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Modernizing the documentation, making it simpler to read an modify and allowing us to possibly switch to hotdoc in the future.
1099 lines
36 KiB
C
1099 lines
36 KiB
C
/* GStreamer
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* Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) 2003,2004 David A. Schleef <ds@schleef.org>
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* Copyright (C) 2007-2008 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-audioresample
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* @title: audioresample
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*
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* audioresample resamples raw audio buffers to different sample rates using
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* a configurable windowing function to enhance quality.
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*
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* By default, the resampler uses a reduced sinc table, with cubic interpolation filling in
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* the gaps. This ensures that the table does not become too big. However, the interpolation
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* increases the CPU usage considerably. As an alternative, a full sinc table can be used.
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* Doing so can drastically reduce CPU usage (4x faster with 44.1 -> 48 kHz conversions for
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* example), at the cost of increased memory consumption, plus the sinc table takes longer
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* to initialize when the element is created. A third mode exists, which uses the full table
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* unless said table would become too large, in which case the interpolated one is used instead.
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*
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* ## Example launch line
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* |[
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* gst-launch-1.0 -v uridecodebin uri=file:///path/to/audio.ogg ! audioconvert ! audioresample ! audio/x-raw, rate=8000 ! autoaudiosink
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* ]|
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* Decode an audio file and downsample it to 8Khz and play sound.
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* To create the Ogg/Vorbis file refer to the documentation of vorbisenc.
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* This assumes there is an audio sink that will accept/handle 8kHz audio.
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*
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*/
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/* TODO:
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* - Enable SSE/ARM optimizations and select at runtime
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <math.h>
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#include "gstaudioresample.h"
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#include <gst/gstutils.h>
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#include <gst/audio/audio.h>
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#include <gst/base/gstbasetransform.h>
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GST_DEBUG_CATEGORY (audio_resample_debug);
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#define GST_CAT_DEFAULT audio_resample_debug
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#if !defined(AUDIORESAMPLE_FORMAT_AUTO)
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_PERFORMANCE);
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#endif
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#undef USE_SPEEX
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#define DEFAULT_QUALITY GST_AUDIO_RESAMPLER_QUALITY_DEFAULT
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#define DEFAULT_RESAMPLE_METHOD GST_AUDIO_RESAMPLER_METHOD_KAISER
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#define DEFAULT_SINC_FILTER_MODE GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO
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#define DEFAULT_SINC_FILTER_AUTO_THRESHOLD (1*1048576)
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#define DEFAULT_SINC_FILTER_INTERPOLATION GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC
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enum
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{
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PROP_0,
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PROP_QUALITY,
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PROP_RESAMPLE_METHOD,
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PROP_SINC_FILTER_MODE,
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PROP_SINC_FILTER_AUTO_THRESHOLD,
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PROP_SINC_FILTER_INTERPOLATION
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};
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#define SUPPORTED_CAPS \
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GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL) \
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", layout = (string) { interleaved, non-interleaved }"
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static GstStaticPadTemplate gst_audio_resample_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (SUPPORTED_CAPS));
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static GstStaticPadTemplate gst_audio_resample_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (SUPPORTED_CAPS));
