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Using the max 120 ms buffer size to ensure we have enough space for decoded data meant that Opus could actually return 120 ms' worth of data. https://bugzilla.gnome.org/show_bug.cgi?id=771723
1035 lines
31 KiB
C
1035 lines
31 KiB
C
/* GStreamer
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* Copyright (C) 2004 Wim Taymans <wim@fluendo.com>
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* Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
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* Copyright (C) 2008 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* Copyright (C) 2011-2012 Vincent Penquerc'h <vincent.penquerch@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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* Based on the speexdec element.
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*/
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/**
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* SECTION:element-opusdec
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* @see_also: opusenc, oggdemux
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*
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* This element decodes a OPUS stream to raw integer audio.
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch-1.0 -v filesrc location=opus.ogg ! oggdemux ! opusdec ! audioconvert ! audioresample ! alsasink
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* ]| Decode an Ogg/Opus file. To create an Ogg/Opus file refer to the documentation of opusenc.
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* </refsect2>
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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 <math.h>
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#include <string.h>
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#include <stdio.h>
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#include "gstopusheader.h"
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#include "gstopuscommon.h"
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#include "gstopusdec.h"
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#include <gst/pbutils/pbutils.h>
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GST_DEBUG_CATEGORY_STATIC (opusdec_debug);
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#define GST_CAT_DEFAULT opusdec_debug
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static GstStaticPadTemplate opus_dec_src_factory =
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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 ("audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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"rate = (int) { 48000, 24000, 16000, 12000, 8000 }, "
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"channels = (int) [ 1, 8 ] ")
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);
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static GstStaticPadTemplate opus_dec_sink_factory =
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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 ("audio/x-opus, "
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"channel-mapping-family = (int) 0; "
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"audio/x-opus, "
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"channel-mapping-family = (int) [1, 255], "
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"channels = (int) [1, 255], "
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"stream-count = (int) [1, 255], " "coupled-count = (int) [0, 255]")
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);
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G_DEFINE_TYPE (GstOpusDec, gst_opus_dec, GST_TYPE_AUDIO_DECODER);
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#define DB_TO_LINEAR(x) pow (10., (x) / 20.)
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#define DEFAULT_USE_INBAND_FEC FALSE
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#define DEFAULT_APPLY_GAIN TRUE
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enum
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{
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PROP_0,
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PROP_USE_INBAND_FEC,
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PROP_APPLY_GAIN
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};
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static GstFlowReturn gst_opus_dec_parse_header (GstOpusDec * dec,
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GstBuffer * buf);
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static gboolean gst_opus_dec_start (GstAudioDecoder * dec);
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static gboolean gst_opus_dec_stop (GstAudioDecoder * dec);
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static GstFlowReturn gst_opus_dec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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static gboolean gst_opus_dec_set_format (GstAudioDecoder * bdec,
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GstCaps * caps);
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static void gst_opus_dec_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_opus_dec_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static GstCaps *gst_opus_dec_getcaps (GstAudioDecoder * dec, GstCaps * filter);
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static void
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gst_opus_dec_class_init (GstOpusDecClass * klass)
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{
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GObjectClass *gobject_class;
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GstAudioDecoderClass *adclass;
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GstElementClass *element_class;
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gobject_class = (GObjectClass *) klass;
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adclass = (GstAudioDecoderClass *) klass;
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element_class = (GstElementClass *) klass;
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gobject_class->set_property = gst_opus_dec_set_property;
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gobject_class->get_property = gst_opus_dec_get_property;
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adclass->start = GST_DEBUG_FUNCPTR (gst_opus_dec_start);
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adclass->stop = GST_DEBUG_FUNCPTR (gst_opus_dec_stop);
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adclass->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_dec_handle_frame);
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adclass->set_format = GST_DEBUG_FUNCPTR (gst_opus_dec_set_format);
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adclass->getcaps = GST_DEBUG_FUNCPTR (gst_opus_dec_getcaps);
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gst_element_class_add_static_pad_template (element_class,
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&opus_dec_src_factory);
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gst_element_class_add_static_pad_template (element_class,
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&opus_dec_sink_factory);
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gst_element_class_set_static_metadata (element_class, "Opus audio decoder",
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"Codec/Decoder/Audio", "decode opus streams to audio",
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"Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>");
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g_object_class_install_property (gobject_class, PROP_USE_INBAND_FEC,
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g_param_spec_boolean ("use-inband-fec", "Use in-band FEC",
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"Use forward error correction if available (needs PLC enabled)",
