLOCK_TIME_IS_VALID (resample->t0)) {
+ GST_BUFFER_TIMESTAMP (outbuf) = resample->t0 +
+ gst_util_uint64_scale_int_round (resample->samples_out, GST_SECOND,
+ resample->outrate);
+ GST_BUFFER_DURATION (outbuf) = resample->t0 +
+ gst_util_uint64_scale_int_round (resample->samples_out + out_processed,
+ GST_SECOND, resample->outrate) - GST_BUFFER_TIMESTAMP (outbuf);
+ } else {
+ GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE;
+ GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
+ }
+ /* offset */
+ if (resample->out_offset0 != GST_BUFFER_OFFSET_NONE) {
+ GST_BUFFER_OFFSET (outbuf) = resample->out_offset0 + resample->samples_out;
+ GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET (outbuf) + out_processed;
+ } else {
+ GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
+ GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
+ }
+ /* move along */
+ resample->samples_out += out_processed;
+ resample->samples_in += in_len;
+
+ 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, 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));
- return audioresample_do_output (audioresample, outbuf);
+ if (out_processed == 0) {
+ GST_DEBUG_OBJECT (resample, "buffer dropped");
+ return GST_BASE_TRANSFORM_FLOW_DROPPED;
+ }
+ return GST_FLOW_OK;
}
-/* push remaining data in the buffers out */
static GstFlowReturn
-audioresample_pushthrough (GstAudioresample * audioresample)
+gst_audio_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
+ GstBuffer * outbuf)
{
- int outsize;
- ResampleState *r;
- GstBuffer *outbuf;
- GstFlowReturn res = GST_FLOW_OK;
- GstBaseTransform *trans;
+ GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
+ GstFlowReturn ret;
+
+ if (resample->state == NULL) {
+ if (G_UNLIKELY (!(resample->state =
+ gst_audio_resample_init_state (resample, resample->width,
+ resample->channels, resample->inrate, resample->outrate,
+ resample->quality, resample->fp))))
+ return GST_FLOW_ERROR;
+
+ resample->funcs =
+ gst_audio_resample_get_funcs (resample->width, resample->fp);
+ }
- r = audioresample->resample;
+ 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,
+ 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));
- outsize = resample_get_output_size (r);
- if (outsize == 0) {
- GST_DEBUG_OBJECT (audioresample, "no internal buffers needing flush");
- goto done;
+ /* check for timestamp discontinuities; flush/reset if needed, and set
+ * flag to resync timestamp and offset counters and send event
+ * downstream */
+ if (G_UNLIKELY (gst_audio_resample_check_discont (resample, inbuf))) {
+ gst_audio_resample_reset_state (resample);
+ resample->need_discont = TRUE;
}
- trans = GST_BASE_TRANSFORM (audioresample);
-
- res = gst_pad_alloc_buffer (trans->srcpad, GST_BUFFER_OFFSET_NONE, outsize,
- GST_PAD_CAPS (trans->srcpad), &outbuf);
- if (G_UNLIKELY (res != GST_FLOW_OK)) {
- GST_WARNING_OBJECT (audioresample, "failed allocating buffer of %d bytes",
- outsize);
- goto done;
+ /* handle discontinuity */
+ if (G_UNLIKELY (resample->need_discont)) {
+ resample->funcs->skip_zeros (resample->state);
+ resample->num_gap_samples = 0;
+ resample->num_nongap_samples = 0;
+ /* reset */
+ resample->samples_in = 0;
+ resample->samples_out = 0;
+ GST_DEBUG_OBJECT (resample, "found discontinuity; resyncing");
