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996 lines
31 KiB
C
996 lines
31 KiB
C
/* GStreamer
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* Copyright (C) 2003 Benjamin Otte <in7y118@public.uni-hamburg.de>
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* Copyright (C) 2005 Thomas Vander Stichele <thomas at apestaart dot org>
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* Copyright (C) 2011 Wim Taymans <wim.taymans at gmail dot com>
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*
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* gstaudioconvert.c: Convert audio to different audio formats automatically
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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-audioconvert
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* @title: audioconvert
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*
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* Audioconvert converts raw audio buffers between various possible formats.
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* It supports integer to float conversion, width/depth conversion,
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* signedness and endianness conversion and channel transformations
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* (ie. upmixing and downmixing), as well as dithering and noise-shaping.
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*
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* ## Example launch line
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* |[
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* gst-launch-1.0 -v -m audiotestsrc ! audioconvert ! audio/x-raw,format=S8,channels=2 ! level ! fakesink silent=TRUE
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* ]|
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* This pipeline converts audio to 8-bit. The level element shows that
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* the output levels still match the one for a sine wave.
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* |[
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* gst-launch-1.0 -v -m uridecodebin uri=file:///path/to/audio.flac ! audioconvert ! vorbisenc ! oggmux ! filesink location=audio.ogg
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* ]|
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* The vorbis encoder takes float audio data instead of the integer data
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* output by most other audio elements. This pipeline decodes a FLAC audio file
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* (or any other audio file for which decoders are installed) and re-encodes
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* it into an Ogg/Vorbis audio file.
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*
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* A mix matrix can be passed to audioconvert, that will govern the
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* remapping of input to output channels.
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* ## Example matrix generation code
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* To generate the matrix using code:
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*
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* |[
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* GValue v = G_VALUE_INIT;
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* GValue v2 = G_VALUE_INIT;
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* GValue v3 = G_VALUE_INIT;
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*
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* g_value_init (&v2, GST_TYPE_ARRAY);
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* g_value_init (&v3, G_TYPE_FLOAT);
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* g_value_set_float (&v3, 1);
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* gst_value_array_append_value (&v2, &v3);
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* g_value_unset (&v3);
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* [ Repeat for as many float as your input channels - unset and reinit v3 ]
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* g_value_init (&v, GST_TYPE_ARRAY);
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* gst_value_array_append_value (&v, &v2);
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* g_value_unset (&v2);
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* [ Repeat for as many v2's as your output channels - unset and reinit v2]
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* g_object_set_property (G_OBJECT (audioconvert), "mix-matrix", &v);
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* g_value_unset (&v);
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* ]|
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*
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* ## Example launch line
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* |[
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* gst-launch-1.0 audiotestsrc ! audio/x-raw, channels=4 ! audioconvert mix-matrix="<<(float)1.0, (float)0.0, (float)0.0, (float)0.0>, <(float)0.0, (float)1.0, (float)0.0, (float)0.0>>" ! audio/x-raw,channels=2 ! autoaudiosink
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* ]|
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*
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* > If an empty mix matrix is specified, a (potentially truncated)
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* > identity matrix will be generated.
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*
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* ## Example empty matrix generation code
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* |[
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* GValue v = G_VALUE_INIT;
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*
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* g_value_init (&v, GST_TYPE_ARRAY);
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* g_object_set_property (G_OBJECT (audioconvert), "mix-matrix", &v);
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* g_value_unset (&v);
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* ]|
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*
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* ## Example empty matrix launch line
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* |[
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* gst-launch-1.0 -v audiotestsrc ! audio/x-raw,channels=8 ! audioconvert mix-matrix="<>" ! audio/x-raw,channels=16,channel-mask=\(bitmask\)0x0000000000000000 ! fakesink
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* ]|
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*/
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/*
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* design decisions:
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* - audioconvert converts buffers in a set of supported caps. If it supports
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* a caps, it supports conversion from these caps to any other caps it
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* supports. (example: if it does A=>B and A=>C, it also does B=>C)
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* - audioconvert does not save state between buffers. Every incoming buffer is
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* converted and the converted buffer is pushed out.
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* conclusion:
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* audioconvert is not supposed to be a one-element-does-anything solution for
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* audio conversions.