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DEFAULT_USE_INBAND_FEC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_APPLY_GAIN,
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g_param_spec_boolean ("apply-gain", "Apply gain",
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"Apply gain if any is specified in the header", DEFAULT_APPLY_GAIN,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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GST_DEBUG_CATEGORY_INIT (opusdec_debug, "opusdec", 0,
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"opus decoding element");
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}
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static void
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gst_opus_dec_reset (GstOpusDec * dec)
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{
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dec->packetno = 0;
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if (dec->state) {
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opus_multistream_decoder_destroy (dec->state);
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dec->state = NULL;
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}
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gst_buffer_replace (&dec->streamheader, NULL);
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gst_buffer_replace (&dec->vorbiscomment, NULL);
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gst_buffer_replace (&dec->last_buffer, NULL);
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dec->primed = FALSE;
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dec->pre_skip = 0;
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dec->r128_gain = 0;
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dec->sample_rate = 0;
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dec->n_channels = 0;
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dec->leftover_plc_duration = 0;
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dec->last_known_buffer_duration = GST_CLOCK_TIME_NONE;
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}
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static void
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gst_opus_dec_init (GstOpusDec * dec)
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{
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dec->use_inband_fec = FALSE;
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dec->apply_gain = DEFAULT_APPLY_GAIN;
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gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
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gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
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(dec), TRUE);
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
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gst_opus_dec_reset (dec);
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}
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static gboolean
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gst_opus_dec_start (GstAudioDecoder * dec)
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{
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GstOpusDec *odec = GST_OPUS_DEC (dec);
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gst_opus_dec_reset (odec);
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/* we know about concealment */
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gst_audio_decoder_set_plc_aware (dec, TRUE);
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if (odec->use_inband_fec) {
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/* opusdec outputs samples directly from an input buffer, except if
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* FEC is on, in which case it buffers one buffer in case one buffer
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* goes missing.
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*/
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gst_audio_decoder_set_latency (dec, 120 * GST_MSECOND, 120 * GST_MSECOND);
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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_opus_dec_stop (GstAudioDecoder * dec)
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{
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GstOpusDec *odec = GST_OPUS_DEC (dec);
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gst_opus_dec_reset (odec);
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return TRUE;
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}
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static double
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gst_opus_dec_get_r128_gain (gint16 r128_gain)
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{
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return r128_gain / (double) (1 << 8);
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}
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static double
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gst_opus_dec_get_r128_volume (gint16 r128_gain)
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{
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return DB_TO_LINEAR (gst_opus_dec_get_r128_gain (r128_gain));
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}
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static gboolean
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gst_opus_dec_negotiate (GstOpusDec * dec, const GstAudioChannelPosition * pos)
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{
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GstCaps *caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
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GstStructure *s;
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GstAudioInfo info;
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if (caps) {
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gint rate = dec->sample_rate, channels = dec->n_channels;
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GstCaps *constraint, *inter;
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constraint = gst_caps_from_string ("audio/x-raw");
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if (dec->n_channels <= 2) { /* including 0 */
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gst_caps_set_simple (constraint, "channels", GST_TYPE_INT_RANGE, 1, 2,
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NULL);
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} else {
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gst_caps_set_simple (constraint, "channels", G_TYPE_INT, dec->n_channels,
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NULL);
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}
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inter = gst_caps_intersect (caps, constraint);
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gst_caps_unref (constraint);
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if (gst_caps_is_empty (inter)) {
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GST_DEBUG_OBJECT (dec, "Empty intersection, failed to negotiate");
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gst_caps_unref (inter);
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gst_caps_unref (caps);
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return FALSE;
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}
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inter = gst_caps_truncate (inter);
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s = gst_caps_get_structure (inter, 0);
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rate = dec->sample_rate > 0 ? dec->sample_rate : 48000;
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gst_structure_fixate_field_nearest_int (s, "rate", dec->sample_rate);
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gst_structure_get_int (s, "rate", &rate);
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channels = dec->n_channels > 0 ? dec->n_channels : 2;