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static void gst_audio_resample_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_audio_resample_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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/* vmethods */
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static gboolean gst_audio_resample_get_unit_size (GstBaseTransform * base,
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GstCaps * caps, gsize * size);
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static GstCaps *gst_audio_resample_transform_caps (GstBaseTransform * base,
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GstPadDirection direction, GstCaps * caps, GstCaps * filter);
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static GstCaps *gst_audio_resample_fixate_caps (GstBaseTransform * base,
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GstPadDirection direction, GstCaps * caps, GstCaps * othercaps);
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static gboolean gst_audio_resample_transform_size (GstBaseTransform * trans,
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GstPadDirection direction, GstCaps * incaps, gsize insize,
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GstCaps * outcaps, gsize * outsize);
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static gboolean gst_audio_resample_set_caps (GstBaseTransform * base,
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GstCaps * incaps, GstCaps * outcaps);
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static GstFlowReturn gst_audio_resample_transform (GstBaseTransform * base,
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GstBuffer * inbuf, GstBuffer * outbuf);
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static gboolean gst_audio_resample_transform_meta (GstBaseTransform * trans,
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GstBuffer * outbuf, GstMeta * meta, GstBuffer * inbuf);
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static GstFlowReturn gst_audio_resample_submit_input_buffer (GstBaseTransform *
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base, gboolean is_discont, GstBuffer * input);
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static gboolean gst_audio_resample_sink_event (GstBaseTransform * base,
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GstEvent * event);
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static gboolean gst_audio_resample_start (GstBaseTransform * base);
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static gboolean gst_audio_resample_stop (GstBaseTransform * base);
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static gboolean gst_audio_resample_query (GstPad * pad, GstObject * parent,
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GstQuery * query);
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#define gst_audio_resample_parent_class parent_class
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G_DEFINE_TYPE (GstAudioResample, gst_audio_resample, GST_TYPE_BASE_TRANSFORM);
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static void
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gst_audio_resample_class_init (GstAudioResampleClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstElementClass *gstelement_class = (GstElementClass *) klass;
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gobject_class->set_property = gst_audio_resample_set_property;
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gobject_class->get_property = gst_audio_resample_get_property;
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g_object_class_install_property (gobject_class, PROP_QUALITY,
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g_param_spec_int ("quality", "Quality", "Resample quality with 0 being "
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"the lowest and 10 being the best",
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GST_AUDIO_RESAMPLER_QUALITY_MIN, GST_AUDIO_RESAMPLER_QUALITY_MAX,
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DEFAULT_QUALITY,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_RESAMPLE_METHOD,
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g_param_spec_enum ("resample-method", "Resample method to use",
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"What resample method to use",
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GST_TYPE_AUDIO_RESAMPLER_METHOD,
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DEFAULT_RESAMPLE_METHOD, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_SINC_FILTER_MODE,
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g_param_spec_enum ("sinc-filter-mode", "Sinc filter table mode",
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"What sinc filter table mode to use",
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GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE,
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DEFAULT_SINC_FILTER_MODE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_SINC_FILTER_AUTO_THRESHOLD,
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g_param_spec_uint ("sinc-filter-auto-threshold",
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"Sinc filter auto mode threshold",
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"Memory usage threshold to use if sinc filter mode is AUTO, given in bytes",
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0, G_MAXUINT, DEFAULT_SINC_FILTER_AUTO_THRESHOLD,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_SINC_FILTER_INTERPOLATION,
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g_param_spec_enum ("sinc-filter-interpolation",
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"Sinc filter interpolation",
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"How to interpolate the sinc filter table",
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GST_TYPE_AUDIO_RESAMPLER_FILTER_INTERPOLATION,