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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 "gstaudioconvert.h"
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#include "plugin.h"
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GST_DEBUG_CATEGORY (audio_convert_debug);
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_PERFORMANCE);
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#define GST_CAT_DEFAULT (audio_convert_debug)
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/*** DEFINITIONS **************************************************************/
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/* type functions */
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static void gst_audio_convert_dispose (GObject * obj);
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/* gstreamer functions */
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static gboolean gst_audio_convert_get_unit_size (GstBaseTransform * base,
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GstCaps * caps, gsize * size);
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static GstCaps *gst_audio_convert_transform_caps (GstBaseTransform * base,
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GstPadDirection direction, GstCaps * caps, GstCaps * filter);
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static GstCaps *gst_audio_convert_fixate_caps (GstBaseTransform * base,
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GstPadDirection direction, GstCaps * caps, GstCaps * othercaps);
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static gboolean gst_audio_convert_set_caps (GstBaseTransform * base,
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GstCaps * incaps, GstCaps * outcaps);
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static GstFlowReturn gst_audio_convert_transform (GstBaseTransform * base,
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GstBuffer * inbuf, GstBuffer * outbuf);
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static GstFlowReturn gst_audio_convert_transform_ip (GstBaseTransform * base,
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GstBuffer * buf);
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static gboolean gst_audio_convert_transform_meta (GstBaseTransform * trans,
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GstBuffer * outbuf, GstMeta * meta, GstBuffer * inbuf);
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static GstFlowReturn gst_audio_convert_submit_input_buffer (GstBaseTransform *
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base, gboolean is_discont, GstBuffer * input);
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static void gst_audio_convert_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_audio_convert_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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/* AudioConvert signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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PROP_0,
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PROP_DITHERING,
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PROP_NOISE_SHAPING,
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PROP_MIX_MATRIX,
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};
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#define DEBUG_INIT \
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GST_DEBUG_CATEGORY_INIT (audio_convert_debug, "audioconvert", 0, "audio conversion element"); \
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GST_DEBUG_CATEGORY_GET (GST_CAT_PERFORMANCE, "GST_PERFORMANCE");
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#define gst_audio_convert_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstAudioConvert, gst_audio_convert,
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GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
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/*** GSTREAMER PROTOTYPES *****************************************************/
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#define STATIC_CAPS \
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GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL) \
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", layout = (string) interleaved")
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static GstStaticPadTemplate gst_audio_convert_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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STATIC_CAPS);
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static GstStaticPadTemplate gst_audio_convert_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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STATIC_CAPS);
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/*** TYPE FUNCTIONS ***********************************************************/
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static void
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gst_audio_convert_class_init (GstAudioConvertClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstBaseTransformClass *basetransform_class = GST_BASE_TRANSFORM_CLASS (klass);
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gobject_class->dispose = gst_audio_convert_dispose;
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gobject_class->set_property = gst_audio_convert_set_property;
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gobject_class->get_property = gst_audio_convert_get_property;
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g_object_class_install_property (gobject_class, PROP_DITHERING,
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g_param_spec_enum ("dithering", "Dithering",
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"Selects between different dithering methods.",
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GST_TYPE_AUDIO_DITHER_METHOD, GST_AUDIO_DITHER_TPDF,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_NOISE_SHAPING,
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g_param_spec_enum ("noise-shaping", "Noise shaping",
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"Selects between different noise shaping methods.",
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GST_TYPE_AUDIO_NOISE_SHAPING_METHOD, GST_AUDIO_NOISE_SHAPING_NONE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_MIX_MATRIX,
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gst_param_spec_array ("mix-matrix",
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"Input/output channel matrix",
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"Transformation matrix for input/output channels",
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gst_param_spec_array ("matrix-rows", "rows", "rows",
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g_param_spec_float ("matrix-cols", "cols", "cols",
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-1, 1, 0,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS),
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS),
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_static_pad_template (element_class,
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&gst_audio_convert_src_template);
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gst_element_class_add_static_pad_template (element_class,
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&gst_audio_convert_sink_template);
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gst_element_class_set_static_metadata (element_class, "Audio converter",
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"Filter/Converter/Audio", "Convert audio to different formats",
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"Benjamin Otte <otte@gnome.org>");
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basetransform_class->get_unit_size =
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GST_DEBUG_FUNCPTR (gst_audio_convert_get_unit_size);
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basetransform_class->transform_caps =
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GST_DEBUG_FUNCPTR (gst_audio_convert_transform_caps);
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basetransform_class->fixate_caps =
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GST_DEBUG_FUNCPTR (gst_audio_convert_fixate_caps);
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basetransform_class->set_caps =
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GST_DEBUG_FUNCPTR (gst_audio_convert_set_caps);
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basetransform_class->transform =
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GST_DEBUG_FUNCPTR (gst_audio_convert_transform);
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basetransform_class->transform_ip =
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GST_DEBUG_FUNCPTR (gst_audio_convert_transform_ip);
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basetransform_class->transform_meta =
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GST_DEBUG_FUNCPTR (gst_audio_convert_transform_meta);
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basetransform_class->submit_input_buffer =
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GST_DEBUG_FUNCPTR (gst_audio_convert_submit_input_buffer);
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basetransform_class->passthrough_on_same_caps = TRUE;
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basetransform_class->transform_ip_on_passthrough = FALSE;
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}
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static void
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gst_audio_convert_init (GstAudioConvert * this)