+ /* resync the timestamp and offset counters if possible */
+ if (GST_BUFFER_TIMESTAMP_IS_VALID (inbuf)) {
+ resample->t0 = GST_BUFFER_TIMESTAMP (inbuf);
+ } else {
+ GST_DEBUG_OBJECT (resample, "... but new timestamp is invalid");
+ resample->t0 = GST_CLOCK_TIME_NONE;
+ }
+ if (GST_BUFFER_OFFSET_IS_VALID (inbuf)) {
+ resample->in_offset0 = GST_BUFFER_OFFSET (inbuf);
+ resample->out_offset0 =
+ gst_util_uint64_scale_int_round (resample->in_offset0,
+ resample->outrate, resample->inrate);
+ } else {
+ GST_DEBUG_OBJECT (resample, "... but new offset is invalid");
+ resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
+ resample->out_offset0 = GST_BUFFER_OFFSET_NONE;
+ }
+ /* set DISCONT flag on output buffer */
+ GST_DEBUG_OBJECT (resample, "marking this buffer with the DISCONT flag");
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ resample->need_discont = FALSE;
}
- res = audioresample_do_output (audioresample, outbuf);
- if (G_UNLIKELY (res != GST_FLOW_OK))
- goto done;
+ ret = gst_audio_resample_process (resample, inbuf, outbuf);
+ if (G_UNLIKELY (ret != GST_FLOW_OK))
+ return ret;
- res = gst_pad_push (trans->srcpad, outbuf);
+ GST_DEBUG_OBJECT (resample, "input = samples [%" G_GUINT64_FORMAT ", %"
+ G_GUINT64_FORMAT ") = [%" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT
+ ") ns; output = samples [%" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT
+ ") = [%" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT ") ns",
+ GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf),
+ GST_BUFFER_TIMESTAMP (inbuf), GST_BUFFER_TIMESTAMP (inbuf) +
+ GST_BUFFER_DURATION (inbuf), GST_BUFFER_OFFSET (outbuf),
+ GST_BUFFER_OFFSET_END (outbuf), GST_BUFFER_TIMESTAMP (outbuf),
+ GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf));
-done:
- return res;
+ return GST_FLOW_OK;
}
static gboolean
-audioresample_query (GstPad * pad, GstQuery * query)
+gst_audio_resample_query (GstPad * pad, GstQuery * query)
{
- GstAudioresample *audioresample =
- GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
- GstBaseTransform *trans = GST_BASE_TRANSFORM (audioresample);
+ GstAudioResample *resample = GST_AUDIO_RESAMPLE (gst_pad_get_parent (pad));
+ GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
gboolean res = TRUE;
switch (GST_QUERY_TYPE (query)) {
gboolean live;
guint64 latency;
GstPad *peer;
- gint rate = audioresample->i_rate;
- gint resampler_latency = audioresample->filter_length / 2;
+ gint rate = resample->inrate;
+ gint resampler_latency;
+
+ if (resample->state)
+ resampler_latency =
+ resample->funcs->get_input_latency (resample->state);
+ else
+ resampler_latency = 0;
if (gst_base_transform_is_passthrough (trans))
resampler_latency = 0;
- if ((peer = gst_pad_get_peer (trans->sinkpad))) {
+ 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);
- GST_DEBUG ("Peer latency: min %"
+ 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 (resampler_latency, GST_SECOND, rate);
+ latency = gst_util_uint64_scale_round (resampler_latency,
+ GST_SECOND, rate);
else
latency = 0;
- GST_DEBUG ("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 (max != GST_CLOCK_TIME_NONE)
+ if (GST_CLOCK_TIME_IS_VALID (max))
max += latency;
- GST_DEBUG ("Calculated total latency : min %"