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DEFAULT_SINC_FILTER_INTERPOLATION,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_audio_resample_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_audio_resample_sink_template);
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gst_element_class_set_static_metadata (gstelement_class, "Audio resampler",
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"Filter/Converter/Audio", "Resamples audio",
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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GST_BASE_TRANSFORM_CLASS (klass)->start =
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GST_DEBUG_FUNCPTR (gst_audio_resample_start);
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GST_BASE_TRANSFORM_CLASS (klass)->stop =
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GST_DEBUG_FUNCPTR (gst_audio_resample_stop);
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GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
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GST_DEBUG_FUNCPTR (gst_audio_resample_transform_size);
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GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
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GST_DEBUG_FUNCPTR (gst_audio_resample_get_unit_size);
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GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
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GST_DEBUG_FUNCPTR (gst_audio_resample_transform_caps);
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GST_BASE_TRANSFORM_CLASS (klass)->fixate_caps =
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GST_DEBUG_FUNCPTR (gst_audio_resample_fixate_caps);
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GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
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GST_DEBUG_FUNCPTR (gst_audio_resample_set_caps);
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GST_BASE_TRANSFORM_CLASS (klass)->transform =
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GST_DEBUG_FUNCPTR (gst_audio_resample_transform);
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GST_BASE_TRANSFORM_CLASS (klass)->sink_event =
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GST_DEBUG_FUNCPTR (gst_audio_resample_sink_event);
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GST_BASE_TRANSFORM_CLASS (klass)->transform_meta =
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GST_DEBUG_FUNCPTR (gst_audio_resample_transform_meta);
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GST_BASE_TRANSFORM_CLASS (klass)->submit_input_buffer =
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GST_DEBUG_FUNCPTR (gst_audio_resample_submit_input_buffer);
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GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE;
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}
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static void
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gst_audio_resample_init (GstAudioResample * resample)
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{
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GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
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resample->method = DEFAULT_RESAMPLE_METHOD;
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resample->quality = DEFAULT_QUALITY;
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resample->sinc_filter_mode = DEFAULT_SINC_FILTER_MODE;
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resample->sinc_filter_auto_threshold = DEFAULT_SINC_FILTER_AUTO_THRESHOLD;
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resample->sinc_filter_interpolation = DEFAULT_SINC_FILTER_INTERPOLATION;
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gst_base_transform_set_gap_aware (trans, TRUE);
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gst_pad_set_query_function (trans->srcpad, gst_audio_resample_query);
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}
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/* vmethods */
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static gboolean
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gst_audio_resample_start (GstBaseTransform * base)
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{
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GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
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resample->need_discont = TRUE;
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resample->num_gap_samples = 0;
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resample->num_nongap_samples = 0;
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resample->t0 = GST_CLOCK_TIME_NONE;
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resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
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resample->out_offset0 = GST_BUFFER_OFFSET_NONE;
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resample->samples_in = 0;
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resample->samples_out = 0;
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return TRUE;
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}
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static gboolean
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gst_audio_resample_stop (GstBaseTransform * base)
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{
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GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
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if (resample->converter) {
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gst_audio_converter_free (resample->converter);
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resample->converter = NULL;