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{
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this->dither = GST_AUDIO_DITHER_TPDF;
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this->ns = GST_AUDIO_NOISE_SHAPING_NONE;
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g_value_init (&this->mix_matrix, GST_TYPE_ARRAY);
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gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (this), TRUE);
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}
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static void
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gst_audio_convert_dispose (GObject * obj)
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{
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GstAudioConvert *this = GST_AUDIO_CONVERT (obj);
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if (this->convert) {
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gst_audio_converter_free (this->convert);
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this->convert = NULL;
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}
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g_value_unset (&this->mix_matrix);
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G_OBJECT_CLASS (parent_class)->dispose (obj);
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}
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/*** GSTREAMER FUNCTIONS ******************************************************/
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/* BaseTransform vmethods */
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static gboolean
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gst_audio_convert_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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g_assert (size);
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if (!gst_audio_info_from_caps (&info, caps))
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goto parse_error;
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*size = info.bpf;
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GST_INFO_OBJECT (base, "unit_size = %" G_GSIZE_FORMAT, *size);
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return TRUE;
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parse_error:
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{
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GST_INFO_OBJECT (base, "failed to parse caps to get unit_size");
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return FALSE;
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}
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}
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static gboolean
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remove_format_from_structure (GstCapsFeatures * features,
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GstStructure * structure, gpointer user_data G_GNUC_UNUSED)
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{
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gst_structure_remove_field (structure, "format");
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return TRUE;
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}
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static gboolean
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remove_channels_from_structure (GstCapsFeatures * features, GstStructure * s,
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gpointer user_data)
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{
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guint64 mask;
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gint channels;
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GstAudioConvert *this = GST_AUDIO_CONVERT (user_data);
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/* Only remove the channels and channel-mask for non-NONE layouts,
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* or if a mix matrix was manually specified */
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if (this->mix_matrix_was_set ||
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!gst_structure_get (s, "channel-mask", GST_TYPE_BITMASK, &mask, NULL) ||
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(mask != 0 || (gst_structure_get_int (s, "channels", &channels)
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&& channels == 1))) {
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gst_structure_remove_fields (s, "channel-mask", "channels", 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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add_other_channels_to_structure (GstCapsFeatures * features, GstStructure * s,
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gpointer user_data)
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{
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gint other_channels = GPOINTER_TO_INT (user_data);
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gst_structure_set (s, "channels", G_TYPE_INT, other_channels, NULL);
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return TRUE;
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}
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/* The caps can be transformed into any other caps with format info removed.
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* However, we should prefer passthrough, so if passthrough is possible,
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* put it first in the list. */
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static GstCaps *
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gst_audio_convert_transform_caps (GstBaseTransform * btrans,
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GstPadDirection direction, GstCaps * caps, GstCaps * filter)
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{
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GstCaps *tmp, *tmp2;
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GstCaps *result;
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GstAudioConvert *this = GST_AUDIO_CONVERT (btrans);
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tmp = gst_caps_copy (caps);
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gst_caps_map_in_place (tmp, remove_format_from_structure, NULL);
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gst_caps_map_in_place (tmp, remove_channels_from_structure, btrans);
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/* We can infer the required input / output channels based on the
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* matrix dimensions */
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if (gst_value_array_get_size (&this->mix_matrix)) {
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gint other_channels;
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if (direction == GST_PAD_SRC) {
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const GValue *first_row =
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gst_value_array_get_value (&this->mix_matrix, 0);
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other_channels = gst_value_array_get_size (first_row);
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} else {
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other_channels = gst_value_array_get_size (&this->mix_matrix);
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}
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gst_caps_map_in_place (tmp, add_other_channels_to_structure,
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GINT_TO_POINTER (other_channels));
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}
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if (filter) {
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tmp2 = gst_caps_intersect_full (filter, tmp, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (tmp);
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tmp = tmp2;
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}
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result = tmp;
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GST_DEBUG_OBJECT (btrans, "transformed %" GST_PTR_FORMAT " into %"
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GST_PTR_FORMAT, caps, result);
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return result;
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}
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/* Count the number of bits set
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* Optimized for the common case, assuming that the number of channels
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* (i.e. bits set) is small
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*/
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static gint
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n_bits_set (guint64 x)
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{
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gint c;
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for (c = 0; x; c++)
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x &= x - 1;
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return c;
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}
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/* Reduce the mask to the n_chans lowest set bits
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*
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* The algorithm clears the n_chans lowest set bits and subtracts the
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* result from the original mask to get the desired mask.