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gst_structure_fixate_field_nearest_int (s, "channels", dec->n_channels);
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gst_structure_get_int (s, "channels", &channels);
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gst_caps_unref (inter);
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dec->sample_rate = rate;
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dec->n_channels = channels;
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gst_caps_unref (caps);
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}
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if (dec->n_channels == 0) {
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GST_DEBUG_OBJECT (dec, "Using a default of 2 channels");
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dec->n_channels = 2;
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pos = NULL;
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}
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if (dec->sample_rate == 0) {
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GST_DEBUG_OBJECT (dec, "Using a default of 48kHz sample rate");
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dec->sample_rate = 48000;
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}
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GST_INFO_OBJECT (dec, "Negotiated %d channels, %d Hz", dec->n_channels,
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dec->sample_rate);
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/* pass valid order to audio info */
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if (pos) {
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memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
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gst_audio_channel_positions_to_valid_order (dec->opus_pos, dec->n_channels);
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}
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/* set up source format */
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gst_audio_info_init (&info);
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gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16,
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dec->sample_rate, dec->n_channels, pos ? dec->opus_pos : NULL);
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gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info);
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/* but we still need the opus order for later reordering */
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if (pos) {
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memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
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} else {
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dec->opus_pos[0] = GST_AUDIO_CHANNEL_POSITION_INVALID;
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}
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dec->info = info;
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return TRUE;
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}
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static GstFlowReturn
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gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
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{
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GstAudioChannelPosition pos[64];
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const GstAudioChannelPosition *posn = NULL;
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if (!gst_opus_header_is_id_header (buf)) {
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GST_ELEMENT_ERROR (dec, STREAM, FORMAT, (NULL),
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("Header is not an Opus ID header"));
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return GST_FLOW_ERROR;
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}
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if (!gst_codec_utils_opus_parse_header (buf,
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&dec->sample_rate,
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&dec->n_channels,
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&dec->channel_mapping_family,
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&dec->n_streams,
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&dec->n_stereo_streams,
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dec->channel_mapping, &dec->pre_skip, &dec->r128_gain)) {
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GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
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("Failed to parse Opus ID header"));
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return GST_FLOW_ERROR;
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}
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dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain);
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GST_INFO_OBJECT (dec,
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"Found pre-skip of %u samples, R128 gain %d (volume %f)",
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dec->pre_skip, dec->r128_gain, dec->r128_gain_volume);
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if (dec->channel_mapping_family == 1) {
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GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
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switch (dec->n_channels) {
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case 1:
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case 2:
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/* nothing */
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break;
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case 3:
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case 4:
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case 5:
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case 6:
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case 7:
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case 8:
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posn = gst_opus_channel_positions[dec->n_channels - 1];
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break;
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default:{
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gint i;
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GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
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(NULL), ("Using NONE channel layout for more than 8 channels"));
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for (i = 0; i < dec->n_channels; i++)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
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posn = pos;
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}
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}
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} else {
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GST_INFO_OBJECT (dec, "Channel mapping family %d",
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dec->channel_mapping_family);
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}
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if (!gst_opus_dec_negotiate (dec, posn))
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return GST_FLOW_NOT_NEGOTIATED;
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return GST_FLOW_OK;
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}
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static GstFlowReturn
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gst_opus_dec_parse_comments (GstOpusDec * dec, GstBuffer * buf)
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{
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return GST_FLOW_OK;
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}
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/* adapted from ext/ogg/gstoggstream.c */