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}
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return TRUE;
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}
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static gboolean
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gst_audio_resample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
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gsize * size)
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{
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GstAudioInfo info;
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if (!gst_audio_info_from_caps (&info, caps))
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goto invalid_caps;
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*size = GST_AUDIO_INFO_BPF (&info);
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return TRUE;
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/* ERRORS */
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invalid_caps:
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{
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GST_ERROR_OBJECT (base, "invalid caps");
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return FALSE;
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}
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}
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static GstCaps *
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gst_audio_resample_transform_caps (GstBaseTransform * base,
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GstPadDirection direction, GstCaps * caps, GstCaps * filter)
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{
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const GValue *val;
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GstStructure *s;
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GstCaps *res;
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gint i, n;
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/* transform single caps into input_caps + input_caps with the rate
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* field set to our supported range. This ensures that upstream knows
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* about downstream's prefered rate(s) and can negotiate accordingly. */
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res = gst_caps_new_empty ();
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n = gst_caps_get_size (caps);
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for (i = 0; i < n; i++) {
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s = gst_caps_get_structure (caps, i);
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/* If this is already expressed by the existing caps
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* skip this structure */
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if (i > 0 && gst_caps_is_subset_structure (res, s))
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continue;
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/* first, however, check if the caps contain a range for the rate field, in
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* which case that side isn't going to care much about the exact sample rate
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* chosen and we should just assume things will get fixated to something sane
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* and we may just as well offer our full range instead of the range in the
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* caps. If the rate is not an int range value, it's likely to express a
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* real preference or limitation and we should maintain that structure as
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* preference by putting it first into the transformed caps, and only add
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* our full rate range as second option */
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s = gst_structure_copy (s);
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val = gst_structure_get_value (s, "rate");
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if (val == NULL || GST_VALUE_HOLDS_INT_RANGE (val)) {
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/* overwrite existing range, or add field if it doesn't exist yet */
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gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
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} else {
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/* append caps with full range to existing caps with non-range rate field */
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gst_caps_append_structure (res, gst_structure_copy (s));
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gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
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}
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gst_caps_append_structure (res, s);
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}
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if (filter) {
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GstCaps *intersection;
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intersection =
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gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (res);
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res = intersection;
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}
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return res;
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}
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/* Fixate rate to the allowed rate that has the smallest difference */
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static GstCaps *