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* It is optimized for the common case where n_chans is a small
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* number. In the worst case, however, it stops after 64 iterations.
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*/
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static guint64
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find_suitable_mask (guint64 mask, gint n_chans)
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{
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guint64 x = mask;
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for (; x && n_chans; n_chans--)
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x &= x - 1;
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g_assert (x || n_chans == 0);
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/* assertion fails if mask contained less bits than n_chans
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* or n_chans was < 0 */
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return mask - x;
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}
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static void
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gst_audio_convert_fixate_format (GstBaseTransform * base, GstStructure * ins,
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GstStructure * outs)
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{
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const gchar *in_format;
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const GValue *format;
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const GstAudioFormatInfo *in_info, *out_info = NULL;
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GstAudioFormatFlags in_flags, out_flags = 0;
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gint in_depth, out_depth = -1;
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gint i, len;
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in_format = gst_structure_get_string (ins, "format");
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if (!in_format)
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return;
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format = gst_structure_get_value (outs, "format");
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/* should not happen */
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if (format == NULL)
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return;
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/* nothing to fixate? */
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if (!GST_VALUE_HOLDS_LIST (format))
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return;
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in_info =
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gst_audio_format_get_info (gst_audio_format_from_string (in_format));
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if (!in_info)
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return;
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in_flags = GST_AUDIO_FORMAT_INFO_FLAGS (in_info);
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in_flags &= ~(GST_AUDIO_FORMAT_FLAG_UNPACK);
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in_flags &= ~(GST_AUDIO_FORMAT_FLAG_SIGNED);
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in_depth = GST_AUDIO_FORMAT_INFO_DEPTH (in_info);
|
|
|
|
len = gst_value_list_get_size (format);
|
|
for (i = 0; i < len; i++) {
|
|
const GstAudioFormatInfo *t_info;
|
|
GstAudioFormatFlags t_flags;
|
|
gboolean t_flags_better;
|
|
const GValue *val;
|
|
const gchar *fname;
|
|
gint t_depth;
|
|
|
|
val = gst_value_list_get_value (format, i);
|
|
if (!G_VALUE_HOLDS_STRING (val))
|
|
continue;
|
|
|
|
fname = g_value_get_string (val);
|
|
t_info = gst_audio_format_get_info (gst_audio_format_from_string (fname));
|
|
if (!t_info)
|
|
