/* GStreamer * Copyright (C) 1999 Erik Walthinsen * Copyright (C) 2003,2004 David A. Schleef * Copyright (C) 2007-2008 Sebastian Dröge * * 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. * * By default, the resampler uses a reduced sinc table, with cubic interpolation filling in * the gaps. This ensures that the table does not become too big. However, the interpolation * increases the CPU usage considerably. As an alternative, a full sinc table can be used. * Doing so can drastically reduce CPU usage (4x faster with 44.1 -> 48 kHz conversions for * example), at the cost of increased memory consumption, plus the sinc table takes longer * to initialize when the element is created. A third mode exists, which uses the full table * unless said table would become too large, in which case the interpolated one is used instead. * * * Example launch line * |[ * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! audio/x-raw, rate=8000 ! alsasink * ]| Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa. * To create the Ogg/Vorbis file refer to the documentation of vorbisenc. * */ /* TODO: * - Enable SSE/ARM optimizations and select at runtime */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include "gstaudioresample.h" #include #include #include #ifndef DISABLE_ORC #include #include #include #endif GST_DEBUG_CATEGORY (audio_resample_debug); #define GST_CAT_DEFAULT audio_resample_debug #if !defined(AUDIORESAMPLE_FORMAT_AUTO) || defined(DISABLE_ORC) GST_DEBUG_CATEGORY_STATIC (GST_CAT_PERFORMANCE); #endif #define GST_TYPE_SPEEX_RESAMPLER_SINC_FILTER_MODE (speex_resampler_sinc_filter_mode_get_type ()) enum { PROP_0, PROP_QUALITY, PROP_SINC_FILTER_MODE, PROP_SINC_FILTER_AUTO_THRESHOLD }; #if G_BYTE_ORDER == G_LITTLE_ENDIAN #define SUPPORTED_CAPS \ GST_AUDIO_CAPS_MAKE ("{ F32LE, F64LE, S32LE, S24LE, S16LE, S8 }") \ ", layout = (string) { interleaved, non-interleaved }" #else #define SUPPORTED_CAPS \ GST_AUDIO_CAPS_MAKE ("{ F32BE, F64BE, S32BE, S24BE, S16BE, S8 }") \ ", layout = (string) { interleaved, non-interleaved }" #endif /* If TRUE integer arithmetic resampling is faster and will be used if appropriate */ #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, GST_STATIC_CAPS (SUPPORTED_CAPS)); static GstStaticPadTemplate gst_audio_resample_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (SUPPORTED_CAPS)); static void gst_audio_resample_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_audio_resample_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GType speex_resampler_sinc_filter_mode_get_type (void); /* vmethods */ 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, GstCaps * filter); static GstCaps *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, gsize insize, GstCaps * outcaps, gsize * outsize); static gboolean gst_audio_resample_set_caps (GstBaseTransform * base, GstCaps * incaps, GstCaps * outcaps); static GstFlowReturn gst_audio_resample_transform (GstBaseTransform * base, GstBuffer * inbuf, GstBuffer * outbuf); static gboolean gst_audio_resample_sink_event (GstBaseTransform * base, GstEvent * event); static gboolean gst_audio_resample_start (GstBaseTransform * base); static gboolean gst_audio_resample_stop (GstBaseTransform * base); static gboolean gst_audio_resample_query (GstPad * pad, GstObject * parent, GstQuery * query); #define gst_audio_resample_parent_class parent_class G_DEFINE_TYPE (GstAudioResample, gst_audio_resample, GST_TYPE_BASE_TRANSFORM); static void gst_audio_resample_class_init (GstAudioResampleClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstElementClass *gstelement_class = (GstElementClass *) klass; gobject_class->set_property = gst_audio_resample_set_property; gobject_class->get_property = gst_audio_resample_get_property; 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)); g_object_class_install_property (gobject_class, PROP_SINC_FILTER_MODE, g_param_spec_enum ("sinc-filter-mode", "Sinc filter table mode", "What sinc filter table mode to use", GST_TYPE_SPEEX_RESAMPLER_SINC_FILTER_MODE, SPEEX_RESAMPLER_SINC_FILTER_DEFAULT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_SINC_FILTER_AUTO_THRESHOLD, g_param_spec_uint ("sinc-filter-auto-threshold", "Sinc filter auto mode threshold", "Memory usage threshold to use if sinc filter mode is AUTO, given in bytes", 0, G_MAXUINT, SPEEX_RESAMPLER_SINC_FILTER_AUTO_THRESHOLD_DEFAULT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_audio_resample_src_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_audio_resample_sink_template)); gst_element_class_set_static_metadata (gstelement_class, "Audio resampler", "Filter/Converter/Audio", "Resamples audio", "Sebastian Dröge "); 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)->sink_event = GST_DEBUG_FUNCPTR (gst_audio_resample_sink_event); GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE; } static void gst_audio_resample_init (GstAudioResample * resample) { GstBaseTransform *trans = GST_BASE_TRANSFORM (resample); resample->quality = SPEEX_RESAMPLER_QUALITY_DEFAULT; resample->sinc_filter_mode = SPEEX_RESAMPLER_SINC_FILTER_DEFAULT; resample->sinc_filter_auto_threshold = SPEEX_RESAMPLER_SINC_FILTER_AUTO_THRESHOLD_DEFAULT; gst_base_transform_set_gap_aware (trans, TRUE); gst_pad_set_query_function (trans->srcpad, gst_audio_resample_query); } /* vmethods */ static gboolean gst_audio_resample_start (GstBaseTransform * base) { GstAudioResample *resample = GST_AUDIO_RESAMPLE (base); resample->need_discont = TRUE; 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->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; return TRUE; } static gboolean gst_audio_resample_get_unit_size (GstBaseTransform * base, GstCaps * caps, gsize * size) { GstAudioInfo info; if (!gst_audio_info_from_caps (&info, caps)) goto invalid_caps; *size = GST_AUDIO_INFO_BPF (&info); return TRUE; /* ERRORS */ invalid_caps: { GST_ERROR_OBJECT (base, "invalid caps"); return FALSE; } } static GstCaps * gst_audio_resample_transform_caps (GstBaseTransform * base, GstPadDirection direction, GstCaps * caps, GstCaps * filter) { const GValue *val; GstStructure *s; GstCaps *res; gint i, n; /* transform single caps into input_caps + input_caps with the rate * field set to our supported range. This ensures that upstream knows * about downstream's prefered rate(s) and can negotiate accordingly. */ res = gst_caps_new_empty (); n = gst_caps_get_size (caps); for (i = 0; i < n; i++) { s = gst_caps_get_structure (caps, i); /* If this is already expressed by the existing caps * skip this structure */ if (i > 0 && gst_caps_is_subset_structure (res, s)) continue; /* first, however, check if the caps contain a range for the rate field, in * which case that side isn't going to care much about the exact sample rate * chosen and we should just assume things will get fixated to something sane * and we may just as well offer our full range instead of the range in the * caps. If the rate is not an int range value, it's likely to express a * real preference or limitation and we should maintain that structure as * preference by putting it first into the transformed caps, and only add * our full rate range as second option */ s = gst_structure_copy (s); val = gst_structure_get_value (s, "rate"); if (val == NULL || GST_VALUE_HOLDS_INT_RANGE (val)) { /* overwrite existing range, or add field if it doesn't exist yet */ gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL); } else { /* append caps with full range to existing caps with non-range rate field */ gst_caps_append_structure (res, gst_structure_copy (s)); gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL); } gst_caps_append_structure (res, s); } if (filter) { GstCaps *intersection; intersection = gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (res); res = intersection; } return res; } /* Fixate rate to the allowed rate that has the smallest difference */ static GstCaps * 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 othercaps; othercaps = gst_caps_truncate (othercaps); othercaps = gst_caps_make_writable (othercaps); s = gst_caps_get_structure (othercaps, 0); gst_structure_fixate_field_nearest_int (s, "rate", rate); return othercaps; } 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, SpeexResamplerSincFilterMode sinc_filter_mode, guint32 sinc_filter_auto_threshold) { 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, sinc_filter_mode, sinc_filter_auto_threshold, &err); if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS)) { GST_ERROR_OBJECT (resample, "Failed to create resampler state: %s", funcs->strerror (err)); return NULL; } if (sinc_filter_mode == SPEEX_RESAMPLER_SINC_FILTER_AUTO) { GST_INFO_OBJECT (resample, "Using the %s sinc filter table", funcs->get_sinc_filter_mode(ret) ? "full" : "interpolated"); } 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, SpeexResamplerSincFilterMode sinc_filter_mode, guint32 sinc_filter_auto_threshold) { 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 || sinc_filter_mode != resample->sinc_filter_mode || sinc_filter_auto_threshold != resample->sinc_filter_auto_threshold) { resample->funcs->destroy (resample->state); resample->state = gst_audio_resample_init_state (resample, width, channels, inrate, outrate, quality, fp, sinc_filter_mode, sinc_filter_auto_threshold); 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; resample->sinc_filter_mode = sinc_filter_mode; resample->sinc_filter_auto_threshold = sinc_filter_auto_threshold; 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 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, gsize size, GstCaps * othercaps, gsize * othersize) { gboolean ret = TRUE; GstAudioInfo in, out; guint32 ratio_den, ratio_num; gint inrate, outrate, gcd; gint bpf; GST_LOG_OBJECT (base, "asked to transform size %" G_GSIZE_FORMAT " in direction %s", size, direction == GST_PAD_SINK ? "SINK" : "SRC"); /* Get sample width -> bytes_per_samp, channels, inrate, outrate */ ret = gst_audio_info_from_caps (&in, caps); ret &= gst_audio_info_from_caps (&out, othercaps); if (G_UNLIKELY (!ret)) { GST_ERROR_OBJECT (base, "Wrong caps"); return FALSE; } /* Number of samples in either buffer is size / (width*channels) -> * calculate the factor */ bpf = GST_AUDIO_INFO_BPF (&in); inrate = GST_AUDIO_INFO_RATE (&in); outrate = GST_AUDIO_INFO_RATE (&out); /* Convert source buffer size to samples */ size /= bpf; /* 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 *= bpf; } else { /* asked to convert size of an outgoing buffer. Round down the input size */ *othersize = gst_util_uint64_scale_int (size, ratio_num, ratio_den); *othersize *= bpf; } GST_LOG_OBJECT (base, "transformed size %" G_GSIZE_FORMAT " to %" G_GSIZE_FORMAT, size * bpf, *othersize); return ret; } static gboolean gst_audio_resample_set_caps (GstBaseTransform * base, GstCaps * incaps, GstCaps * outcaps) { gboolean ret; gint width, inrate, outrate, channels; gboolean fp; GstAudioResample *resample = GST_AUDIO_RESAMPLE (base); GstAudioInfo in, out; GST_LOG ("incaps %" GST_PTR_FORMAT ", outcaps %" GST_PTR_FORMAT, incaps, outcaps); if (!gst_audio_info_from_caps (&in, incaps)) goto invalid_incaps; if (!gst_audio_info_from_caps (&out, outcaps)) goto invalid_outcaps; /* FIXME do some checks */ /* take new values */ width = GST_AUDIO_FORMAT_INFO_WIDTH (in.finfo); channels = GST_AUDIO_INFO_CHANNELS (&in); inrate = GST_AUDIO_INFO_RATE (&in); outrate = GST_AUDIO_INFO_RATE (&out); fp = GST_AUDIO_FORMAT_INFO_IS_FLOAT (in.finfo); ret = gst_audio_resample_update_state (resample, width, channels, inrate, outrate, resample->quality, fp, resample->sinc_filter_mode, resample->sinc_filter_auto_threshold); if (G_UNLIKELY (!ret)) return FALSE; return TRUE; /* ERROR */ invalid_incaps: { GST_ERROR_OBJECT (base, "invalid incaps"); return FALSE; } invalid_outcaps: { GST_ERROR_OBJECT (base, "invalid outcaps"); return FALSE; } } #define GST_MAXINT24 (8388607) #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; } /* 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 G_GNUC_UNUSED, 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; GstMapInfo map; 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; outbuf = gst_buffer_new_and_alloc (outsize); gst_buffer_map (outbuf, &map, 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, map.data, out_processed, TRUE); } else { /* don't need to convert data format; process */ err = resample->funcs->process (resample->state, NULL, &in_processed, map.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, &map); gst_buffer_resize (outbuf, 0, 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_sink_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); 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_SEGMENT: 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: 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 GST_BASE_TRANSFORM_CLASS (parent_class)->sink_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)))) return FALSE; /* 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; 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) { GstMapInfo in_map, out_map; gsize outsize; guint32 in_len, in_processed; guint32 out_len, out_processed; guint filt_len = resample->funcs->get_filt_len (resample->state); gst_buffer_map (inbuf, &in_map, GST_MAP_READ); gst_buffer_map (outbuf, &out_map, GST_MAP_WRITE); in_len = in_map.size / resample->channels; out_len = out_map.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; } { 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_map.data, 0, out_map.size); GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP); resample->num_gap_samples += in_len; in_processed = in_len; } } else { /* not a gap */ gint err; 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_map); gst_buffer_unmap (outbuf, &out_map); return GST_FLOW_ERROR; } /* convert input */ gst_audio_resample_convert_buffer (resample, in_map.