+ GST_DEBUG_OBJECT (resample, "Calculated total latency : min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
res = gst_pad_query_default (pad, query);
break;
}
- gst_object_unref (audioresample);
+ gst_object_unref (resample);
return res;
}
static const GstQueryType *
-audioresample_query_type (GstPad * pad)
+gst_audio_resample_query_type (GstPad * pad)
{
static const GstQueryType types[] = {
GST_QUERY_LATENCY,
}
static void
-gst_audioresample_set_property (GObject * object, guint prop_id,
+gst_audio_resample_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
- GstAudioresample *audioresample;
+ GstAudioResample *resample;
- audioresample = GST_AUDIORESAMPLE (object);
+ resample = GST_AUDIO_RESAMPLE (object);
switch (prop_id) {
- case PROP_FILTERLEN:
- audioresample->filter_length = g_value_get_int (value);
- GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d",
- audioresample->filter_length);
- if (audioresample->resample) {
- resample_set_filter_length (audioresample->resample,
- audioresample->filter_length);
- gst_element_post_message (GST_ELEMENT (audioresample),
- gst_message_new_latency (GST_OBJECT (audioresample)));
- }
+ case PROP_QUALITY:
+ GST_BASE_TRANSFORM_LOCK (resample);
+ resample->quality = g_value_get_int (value);
+ GST_DEBUG_OBJECT (resample, "new quality %d", resample->quality);
+
+ gst_audio_resample_update_state (resample, resample->width,
+ resample->channels, resample->inrate, resample->outrate,
+ resample->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)
+ resample->quality = 0;
+ else if (filter_length <= 16)
+ resample->quality = 1;
+ else if (filter_length <= 32)
+ resample->quality = 2;
+ else if (filter_length <= 48)
+ resample->quality = 3;
+ else if (filter_length <= 64)
+ resample->quality = 4;
+ else if (filter_length <= 80)
+ resample->quality = 5;
+ else if (filter_length <= 96)
+ resample->quality = 6;
+ else if (filter_length <= 128)
+ resample->quality = 7;
+ else if (filter_length <= 160)
+ resample->quality = 8;
+ else if (filter_length <= 192)
+ resample->quality = 9;
+ else
+ resample->quality = 10;
+
+ GST_DEBUG_OBJECT (resample, "new quality %d", resample->quality);
+
+ gst_audio_resample_update_state (resample, resample->width,
+ resample->channels, resample->inrate, resample->outrate,
+ resample->quality, resample->fp);
+ GST_BASE_TRANSFORM_UNLOCK (resample);
break;
+ }
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
static void
-gst_audioresample_get_property (GObject * object, guint prop_id,
+gst_audio_resample_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
- GstAudioresample *audioresample;
+ GstAudioResample *resample;
- audioresample = GST_AUDIORESAMPLE (object);
+ resample = GST_AUDIO_RESAMPLE (object);
switch (prop_id) {
- case PROP_FILTERLEN:
- g_value_set_int (value, audioresample->filter_length);
+ 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);
}
}
+/* FIXME: should have a benchmark fallback for the case where orc is disabled */
+#if defined(AUDIORESAMPLE_FORMAT_AUTO) && !defined(DISABLE_ORC)
+
+#define BENCHMARK_SIZE 512
+
+static gboolean
+_benchmark_int_float (SpeexResamplerState * st)
+{
+ gint16 in[BENCHMARK_SIZE] = { 0, }, out[BENCHMARK_SIZE / 2];
+ gfloat in_tmp[BENCHMARK_SIZE], out_tmp[BENCHMARK_SIZE / 2];
+ gint i;