continue;
|
|
|
|
/* accept input format immediately */
|
|
if (strcmp (fname, in_format) == 0) {
|
|
out_info = t_info;
|
|
break;
|
|
}
|
|
|
|
t_flags = GST_AUDIO_FORMAT_INFO_FLAGS (t_info);
|
|
t_flags &= ~(GST_AUDIO_FORMAT_FLAG_UNPACK);
|
|
t_flags &= ~(GST_AUDIO_FORMAT_FLAG_SIGNED);
|
|
|
|
t_depth = GST_AUDIO_FORMAT_INFO_DEPTH (t_info);
|
|
|
|
/* Any output format is better than no output format at all */
|
|
if (!out_info) {
|
|
out_info = t_info;
|
|
out_depth = t_depth;
|
|
out_flags = t_flags;
|
|
continue;
|
|
}
|
|
|
|
t_flags_better = (t_flags == in_flags && out_flags != in_flags);
|
|
|
|
if (t_depth == in_depth && (out_depth != in_depth || t_flags_better)) {
|
|
/* Prefer to use the first format that has the same depth with the same
|
|
* flags, and if none with the same flags exist use the first other one
|
|
* that has the same depth */
|
|
out_info = t_info;
|
|
out_depth = t_depth;
|
|
out_flags = t_flags;
|
|
} else if (t_depth >= in_depth && (in_depth > out_depth
|
|
|| (out_depth >= in_depth && t_flags_better))) {
|
|
/* Otherwise use the first format that has a higher depth with the same flags,
|
|
* if none with the same flags exist use the first other one that has a higher
|
|
* depth */
|
|
out_info = t_info;
|
|
out_depth = t_depth;
|
|
out_flags = t_flags;
|
|
} else if ((t_depth > out_depth && out_depth < in_depth)
|
|
|| (t_flags_better && out_depth == t_depth)) {
|
|
/* Else get at least the one with the highest depth, ideally with the same flags */
|
|
out_info = t_info;
|
|
out_depth = t_depth;
|
|
out_flags = t_flags;
|
|
}
|
|
|
|
}
|
|
|
|
if (out_info)
|
|
gst_structure_set (outs, "format", G_TYPE_STRING,
|
|
GST_AUDIO_FORMAT_INFO_NAME (out_info), NULL);
|
|
}
|
|
|
|
static void
|
|
gst_audio_convert_fixate_channels (GstBaseTransform * base, GstStructure * ins,
|
|
GstStructure * outs)
|
|
{
|
|
gint in_chans, out_chans;
|
|
guint64 in_mask = 0, out_mask = 0;
|
|
gboolean has_in_mask = FALSE, has_out_mask = FALSE;
|
|
|
|
if (!gst_structure_get_int (ins, "channels", &in_chans))
|
|
return; /* this shouldn't really happen, should it? */
|
|
|
|
if (!gst_structure_has_field (outs, "channels")) {
|
|
/* we could try to get the implied number of channels from the layout,
|
|
* but that seems overdoing it for a somewhat exotic corner case */
|
|
gst_structure_remove_field (outs, "channel-mask");
|
|
return;
|
|
}
|
|
|
|
/* ok, let's fixate the channels if they are not fixated yet */
|
|
gst_structure_fixate_field_nearest_int (outs, "channels", in_chans);
|
|
|
|
if (!gst_structure_get_int (outs, "channels", &out_chans)) {
|
|
/* shouldn't really happen ... */
|
|
gst_structure_remove_field (outs, "channel-mask");
|
|
return;
|
|
}
|
|
|
|
/* get the channel layout of the output if any */
|
|
has_out_mask = gst_structure_has_field (outs, "channel-mask");
|
|
if (has_out_mask) {
|
|
gst_structure_get (outs, "channel-mask", GST_TYPE_BITMASK, &out_mask, NULL);
|
|
} else {
|
|
/* channels == 1 => MONO */
|
|
if (out_chans == 2) {
|
|
out_mask =
|
|
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) |
|
|
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT);
|
|
has_out_mask = TRUE;
|
|
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, out_mask,
|
|
NULL);
|
|
}
|
|
}
|
|
|
|
/* get the channel layout of the input if any */
|
|
has_in_mask = gst_structure_has_field (ins, "channel-mask");
|
|
if (has_in_mask) {
|
|
gst_structure_get (ins, "channel-mask", GST_TYPE_BITMASK, &in_mask, NULL);
|
|
} else {
|
|
/* channels == 1 => MONO */
|
|
if (in_chans == 2) {
|
|
in_mask =
|
|
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) |
|
|
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT);
|
|
has_in_mask = TRUE;
|
|
} else if (in_chans > 2)
|
|
g_warning ("%s: Upstream caps contain no channel mask",
|
|
GST_ELEMENT_NAME (base));
|
|
}
|
|
|
|
if (!has_out_mask && out_chans == 1 && (in_chans != out_chans
|
|
|| !has_in_mask))
|
|
