diff options
Diffstat (limited to 'gst/audioresample/gstaudioresample.c')
| -rw-r--r-- | gst/audioresample/gstaudioresample.c | 1479 |
1 files changed, 0 insertions, 1479 deletions
diff --git a/gst/audioresample/gstaudioresample.c b/gst/audioresample/gstaudioresample.c deleted file mode 100644 index 6336db57..00000000 --- a/gst/audioresample/gstaudioresample.c +++ /dev/null @@ -1,1479 +0,0 @@ -/* 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 - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -/** - * SECTION:element-audioresample - * - * 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. - * To create the Ogg/Vorbis file refer to the documentation of vorbisenc. - * </refsect2> - */ - -/* TODO: - * - Enable SSE/ARM optimizations and select at runtime - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include <string.h> -#include <math.h> - -#include "gstaudioresample.h" -#include <gst/gstutils.h> -#include <gst/audio/audio.h> -#include <gst/base/gstbasetransform.h> - -#if defined AUDIORESAMPLE_FORMAT_AUTO -#define OIL_ENABLE_UNSTABLE_API -#include <liboil/liboilprofile.h> -#include <liboil/liboil.h> -#endif - -GST_DEBUG_CATEGORY (audio_resample_debug); -#define GST_CAT_DEFAULT audio_resample_debug - -enum -{ - PROP_0, - PROP_QUALITY, - PROP_FILTER_LENGTH -}; - -#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" \ -) - -/* 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_audio_resample_src_template = -GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS); - -static void gst_audio_resample_set_property (GObject * object, - guint prop_id, const GValue * value, GParamSpec * pspec); -static void gst_audio_resample_get_property (GObject * object, - guint prop_id, GValue * value, GParamSpec * pspec); - -/* vmethods */ -static gboolean gst_audio_resample_get_unit_size (GstBaseTransform * base, - GstCaps * caps, guint * size); -static GstCaps *gst_audio_resample_transform_caps (GstBaseTransform * base, - GstPadDirection direction, GstCaps * caps); -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); -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, - 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_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>"); -} - -static void -gst_audio_resample_class_init (GstAudioResampleClass * klass) -{ - GObjectClass *gobject_class = (GObjectClass *) klass; - - 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_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)); - - /* 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)); - - GST_BASE_TRANSFORM_CLASS (klass)->start = - GST_DEBUG_FUNCPTR (gst_audio_resample_start); - GST_BASE_TRANSFORM_CLASS (klass)->stop = - GST_DEBUG_FUNCPTR (gst_audio_resample_stop); - GST_BASE_TRANSFORM_CLASS (klass)->transform_size = - GST_DEBUG_FUNCPTR (gst_audio_resample_transform_size); - GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size = - GST_DEBUG_FUNCPTR (gst_audio_resample_get_unit_size); - GST_BASE_TRANSFORM_CLASS (klass)->transform_caps = - GST_DEBUG_FUNCPTR (gst_audio_resample_transform_caps); - GST_BASE_TRANSFORM_CLASS (klass)->fixate_caps = - GST_DEBUG_FUNCPTR (gst_audio_resample_fixate_caps); - GST_BASE_TRANSFORM_CLASS (klass)->set_caps = - 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)->passthrough_on_same_caps = TRUE; -} - -static void -gst_audio_resample_init (GstAudioResample * resample, - GstAudioResampleClass * klass) -{ - GstBaseTransform *trans = GST_BASE_TRANSFORM (resample); - - resample->quality = SPEEX_RESAMPLER_QUALITY_DEFAULT; - - 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 -gst_audio_resample_start (GstBaseTransform * base) -{ - GstAudioResample *resample = GST_AUDIO_RESAMPLE (base); - - resample->need_discont = TRUE; - - resample->t0 = GST_CLOCK_TIME_NONE; - resample->in_offset0 = GST_BUFFER_OFFSET_NONE; - resample->out_offset0 = GST_BUFFER_OFFSET_NONE; - resample->next_in_offset = GST_BUFFER_OFFSET_NONE; - resample->next_out_offset = GST_BUFFER_OFFSET_NONE; - - resample->tmp_in = NULL; - resample->tmp_in_size = 0; - resample->tmp_out = NULL; - resample->tmp_out_size = 0; - - return TRUE; -} - -static gboolean -gst_audio_resample_stop (GstBaseTransform * base) -{ - GstAudioResample *resample = GST_AUDIO_RESAMPLE (base); - - if (resample->state) { - resample->funcs->destroy (resample->state); - resample->state = 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 -gst_audio_resample_get_unit_size (GstBaseTransform * base, GstCaps * caps, - guint * size) -{ - gint width, channels; - GstStructure *structure; - gboolean ret; - - 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); - - if (G_UNLIKELY (!ret)) - return FALSE; - - *size = (width / 8) * channels; - - return TRUE; -} - -static GstCaps * -gst_audio_resample_transform_caps (GstBaseTransform * base, - GstPadDirection direction, GstCaps * caps) -{ - const GValue *val; - GstStructure *s; - GstCaps *res; - - /* 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 */ - 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 -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 (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 -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 (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); - 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) -{ - while (b != 0) { - int temp = a; - - a = b; - b = temp % b; - } - - return ABS (a); -} - -static gboolean -gst_audio_resample_transform_size (GstBaseTransform * base, - GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps, - guint * 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"); - - /* 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. 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. Round down the input size */ - *othersize = gst_util_uint64_scale_int (size, ratio_num, ratio_den); - *othersize *= bytes_per_samp; - } - - GST_LOG_OBJECT (base, "transformed size %d to %d", size * bytes_per_samp, - *othersize); - - return ret; -} - -static gboolean -gst_audio_resample_set_caps (GstBaseTransform * base, GstCaps * incaps, - GstCaps * outcaps) -{ - gboolean ret; - gint width = 0, inrate = 0, outrate = 0, channels = 0; - gboolean fp; - GstAudioResample *resample = GST_AUDIO_RESAMPLE (base); - - 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 (G_UNLIKELY (!ret)) - return FALSE; - - 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 (&resample->sinkcaps, incaps); - gst_caps_replace (&resample->srccaps, outcaps); - - return TRUE; -} - -#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) -{ - 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 (); - } - } -} - -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; -} - -static void -gst_audio_resample_push_drain (GstAudioResample * resample) -{ - GstBuffer *outbuf; - GstFlowReturn res; - gint outsize; - guint history_len, out_len, out_processed; - gint err; - guint num, den; - - if (!resample->state) - return; - - /* 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); - - history_len = resample->funcs->get_input_latency (resample->state); - out_len = out_processed = - gst_util_uint64_scale_int_ceil (history_len, den, num); - outsize = out_len * resample->channels * (resample->width / 8); - - 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; - } - - 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, &history_len, - resample->tmp_out, &out_processed); - - /* convert output format */ - gst_audio_resample_convert_buffer (resample, resample->tmp_out, - GST_BUFFER_DATA (outbuf), out_processed, TRUE); - } else { - /* don't need to convert data format; process */ - err = resample->funcs->process (resample->state, NULL, &history_len, - GST_BUFFER_DATA (outbuf), &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); - - 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; - } - - if (G_UNLIKELY (out_processed == 0)) { - GST_WARNING_OBJECT (resample, "Failed to get drain, dropping buffer"); - gst_buffer_unref (outbuf); - return; - } - - if (GST_CLOCK_TIME_IS_VALID (resample->t0)) { - GST_BUFFER_OFFSET (outbuf) = resample->next_out_offset; - GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET (outbuf) + out_processed; - GST_BUFFER_TIMESTAMP (outbuf) = resample->t0 + - gst_util_uint64_scale_int_round (GST_BUFFER_OFFSET (outbuf) - - resample->out_offset0, GST_SECOND, resample->outrate); - GST_BUFFER_DURATION (outbuf) = resample->t0 + - gst_util_uint64_scale_int_round (GST_BUFFER_OFFSET_END (outbuf) - - resample->out_offset0, GST_SECOND, resample->outrate) - - GST_BUFFER_TIMESTAMP (outbuf); - resample->next_out_offset += out_processed; - resample->next_in_offset += 0; - } else { - GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE; - GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE; - GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE; - GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE; - } - - 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), - 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_STOP: - gst_audio_resample_reset_state (resample); - resample->t0 = GST_CLOCK_TIME_NONE; - resample->in_offset0 = GST_BUFFER_OFFSET_NONE; - resample->out_offset0 = GST_BUFFER_OFFSET_NONE; - resample->next_in_offset = GST_BUFFER_OFFSET_NONE; - resample->next_out_offset = GST_BUFFER_OFFSET_NONE; - resample->need_discont = TRUE; - break; - case GST_EVENT_NEWSEGMENT: - gst_audio_resample_push_drain (resample); - gst_audio_resample_reset_state (resample); - resample->t0 = GST_CLOCK_TIME_NONE; - resample->in_offset0 = GST_BUFFER_OFFSET_NONE; - resample->out_offset0 = GST_BUFFER_OFFSET_NONE; - resample->next_in_offset = GST_BUFFER_OFFSET_NONE; - resample->next_out_offset = GST_BUFFER_OFFSET_NONE; - resample->need_discont = TRUE; - break; - case GST_EVENT_EOS: - gst_audio_resample_push_drain (resample); - gst_audio_resample_reset_state (resample); - break; - default: - break; - } - - return parent_class->event (base, event); -} - -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; - - /* 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) && - resample->in_offset0 != GST_BUFFER_OFFSET_NONE && - resample->next_in_offset != GST_BUFFER_OFFSET_NONE))) - return FALSE; - - /* convert the inbound timestamp to an offset. */ - offset = - resample->in_offset0 + - 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) */ - delta = ABS ((gint64) (offset - resample->next_in_offset)); - if (delta <= 1) - 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 -gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf, - GstBuffer * outbuf) -{ - guint32 in_len, in_processed; - guint32 out_len, out_processed; - gint err; - - in_len = GST_BUFFER_SIZE (inbuf) / resample->channels; - out_len = GST_BUFFER_SIZE (outbuf) / resample->channels; - - in_len /= (resample->width / 8); - out_len /= (resample->width / 8); - - in_processed = in_len; - out_processed = out_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"); - return GST_FLOW_ERROR; - } - - /* convert input */ - gst_audio_resample_convert_buffer (resample, GST_BUFFER_DATA (inbuf), - 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, - GST_BUFFER_DATA (outbuf), 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); - } - - /* 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 (err != RESAMPLER_ERR_SUCCESS)) { - GST_ERROR_OBJECT (resample, "Failed to convert data: %s", - resample->funcs->strerror (err)); - return GST_FLOW_ERROR; - } - - if (G_UNLIKELY (in_len != in_processed)) { - GST_WARNING_OBJECT (resample, "converted %d of %d input samples", - in_processed, in_len); - } - - if (GST_CLOCK_TIME_IS_VALID (resample->t0)) { - GST_BUFFER_OFFSET (outbuf) = resample->next_out_offset; - GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET (outbuf) + out_processed; - GST_BUFFER_TIMESTAMP (outbuf) = resample->t0 + - gst_util_uint64_scale_int_round (GST_BUFFER_OFFSET (outbuf) - - resample->out_offset0, GST_SECOND, resample->outrate); - GST_BUFFER_DURATION (outbuf) = resample->t0 + - gst_util_uint64_scale_int_round (GST_BUFFER_OFFSET_END (outbuf) - - resample->out_offset0, GST_SECOND, resample->outrate) - - GST_BUFFER_TIMESTAMP (outbuf); - resample->next_out_offset += out_processed; - resample->next_in_offset += in_len; - } else { - GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE; - GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE; - GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE; - GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE; - } - - GST_BUFFER_SIZE (outbuf) = - out_processed * resample->channels * (resample->width / 8); - - 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), - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), - GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), - GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf)); - - if (out_processed == 0) { - GST_DEBUG_OBJECT (resample, "buffer