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gst_audio_resample_fixate_caps (GstBaseTransform * base,
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GstPadDirection direction, GstCaps * caps, GstCaps * othercaps)
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{
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GstStructure *s;
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gint rate;
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s = gst_caps_get_structure (caps, 0);
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if (G_UNLIKELY (!gst_structure_get_int (s, "rate", &rate)))
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return othercaps;
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othercaps = gst_caps_truncate (othercaps);
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othercaps = gst_caps_make_writable (othercaps);
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s = gst_caps_get_structure (othercaps, 0);
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gst_structure_fixate_field_nearest_int (s, "rate", rate);
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return othercaps;
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}
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static GstStructure *
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make_options (GstAudioResample * resample, GstAudioInfo * in,
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GstAudioInfo * out)
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{
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GstStructure *options;
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options = gst_structure_new_empty ("resampler-options");
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if (in != NULL && out != NULL)
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gst_audio_resampler_options_set_quality (resample->method,
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resample->quality, in->rate, out->rate, options);
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gst_structure_set (options,
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GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD, GST_TYPE_AUDIO_RESAMPLER_METHOD,
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resample->method,
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GST_AUDIO_RESAMPLER_OPT_FILTER_MODE, GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE,
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resample->sinc_filter_mode, GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD,
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G_TYPE_UINT, resample->sinc_filter_auto_threshold,
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GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION,
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GST_TYPE_AUDIO_RESAMPLER_FILTER_INTERPOLATION,
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resample->sinc_filter_interpolation, NULL);
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return options;
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}
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static gboolean
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gst_audio_resample_update_state (GstAudioResample * resample, GstAudioInfo * in,
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GstAudioInfo * out)
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{
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gboolean updated_latency = FALSE;
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gsize old_latency = -1;
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GstStructure *options;
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if (resample->converter == NULL && in == NULL && out == NULL)
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return TRUE;
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options = make_options (resample, in, out);
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|
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if (resample->converter)
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old_latency = gst_audio_converter_get_max_latency (resample->converter);
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/* if channels and layout changed, destroy existing resampler */
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if (in != NULL && (in->finfo != resample->in.finfo ||
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in->channels != resample->in.channels ||
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in->layout != resample->in.layout) && resample->converter) {
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gst_audio_converter_free (resample->converter);
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resample->converter = NULL;
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}
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if (resample->converter == NULL) {
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resample->converter =
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gst_audio_converter_new (GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE, in,
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out, options);
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if (resample->converter == NULL)
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goto resampler_failed;
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} else if (in && out) {
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gboolean ret;
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ret =
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gst_audio_converter_update_config (resample->converter, in->rate,
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out->rate, options);
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if (!ret)