return; /* nothing to do, default layout will be assumed */
|
|
|
|
if (in_chans == out_chans && (has_in_mask || in_chans == 1)) {
|
|
/* same number of channels and no output layout: just use input layout */
|
|
if (!has_out_mask) {
|
|
/* in_chans == 1 handled above already */
|
|
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, in_mask, NULL);
|
|
return;
|
|
}
|
|
|
|
/* If both masks are the same we're done, this includes the NONE layout case */
|
|
if (in_mask == out_mask)
|
|
return;
|
|
|
|
/* if output layout is fixed already and looks sane, we're done */
|
|
if (n_bits_set (out_mask) == out_chans)
|
|
return;
|
|
|
|
if (n_bits_set (out_mask) < in_chans) {
|
|
/* Not much we can do here, this shouldn't just happen */
|
|
g_warning ("%s: Invalid downstream channel-mask with too few bits set",
|
|
GST_ELEMENT_NAME (base));
|
|
} else {
|
|
guint64 intersection;
|
|
|
|
/* if the output layout is not fixed, check if the output layout contains
|
|
* the input layout */
|
|
intersection = in_mask & out_mask;
|
|
if (n_bits_set (intersection) >= in_chans) {
|
|
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, in_mask,
|
|
NULL);
|
|
return;
|
|
}
|
|
|
|
/* output layout is not fixed and does not contain the input layout, so
|
|
* just pick the first possibility */
|
|
intersection = find_suitable_mask (out_mask, out_chans);
|
|
if (intersection) {
|
|
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, intersection,
|
|
NULL);
|
|
return;
|
|
}
|
|
}
|
|
|
|
/* ... else fall back to default layout (NB: out_layout is NULL here) */
|
|
GST_WARNING_OBJECT (base, "unexpected output channel layout");
|
|
} else {
|
|
guint64 intersection;
|
|
|
|
/* number of input channels != number of output channels:
|
|
* if this value contains a list of channel layouts (or even worse: a list
|
|
* with another list), just pick the first value and repeat until we find a
|
|
* channel position array or something else that's not a list; we assume
|
|
* the input if half-way sane and don't try to fall back on other list items
|
|
* if the first one is something unexpected or non-channel-pos-array-y */
|
|
if (n_bits_set (out_mask) >= out_chans) {
|
|
intersection = find_suitable_mask (out_mask, out_chans);
|
|
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, intersection,
|
|
NULL);
|
|
return;
|
|
}
|
|
|
|
/* what now?! Just ignore what we're given and use default positions */
|
|
GST_WARNING_OBJECT (base, "invalid or unexpected channel-positions");
|
|
}
|
|
|
|
/* missing or invalid output layout and we can't use the input layout for
|
|
* one reason or another, so just pick a default layout (we could be smarter
|
|
* and try to add/remove channels from the input layout, or pick a default
|
|
* layout based on LFE-presence in input layout, but let's save that for
|
|
* another day). For mono, no mask is required and the fallback mask is 0 */
|
|
if (out_chans > 1
|
|
&& (out_mask = gst_audio_channel_get_fallback_mask (out_chans))) {
|
|
GST_DEBUG_OBJECT (base, "using default channel layout as fallback");
|
|
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, out_mask, NULL);
|
|
} else if (out_chans > 1) {
|
|
GST_ERROR_OBJECT (base, "Have no default layout for %d channels",
|
|
out_chans);
|
|
}
|
|
}
|
|
|
|
/* try to keep as many of the structure members the same by fixating the
|
|
* possible ranges; this way we convert the least amount of things as possible
|
|
*/
|
|
static GstCaps *
|
|
gst_audio_convert_fixate_caps (GstBaseTransform * base,
|
|
GstPadDirection direction, GstCaps * caps, GstCaps * othercaps)
|
|
{
|
|
GstStructure *ins, *outs;
|
|
GstCaps *result;
|
|
|
|
GST_DEBUG_OBJECT (base, "trying to fixate othercaps %" GST_PTR_FORMAT
|
|
" based on caps %" GST_PTR_FORMAT, othercaps, caps);
|
|
|
|
result = gst_caps_intersect (othercaps, caps);
|
|
if (gst_caps_is_empty (result)) {
|
|
GstCaps *removed = gst_caps_copy (caps);
|
|
|
|
if (result)
|
|
gst_caps_unref (result);