dropped"); - return GST_BASE_TRANSFORM_FLOW_DROPPED; - } - return GST_FLOW_OK; -} - -static GstFlowReturn -gst_audio_resample_transform (GstBaseTransform * base, GstBuffer * inbuf, - GstBuffer * outbuf) -{ - GstAudioResample *resample = GST_AUDIO_RESAMPLE (base); - gulong size; - 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); - } - - 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_TIME_ARGS (GST_BUFFER_DURATION (inbuf)), - GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf)); - - /* 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; - } - - /* handle discontinuity */ - if (G_UNLIKELY (resample->need_discont)) { - /* resync the timestamp and offset counters if possible */ - if (GST_BUFFER_TIMESTAMP_IS_VALID (inbuf) && - GST_BUFFER_OFFSET_IS_VALID (inbuf)) { - resample->t0 = GST_BUFFER_TIMESTAMP (inbuf); - resample->in_offset0 = GST_BUFFER_OFFSET (inbuf); - resample->out_offset0 = - gst_util_uint64_scale_int_round (resample->in_offset0, - resample->outrate, resample->inrate); - resample->next_in_offset = resample->in_offset0; - resample->next_out_offset = resample->out_offset0; - } else { - GST_DEBUG_OBJECT (resample, "found discontinuity but timestamp and/or " - "offset is invalid, cannot sync output timestamp and offset counter"); - resample->t0 = GST_CLOCK_TIME_NONE; - resample->in_offset0 = GST_BUFFER_OFFSET_NONE; - resample->out_offset0 = GST_BUFFER_OFFSET_NONE; - resample->next_in_offset = GST_BUFFER_OFFSET_NONE; - resample->next_out_offset = 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; - } - - ret = gst_audio_resample_process (resample, inbuf, outbuf); - if (G_UNLIKELY (ret != GST_FLOW_OK)) - return ret; - - 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)); - - return GST_FLOW_OK; -} - -static gboolean -gst_audio_resample_query (GstPad * pad, GstQuery * query) -{ - GstAudioResample *resample = GST_AUDIO_RESAMPLE (gst_pad_get_parent (pad)); - GstBaseTransform *trans = GST_BASE_TRANSFORM (resample); - gboolean res = TRUE; - - switch (GST_QUERY_TYPE (query)) { - case GST_QUERY_LATENCY: - { - GstClockTime min, max; - gboolean live; - guint64 latency; - GstPad *peer; - 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 (GST_BASE_TRANSFORM_SINK_PAD (trans)))) { - if ((res = gst_pad_query (peer, 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)); - - /* 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)); - - 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_query_set_latency (query, live, min, max); - } - gst_object_unref (peer); - } - break; - } - default: - res = gst_pad_query_default (pad, 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) -{ - GstAudioResample *resample; - - resample = GST_AUDIO_RESAMPLE (object); - - switch (prop_id) { - 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_audio_resample_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstAudioResample *resample; - - resample = GST_AUDIO_RESAMPLE (object); - - switch (prop_id) { - 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; - } -} - -#if defined AUDIORESAMPLE_FORMAT_AUTO -#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) -{ - OilProfile a, b; - gdouble av, bv; - SpeexResamplerState *sta, *stb; - int i; - - oil_profile_init (&a); - oil_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++) { - oil_profile_start (&a); - if (!_benchmark_int_float (sta)) - goto error; - oil_profile_stop (&a); - } - - /* Benchmark */ - for (i = 0; i < 10; i++) { - oil_profile_start (&b); - if (!_benchmark_int_int (stb)) - goto error; - oil_profile_stop (&b); - } - - /* Handle results */ - oil_profile_get_ave_std (&a, &av, NULL); - oil_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 - -static gboolean -plugin_init (GstPlugin * plugin) -{ - GST_DEBUG_CATEGORY_INIT (audio_resample_debug, "audioresample", 0, - "audio resampling element"); -#if defined AUDIORESAMPLE_FORMAT_AUTO - oil_init (); - - if (!_benchmark_integer_resampling ()) - return FALSE; -#endif - - if (!gst_element_register (plugin, "audioresample", GST_RANK_PRIMARY, - GST_TYPE_AUDIO_RESAMPLE)) { - return FALSE; - } - - return TRUE; -} - -GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, - GST_VERSION_MINOR, - "audioresample", - "Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, - GST_PACKAGE_ORIGIN); |