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goto update_failed;
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} else {
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gst_structure_free (options);
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}
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if (old_latency != -1)
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updated_latency =
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old_latency !=
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gst_audio_converter_get_max_latency (resample->converter);
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if (updated_latency)
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gst_element_post_message (GST_ELEMENT (resample),
|
|
gst_message_new_latency (GST_OBJECT (resample)));
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
resampler_failed:
|
|
{
|
|
GST_ERROR_OBJECT (resample, "failed to create resampler");
|
|
return FALSE;
|
|
}
|
|
update_failed:
|
|
{
|
|
GST_ERROR_OBJECT (resample, "failed to update resampler");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_audio_resample_reset_state (GstAudioResample * resample)
|
|
{
|
|
if (resample->converter)
|
|
gst_audio_converter_reset (resample->converter);
|
|
}
|
|
|
|
static gboolean
|
|
gst_audio_resample_transform_size (GstBaseTransform * base,
|
|
GstPadDirection direction, GstCaps * caps, gsize size, GstCaps * othercaps,
|
|
gsize * othersize)
|
|
{
|
|
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
|
|
gboolean ret = TRUE;
|
|
gint bpf;
|
|
|
|
GST_LOG_OBJECT (base, "asked to transform size %" G_GSIZE_FORMAT
|
|
" in direction %s", size, direction == GST_PAD_SINK ? "SINK" : "SRC");
|
|
|
|
/* Number of samples in either buffer is size / (width*channels) ->
|
|
* calculate the factor */
|
|
bpf = GST_AUDIO_INFO_BPF (&resample->in);
|
|
|
|
/* Convert source buffer size to samples */
|
|
size /= bpf;
|
|
|
|
if (direction == GST_PAD_SINK) {
|
|
/* asked to convert size of an incoming buffer */
|
|
*othersize = gst_audio_converter_get_out_frames (resample->converter, size);
|
|
*othersize *= bpf;
|
|
} else {
|
|
/* asked to convert size of an outgoing buffer */
|
|
*othersize = gst_audio_converter_get_in_frames (resample->converter, size);
|
|
*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)
|
|
{
|
|
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 */
|
|
gst_audio_resample_update_state (resample, &in, &out);
|
|
|
|
resample->in = in;
|
|
resample->out = out;
|
|
|
|
return TRUE;
|
|
|
|
/* ERROR */
|
|
invalid_incaps:
|
|
{
|
|
GST_ERROR_OBJECT (base, "invalid incaps");
|
|
return FALSE;
|
|
}
|
|
invalid_outcaps:
|
|
{
|
|
GST_ERROR_OBJECT (base, "invalid outcaps");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* Push history_len zeros into the filter, but discard the output. */
|
|
static void
|
|
gst_audio_resample_dump_drain (GstAudioResample * resample, guint history_len)
|
|
{
|
|
gsize out_len, outsize;
|
|
gpointer out[1];
|
|
|
|
out_len =
|
|
gst_audio_converter_get_out_frames (resample->converter, history_len);
|
|
if (out_len == 0)
|
|
return;
|
|
|
|
outsize = out_len * resample->out.bpf;
|
|
|
|
out[0] = g_malloc (outsize);
|
|
gst_audio_converter_samples (resample->converter, 0, NULL, history_len,
|
|
out, out_len);
|
|
g_free (out[0]);
|
|
}
|
|
|
|
static void
|
|
gst_audio_resample_push_drain (GstAudioResample * resample, guint history_len)
|
|
{
|
|
GstBuffer *outbuf;
|
|
GstFlowReturn res;
|
|
gint outsize;
|
|
gsize out_len;
|
|
GstMapInfo map;
|
|
gpointer out[1];
|
|
|
|
g_assert (resample->converter != NULL);
|
|
|
|
/* Don't drain samples if we were reset. */
|
|
if (!GST_CLOCK_TIME_IS_VALID (resample->t0))
|
|
return;
|
|
|
|
out_len =
|
|
gst_audio_converter_get_out_frames (resample->converter, history_len);
|
|
if (out_len == 0)
|
|
return;
|
|
|
|
outsize = out_len * resample->in.bpf;
|
|
outbuf = gst_buffer_new_and_alloc (outsize);
|
|
|
|
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
|
|
|
|
out[0] = map.data;
|
|
gst_audio_converter_samples (resample->converter, 0, NULL, history_len,
|
|
out, out_len);
|
|
|
|
gst_buffer_unmap (outbuf, &map);
|
|
|
|
/* 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->out.rate);
|
|
GST_BUFFER_DURATION (outbuf) = resample->t0 +
|
|
gst_util_uint64_scale_int_round (resample->samples_out + out_len,
|
|
GST_SECOND, resample->out.rate) - 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_len;
|
|
} else {
|
|
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
|
|
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
|
|
}
|
|
/* move along */
|
|
resample->samples_out += out_len;
|
|
resample->samples_in += history_len;
|
|
|
|
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);
|
|
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->converter) {
|
|
gsize latency =
|
|
gst_audio_converter_get_max_latency (resample->converter);
|
|
gst_audio_resample_push_drain (resample, latency);
|
|
}
|
|
gst_audio_resample_reset_state (resample);
|
|
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->converter) {
|
|
gsize latency =
|
|
gst_audio_converter_get_max_latency (resample->converter);
|
|
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->in.rate, 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->in.rate >> 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->in.rate)));
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf,
|
|
GstBuffer * outbuf)
|
|
{
|
|
GstMapInfo in_map, out_map;
|
|
gsize outsize;
|
|
guint32 in_len;
|
|
guint32 out_len;
|
|
guint filt_len =
|
|
gst_audio_converter_get_max_latency (resample->converter) * 2;
|
|
gboolean inbuf_writable;
|
|
|
|
inbuf_writable = gst_buffer_is_writable (inbuf)
|
|
&& gst_buffer_n_memory (inbuf) == 1
|
|
&& gst_memory_is_writable (gst_buffer_peek_memory (inbuf, 0));
|
|
|
|
gst_buffer_map (inbuf, &in_map,
|
|