|
|
gst_caps_map_in_place (removed, remove_format_from_structure, NULL);
|
|
result = gst_caps_intersect (othercaps, removed);
|
|
gst_caps_unref (removed);
|
|
if (gst_caps_is_empty (result)) {
|
|
if (result)
|
|
gst_caps_unref (result);
|
|
result = othercaps;
|
|
} else {
|
|
gst_caps_unref (othercaps);
|
|
}
|
|
} else {
|
|
gst_caps_unref (othercaps);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (base, "now fixating %" GST_PTR_FORMAT, result);
|
|
|
|
/* fixate remaining fields */
|
|
result = gst_caps_make_writable (result);
|
|
|
|
ins = gst_caps_get_structure (caps, 0);
|
|
outs = gst_caps_get_structure (result, 0);
|
|
|
|
gst_audio_convert_fixate_channels (base, ins, outs);
|
|
gst_audio_convert_fixate_format (base, ins, outs);
|
|
|
|
/* fixate remaining */
|
|
result = gst_caps_fixate (result);
|
|
|
|
GST_DEBUG_OBJECT (base, "fixated othercaps to %" GST_PTR_FORMAT, result);
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_audio_convert_set_caps (GstBaseTransform * base, GstCaps * incaps,
|
|
GstCaps * outcaps)
|
|
{
|
|
GstAudioConvert *this = GST_AUDIO_CONVERT (base);
|
|
GstAudioInfo in_info;
|
|
GstAudioInfo out_info;
|
|
gboolean in_place;
|
|
GstStructure *config;
|
|
|
|
GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %"
|
|
GST_PTR_FORMAT, incaps, outcaps);
|
|
|
|
if (this->convert) {
|
|
gst_audio_converter_free (this->convert);
|
|
this->convert = NULL;
|
|
}
|
|
|
|
if (!gst_audio_info_from_caps (&in_info, incaps))
|
|
goto invalid_in;
|
|
if (!gst_audio_info_from_caps (&out_info, outcaps))
|
|
goto invalid_out;
|
|
|
|
config = gst_structure_new ("GstAudioConverterConfig",
|
|
GST_AUDIO_CONVERTER_OPT_DITHER_METHOD, GST_TYPE_AUDIO_DITHER_METHOD,
|
|
this->dither,
|
|
GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD,
|
|
GST_TYPE_AUDIO_NOISE_SHAPING_METHOD, this->ns, NULL);
|
|
|
|
if (this->mix_matrix_was_set)
|
|
gst_structure_set_value (config, GST_AUDIO_CONVERTER_OPT_MIX_MATRIX,
|
|
&this->mix_matrix);
|
|
|
|
this->convert = gst_audio_converter_new (0, &in_info, &out_info, config);
|
|
|
|
if (this->convert == NULL)
|
|
goto no_converter;
|
|
|
|
in_place = gst_audio_converter_supports_inplace (this->convert);
|
|
gst_base_transform_set_in_place (base, in_place);
|
|
|
|
this->in_info = in_info;
|
|
this->out_info = out_info;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
invalid_in:
|
|
{
|
|
GST_ERROR_OBJECT (base, "invalid input caps");
|
|
return FALSE;
|
|
}
|
|
invalid_out:
|
|
{
|
|
GST_ERROR_OBJECT (base, "invalid output caps");
|
|
return FALSE;
|
|
}
|
|
no_converter:
|
|
{
|
|
GST_ERROR_OBJECT (base, "could not make converter");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* if called through gst_audio_convert_transform_ip() inbuf == outbuf */
|
|
static GstFlowReturn
|
|
gst_audio_convert_transform (GstBaseTransform * base, GstBuffer * inbuf,
|
|
GstBuffer * outbuf)
|
|
{
|
|
GstFlowReturn ret;
|
|
GstAudioConvert *this = GST_AUDIO_CONVERT (base);
|
|
GstMapInfo srcmap = { NULL, }, dstmap;
|
|
gint insize, outsize;
|
|
gboolean inbuf_writable;
|
|
GstAudioConverterFlags flags;
|
|
gsize samples;
|
|
|
|
/* get amount of samples to convert. */
|
|
samples = gst_buffer_get_size (inbuf) / this->in_info.bpf;
|
|
|
|
/* get in/output sizes, to see if the buffers we got are of correct
|
|
* sizes */
|
|
insize = samples * this->in_info.bpf;
|
|
outsize = samples * this->out_info.bpf;
|
|
|
|
if (insize == 0 || outsize == 0)
|
|
return GST_FLOW_OK;
|
|
|
|
gst_buffer_resize (outbuf, 0, outsize);
|
|
|
|
/* get src and dst data */
|
|
if (inbuf != outbuf) {
|
|
inbuf_writable = gst_buffer_is_writable (inbuf)
|
|
&& gst_buffer_n_memory (inbuf) == 1
|
|
&& gst_memory_is_writable (gst_buffer_peek_memory (inbuf, 0));
|
|
|
|
if (!gst_buffer_map (inbuf, &srcmap,
|
|
inbuf_writable ? GST_MAP_READWRITE : GST_MAP_READ))
|
|
goto inmap_error;
|
|
} else {
|
|
inbuf_writable = TRUE;
|
|
}
|
|