+ guint32 inlen = BENCHMARK_SIZE, outlen = BENCHMARK_SIZE / 2;
+
+ for (i = 0; i < BENCHMARK_SIZE; i++) {
+ gfloat tmp = in[i];
+ in_tmp[i] = tmp / G_MAXINT16;
+ }
+
+ resample_float_resampler_process_interleaved_float (st,
+ (const guint8 *) in_tmp, &inlen, (guint8 *) out_tmp, &outlen);
+
+ if (outlen == 0) {
+ GST_ERROR ("Failed to use float resampler");
+ return FALSE;
+ }
+
+ for (i = 0; i < outlen; i++) {
+ gfloat tmp = out_tmp[i];
+ out[i] = CLAMP (tmp * G_MAXINT16 + 0.5, G_MININT16, G_MAXINT16);
+ }
+
+ return TRUE;
+}
+
+static gboolean
+_benchmark_int_int (SpeexResamplerState * st)
+{
+ gint16 in[BENCHMARK_SIZE] = { 0, }, out[BENCHMARK_SIZE / 2];
+ guint32 inlen = BENCHMARK_SIZE, outlen = BENCHMARK_SIZE / 2;
+
+ resample_int_resampler_process_interleaved_int (st, (const guint8 *) in,
+ &inlen, (guint8 *) out, &outlen);
+
+ if (outlen == 0) {
+ GST_ERROR ("Failed to use int resampler");
+ return FALSE;
+ }
+
+ return TRUE;
+}
+
+static gboolean
+_benchmark_integer_resampling (void)
+{
+ OrcProfile a, b;
+ gdouble av, bv;
+ SpeexResamplerState *sta, *stb;
+ int i;
+
+ orc_profile_init (&a);
+ orc_profile_init (&b);
+
+ sta = resample_float_resampler_init (1, 48000, 24000, 4, NULL);
+ if (sta == NULL) {
+ GST_ERROR ("Failed to create float resampler state");
+ return FALSE;
+ }
+
+ stb = resample_int_resampler_init (1, 48000, 24000, 4, NULL);
+ if (stb == NULL) {
+ resample_float_resampler_destroy (sta);
+ GST_ERROR ("Failed to create int resampler state");
+ return FALSE;
+ }
+
+ /* Benchmark */
+ for (i = 0; i < 10; i++) {
+ orc_profile_start (&a);
+ if (!_benchmark_int_float (sta))
+ goto error;
+ orc_profile_stop (&a);
+ }
+
+ /* Benchmark */
+ for (i = 0; i < 10; i++) {
+ orc_profile_start (&b);
+ if (!_benchmark_int_int (stb))
+ goto error;
+ orc_profile_stop (&b);
+ }
+
+ /* Handle results */
+ orc_profile_get_ave_std (&a, &av, NULL);
+ orc_profile_get_ave_std (&b, &bv, NULL);
+
+ /* Remember benchmark result in global variable */
+ gst_audio_resample_use_int = (av > bv);
+ resample_float_resampler_destroy (sta);
+ resample_int_resampler_destroy (stb);
+
+ if (av > bv)
+ GST_INFO ("Using integer resampler if appropiate: %lf < %lf", bv, av);
+ else
+ GST_INFO ("Using float resampler for everything: %lf <= %lf", av, bv);
+
+ return TRUE;
+
+error:
+ resample_float_resampler_destroy (sta);
+ resample_int_resampler_destroy (stb);
+
+ return FALSE;
+}
+#endif /* defined(AUDIORESAMPLE_FORMAT_AUTO) && !defined(DISABLE_ORC) */
static gboolean
plugin_init (GstPlugin * plugin)
{
- resample_init ();
+ GST_DEBUG_CATEGORY_INIT (audio_resample_debug, "audioresample", 0,
+ "audio resampling element");
+
+#if defined(AUDIORESAMPLE_FORMAT_AUTO) && !defined(DISABLE_ORC)
+ if (!_benchmark_integer_resampling ())
+ return FALSE;
+#else
+ GST_WARNING ("Orc disabled, can't benchmark int vs. float resampler");
+ {
+ GST_DEBUG_CATEGORY_STATIC (GST_CAT_PERFORMANCE);
+ GST_DEBUG_CATEGORY_GET (GST_CAT_PERFORMANCE, "GST_PERFORMANCE");
+ GST_CAT_WARNING (GST_CAT_PERFORMANCE, "orc disabled, no benchmarking done");
+ }
+#endif
if (!gst_element_register (plugin, "audioresample", GST_RANK_PRIMARY,
- GST_TYPE_AUDIORESAMPLE)) {
+ GST_TYPE_AUDIO_RESAMPLE)) {
return FALSE;
}
projects / platform / upstream / gstreamer.git / blobdiff