inbuf_writable ? GST_MAP_READWRITE : GST_MAP_READ);
|
|
gst_buffer_map (outbuf, &out_map, GST_MAP_WRITE);
|
|
|
|
in_len = in_map.size / resample->in.bpf;
|
|
out_len = out_map.size / resample->out.bpf;
|
|
|
|
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;
|
|
|
|
num = resample->in.rate;
|
|
den = resample->out.rate;
|
|
|
|
if (resample->samples_in + in_len >= filt_len / 2)
|
|
out_len =
|
|
gst_util_uint64_scale_int_ceil (resample->samples_in + in_len -
|
|
filt_len / 2, den, num) - resample->samples_out;
|
|
else
|
|
out_len = 0;
|
|
|
|
memset (out_map.data, 0, out_map.size);
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
|
|
resample->num_gap_samples += in_len;
|
|
}
|
|
} else { /* not a gap */
|
|
if (resample->num_gap_samples > filt_len) {
|
|
/* push in enough zeros to restore the filter to the right offset */
|
|
guint num;
|
|
|
|
num = resample->in.rate;
|
|
|
|
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;
|
|
}
|
|
{
|
|
/* process */
|
|
gpointer in[1], out[1];
|
|
GstAudioConverterFlags flags;
|
|
|
|
flags = 0;
|
|
if (inbuf_writable)
|
|
flags |= GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE;
|
|
|
|
in[0] = in_map.data;
|
|
out[0] = out_map.data;
|
|
gst_audio_converter_samples (resample->converter, flags, in, in_len,
|
|
out, out_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->out.rate);
|
|
GST_BUFFER_DURATION (outbuf) = resample->t0 +
|
|
gst_util_uint64_scale_int_round (resample->samples_out + out_len,
|
|
GST_SECOND, resample->out.rate) - 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_len;
|
|
} else {
|
|
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
|
|
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
|
|
}
|
|
/* move along */
|
|
resample->samples_out += out_len;
|
|
resample->samples_in += in_len;
|
|
|
|
gst_buffer_unmap (inbuf, &in_map);
|
|
gst_buffer_unmap (outbuf, &out_map);
|
|
|
|
outsize = out_len * resample->in.bpf;
|
|
|
|
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_len, outsize,
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
|
|
GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
|
|
|
|
if (outsize == 0)
|
|
return GST_BASE_TRANSFORM_FLOW_DROPPED;
|
|
else
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_audio_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
|
|
GstBuffer * outbuf)
|
|
{
|
|
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
|
|
GstFlowReturn ret;
|
|
|
|
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))) {
|
|
gsize size;
|
|
gint bpf = GST_AUDIO_INFO_BPF (&resample->in);
|
|
|
|
gst_audio_resample_reset_state (resample);
|
|
resample->need_discont = TRUE;
|
|
|
|
/* need to recalculate the output size */
|
|
size = gst_buffer_get_size (inbuf) / bpf;
|
|
size = gst_audio_converter_get_out_frames (resample->converter, size);
|
|
gst_buffer_set_size (outbuf, size * bpf);
|
|
}
|
|
|
|
/* handle discontinuity */
|
|
if (G_UNLIKELY (resample->need_discont)) {
|
|
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->out.rate, resample->in.rate);
|
|
} 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_transform_meta (GstBaseTransform * trans, GstBuffer * outbuf,
|
|
GstMeta * meta, GstBuffer * inbuf)
|
|
{
|
|
const GstMetaInfo *info = meta->info;
|
|
const gchar *const *tags;
|
|
|
|
tags = gst_meta_api_type_get_tags (info->api);
|
|
|
|
if (!tags || (g_strv_length ((gchar **) tags) == 1
|
|
&& gst_meta_api_type_has_tag (info->api,
|
|
g_quark_from_string (GST_META_TAG_AUDIO_STR))))
|
|
return TRUE;
|
|
|
|
return FALSE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_audio_resample_submit_input_buffer (GstBaseTransform * base,
|
|
gboolean is_discont, GstBuffer * input)
|
|
{
|
|
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
|
|
|
|
if (base->segment.format == GST_FORMAT_TIME) {
|
|
input =
|
|
gst_audio_buffer_clip (input, &base->segment, resample->in.rate,
|
|
resample->in.bpf);
|
|
|
|
if (!input)
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
return GST_BASE_TRANSFORM_CLASS (parent_class)->submit_input_buffer (base,
|
|
is_discont, input);
|
|
}
|
|
|
|
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->in.rate;
|
|
gint resampler_latency;
|
|
|
|
if (resample->converter)
|
|
resampler_latency =
|
|
gst_audio_converter_get_max_latency (resample->converter);
|
|
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;
|
|
|
|
resample = GST_AUDIO_RESAMPLE (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_QUALITY:
|
|
/* FIXME locking! */
|
|
resample->quality = g_value_get_int (value);
|
|
GST_DEBUG_OBJECT (resample, "new quality %d", resample->quality);
|
|
gst_audio_resample_update_state (resample, NULL, NULL);
|
|
break;
|
|
case PROP_RESAMPLE_METHOD:
|
|
resample->method = g_value_get_enum (value);
|
|
gst_audio_resample_update_state (resample, NULL, NULL);
|
|
break;
|
|
case PROP_SINC_FILTER_MODE:
|
|
/* FIXME locking! */
|
|
resample->sinc_filter_mode = g_value_get_enum (value);
|
|
gst_audio_resample_update_state (resample, NULL, NULL);
|
|
break;
|
|
case PROP_SINC_FILTER_AUTO_THRESHOLD:
|
|
/* FIXME locking! */
|
|
resample->sinc_filter_auto_threshold = g_value_get_uint (value);
|
|
gst_audio_resample_update_state (resample, NULL, NULL);
|
|
break;
|
|
case PROP_SINC_FILTER_INTERPOLATION:
|
|
/* FIXME locking! */
|
|
resample->sinc_filter_interpolation = g_value_get_enum (value);
|
|
gst_audio_resample_update_state (resample, NULL, NULL);
|
|
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_RESAMPLE_METHOD:
|
|
g_value_set_enum (value, resample->method);
|
|
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;
|
|
case PROP_SINC_FILTER_INTERPOLATION:
|
|
g_value_set_enum (value, resample->sinc_filter_interpolation);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
GST_DEBUG_CATEGORY_INIT (audio_resample_debug, "audioresample", 0,
|
|
"audio resampling element");
|
|
|
|
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);
|