if (!gst_buffer_map (outbuf, &dstmap, GST_MAP_WRITE))
|
|
goto outmap_error;
|
|
|
|
/* check in and outsize */
|
|
if (inbuf != outbuf) {
|
|
if (srcmap.size < insize)
|
|
goto wrong_size;
|
|
}
|
|
if (dstmap.size < outsize)
|
|
goto wrong_size;
|
|
|
|
/* and convert the samples */
|
|
flags = 0;
|
|
if (inbuf_writable)
|
|
flags |= GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE;
|
|
|
|
if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
|
|
gpointer in[1] = { srcmap.data };
|
|
gpointer out[1] = { dstmap.data };
|
|
|
|
if (!gst_audio_converter_samples (this->convert, flags,
|
|
inbuf != outbuf ? in : out, samples, out, samples))
|
|
goto convert_error;
|
|
} else {
|
|
/* Create silence buffer */
|
|
gst_audio_format_fill_silence (this->out_info.finfo, dstmap.data, outsize);
|
|
}
|
|
ret = GST_FLOW_OK;
|
|
|
|
done:
|
|
gst_buffer_unmap (outbuf, &dstmap);
|
|
if (inbuf != outbuf)
|
|
gst_buffer_unmap (inbuf, &srcmap);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
wrong_size:
|
|
{
|
|
GST_ELEMENT_ERROR (this, STREAM, FORMAT,
|
|
(NULL),
|
|
("input/output buffers are of wrong size in: %" G_GSIZE_FORMAT " < %d"
|
|
" or out: %" G_GSIZE_FORMAT " < %d",
|
|
srcmap.size, insize, dstmap.size, outsize));
|
|
ret = GST_FLOW_ERROR;
|
|
goto done;
|
|
}
|
|
convert_error:
|
|
{
|
|
GST_ELEMENT_ERROR (this, STREAM, FORMAT,
|
|
(NULL), ("error while converting"));
|
|
ret = GST_FLOW_ERROR;
|
|
goto done;
|
|
}
|
|
inmap_error:
|
|
{
|
|
GST_ELEMENT_ERROR (this, STREAM, FORMAT,
|
|
(NULL), ("failed to map input buffer"));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
outmap_error:
|
|
{
|
|
GST_ELEMENT_ERROR (this, STREAM, FORMAT,
|
|
(NULL), ("failed to map output buffer"));
|
|
if (inbuf != outbuf)
|
|
gst_buffer_unmap (inbuf, &srcmap);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_audio_convert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
|
|
{
|
|
return gst_audio_convert_transform (base, buf, buf);
|
|
}
|
|
|
|
static gboolean
|
|
gst_audio_convert_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_convert_submit_input_buffer (GstBaseTransform * base,
|
|
gboolean is_discont, GstBuffer * input)
|
|
{
|
|
GstAudioConvert *this = GST_AUDIO_CONVERT (base);
|
|
|
|
if (base->segment.format == GST_FORMAT_TIME) {
|
|
input =
|
|
gst_audio_buffer_clip (input, &base->segment, this->in_info.rate,
|
|
this->in_info.bpf);
|
|
|
|
if (!input)
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
return GST_BASE_TRANSFORM_CLASS (parent_class)->submit_input_buffer (base,
|
|
is_discont, input);
|
|
}
|
|
|
|
static void
|
|
gst_audio_convert_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstAudioConvert *this = GST_AUDIO_CONVERT (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_DITHERING:
|
|
this->dither = g_value_get_enum (value);
|
|
break;
|
|
case PROP_NOISE_SHAPING:
|
|
this->ns = g_value_get_enum (value);
|
|
break;
|
|
case PROP_MIX_MATRIX:
|
|
if (!gst_value_array_get_size (value)) {
|
|
g_value_copy (value, &this->mix_matrix);
|
|
this->mix_matrix_was_set = TRUE;
|
|
} else {
|
|
const GValue *first_row = gst_value_array_get_value (value, 0);
|
|
|
|
if (gst_value_array_get_size (first_row)) {
|
|
if (gst_value_array_get_size (&this->mix_matrix))
|
|
g_value_unset (&this->mix_matrix);
|
|
|
|
g_value_copy (value, &this->mix_matrix);
|
|
this->mix_matrix_was_set = TRUE;
|
|
} else {
|
|
g_warning ("Empty mix matrix's first row");
|
|
}
|
|
}
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_audio_convert_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstAudioConvert *this = GST_AUDIO_CONVERT (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_DITHERING:
|
|
g_value_set_enum (value, this->dither);
|
|
break;
|
|
case PROP_NOISE_SHAPING:
|
|
g_value_set_enum (value, this->ns);
|
|
break;
|
|
case PROP_MIX_MATRIX:
|
|
if (this->mix_matrix_was_set)
|
|
g_value_copy (&this->mix_matrix, value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|