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_map.data, out_processed, TRUE); } else { /* no format conversion required; process */ err = resample->funcs->process (resample->state, in_map.data, &in_processed, out_map.data, &out_processed); } if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS)) { GST_ERROR_OBJECT (resample, "Failed to convert data: %s", resample->funcs->strerror (err)); gst_buffer_unmap (inbuf, &in_map); gst_buffer_unmap (outbuf, &out_map); return GST_FLOW_ERROR; } } /* 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; gst_buffer_unmap (inbuf, &in_map); gst_buffer_unmap (outbuf, &out_map); outsize = out_processed * resample->channels * (resample->width / 8); gst_buffer_resize (outbuf, 0, outsize); GST_LOG_OBJECT (resample, "Converted to buffer of %" G_GUINT32_FORMAT " samples (%" G_GSIZE_FORMAT " bytes) with timestamp %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT ", offset_end %" G_GUINT64_FORMAT, out_processed, outsize, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf)); return GST_FLOW_OK; } static GstFlowReturn gst_audio_resample_transform (GstBaseTransform * base, GstBuffer * inbuf, GstBuffer * outbuf) { 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, resample->sinc_filter_mode, resample->sinc_filter_auto_threshold)))) return GST_FLOW_ERROR; resample->funcs = gst_audio_resample_get_funcs (resample->width, resample->fp); } GST_LOG_OBJECT (resample, "transforming buffer of %" G_GSIZE_FORMAT " 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)); /* 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)) { 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; } 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, GstObject * parent, GstQuery * query) { GstAudioResample *resample = GST_AUDIO_RESAMPLE (parent); GstBaseTransform *trans; gboolean res = TRUE; trans = GST_BASE_TRANSFORM (resample); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_LATENCY: { GstClockTime min, max; gboolean live; guint64 latency; 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 ((res = gst_pad_peer_query (GST_BASE_TRANSFORM_SINK_PAD (trans), query))) { gst_query_parse_latency (query, &live, &min, &max); GST_DEBUG_OBJECT (resample, "Peer latency: min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT, GST_TIME_ARGS (min), GST_TIME_ARGS (max)); /* 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); } break; } default: res = gst_pad_query_default (pad, parent, query); break; } return res; } static void gst_audio_resample_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioResample *resample; gint quality; resample = GST_AUDIO_RESAMPLE (object); switch (prop_id) { case PROP_QUALITY: /* FIXME locking! */ quality = g_value_get_int (value); GST_DEBUG_OBJECT (resample, "new quality %d", quality); gst_audio_resample_update_state (resample, resample->width, resample->channels, resample->inrate, resample->outrate, quality, resample->fp, resample->sinc_filter_mode, resample->sinc_filter_auto_threshold); break; case PROP_SINC_FILTER_MODE: { /* FIXME locking! */ SpeexResamplerSincFilterMode sinc_filter_mode = g_value_get_enum (value); gst_audio_resample_update_state (resample, resample->width, resample->channels, resample->inrate, resample->outrate, resample->quality, resample->fp, sinc_filter_mode, resample->sinc_filter_auto_threshold); break; } case PROP_SINC_FILTER_AUTO_THRESHOLD: { /* FIXME locking! */ guint32 sinc_filter_auto_threshold = g_value_get_uint (value); gst_audio_resample_update_state (resample, resample->width, resample->channels, resample->inrate, resample->outrate, resample->quality, resample->fp, resample->sinc_filter_mode, sinc_filter_auto_threshold); 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_SINC_FILTER_MODE: g_value_set_enum(value, resample->sinc_filter_mode); break; case PROP_SINC_FILTER_AUTO_THRESHOLD: g_value_set_uint(value, resample->sinc_filter_auto_threshold); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GType speex_resampler_sinc_filter_mode_get_type (void) { static GType speex_resampler_sinc_filter_mode_type = 0; if (!speex_resampler_sinc_filter_mode_type) { static GEnumValue sinc_filter_modes[] = { { SPEEX_RESAMPLER_SINC_FILTER_INTERPOLATED, "Use interpolated sinc table", "interpolated" }, { SPEEX_RESAMPLER_SINC_FILTER_FULL, "Use full sinc table", "full" }, { SPEEX_RESAMPLER_SINC_FILTER_AUTO, "Use full table if table size below threshold", "auto" }, { 0, NULL, NULL }, }; speex_resampler_sinc_filter_mode_type = g_enum_register_static ( "SpeexResamplerSincFilterMode", sinc_filter_modes); } return speex_resampler_sinc_filter_mode_type; } /* 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, }, G_GNUC_UNUSED 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, SPEEX_RESAMPLER_SINC_FILTER_INTERPOLATED, SPEEX_RESAMPLER_SINC_FILTER_AUTO_THRESHOLD_DEFAULT, NULL); if (sta == NULL) { GST_ERROR ("Failed to create float resampler state"); return FALSE; } stb = resample_int_resampler_init (1, 48000, 24000, 4, SPEEX_RESAMPLER_SINC_FILTER_INTERPOLATED, SPEEX_RESAMPLER_SINC_FILTER_AUTO_THRESHOLD_DEFAULT, 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 appropriate: %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) { 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_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_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);