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static gint64
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packet_duration_opus (const unsigned char *data, size_t bytes)
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{
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static const guint64 durations[32] = {
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480, 960, 1920, 2880, /* Silk NB */
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480, 960, 1920, 2880, /* Silk MB */
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480, 960, 1920, 2880, /* Silk WB */
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480, 960, /* Hybrid SWB */
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480, 960, /* Hybrid FB */
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120, 240, 480, 960, /* CELT NB */
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120, 240, 480, 960, /* CELT NB */
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120, 240, 480, 960, /* CELT NB */
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120, 240, 480, 960, /* CELT NB */
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};
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gint64 duration;
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gint64 frame_duration;
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gint nframes = 0;
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guint8 toc;
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if (bytes < 1)
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return 0;
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/* headers */
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if (bytes >= 8 && !memcmp (data, "Opus", 4))
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return 0;
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toc = data[0];
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frame_duration = durations[toc >> 3];
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switch (toc & 3) {
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case 0:
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nframes = 1;
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break;
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case 1:
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nframes = 2;
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break;
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case 2:
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nframes = 2;
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break;
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case 3:
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if (bytes < 2) {
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GST_WARNING ("Code 3 Opus packet has less than 2 bytes");
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return 0;
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}
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nframes = data[1] & 63;
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break;
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}
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duration = nframes * frame_duration;
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if (duration > 5760) {
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GST_WARNING ("Opus packet duration > 120 ms, invalid");
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return 0;
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}
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GST_LOG ("Opus packet: frame size %.1f ms, %d frames, duration %.1f ms",
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frame_duration / 48.f, nframes, duration / 48.f);
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return duration / 48.f * 1000000;
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}
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static GstFlowReturn
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opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
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{
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GstFlowReturn res = GST_FLOW_OK;
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gsize size;
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guint8 *data;
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GstBuffer *outbuf, *bufd;
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gint16 *out_data;
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int n, err;
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int samples;
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unsigned int packet_size;
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GstBuffer *buf;
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GstMapInfo map, omap;
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GstAudioClippingMeta *cmeta = NULL;
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if (dec->state == NULL) {
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/* If we did not get any headers, default to 2 channels */
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if (dec->n_channels == 0) {
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GST_INFO_OBJECT (dec, "No header, assuming single stream");
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dec->n_channels = 2;
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dec->sample_rate = 48000;
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/* default stereo mapping */
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dec->channel_mapping_family = 0;
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dec->channel_mapping[0] = 0;
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dec->channel_mapping[1] = 1;
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dec->n_streams = 1;
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dec->n_stereo_streams = 1;
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if (!gst_opus_dec_negotiate (dec, NULL))
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return GST_FLOW_NOT_NEGOTIATED;
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}
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|
|
if (dec->n_channels == 2 && dec->n_streams == 1
|
|
&& dec->n_stereo_streams == 0) {
|
|
/* if we are automatically decoding 2 channels, but only have
|
|
a single encoded one, direct both channels to it */
|
|
dec->channel_mapping[1] = 0;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
|
|
dec->n_channels, dec->sample_rate);
|
|
#ifndef GST_DISABLE_GST_DEBUG
|
|
gst_opus_common_log_channel_mapping_table (GST_ELEMENT (dec), opusdec_debug,
|
|
"Mapping table", dec->n_channels, dec->channel_mapping);
|
|
#endif
|
|
|
|
GST_DEBUG_OBJECT (dec, "%d streams, %d stereo", dec->n_streams,
|
|
dec->n_stereo_streams);
|
|
dec->state =
|
|
opus_multistream_decoder_create (dec->sample_rate, dec->n_channels,
|
|
dec->n_streams, dec->n_stereo_streams, dec->channel_mapping, &err);
|
|
if (!dec->state || err != OPUS_OK)
|
|
goto creation_failed;
|
|
}
|
|
|
|
if (buffer) {
|
|
GST_DEBUG_OBJECT (dec, "Received buffer of size %" G_GSIZE_FORMAT,
|
|
gst_buffer_get_size (buffer));
|
|
} else {
|
|
GST_DEBUG_OBJECT (dec, "Received missing buffer");
|
|
}
|
|
|
|
/* if using in-band FEC, we introdude one extra frame's delay as we need
|
|
to potentially wait for next buffer to decode a missing buffer */
|
|
if (dec->use_inband_fec && !dec->primed) {
|
|
GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out");
|
|
gst_buffer_replace (&dec->last_buffer, buffer);
|
|
dec->primed = TRUE;
|
|
goto done;
|
|
}
|
|
|
|
/* That's the buffer we'll be sending to the opus decoder. */
|
|
buf = (dec->use_inband_fec
|
|
&& gst_buffer_get_size (dec->last_buffer) >
|
|
0) ? dec->last_buffer : buffer;
|
|
|
|
/* That's the buffer we get duration from */
|
|
bufd = dec->use_inband_fec ? dec->last_buffer : buffer;
|
|
|
|
if (buf && gst_buffer_get_size (buf) > 0) {
|
|
gst_buffer_map (buf, &map, GST_MAP_READ);
|
|
data = map.data;
|
|
size = map.size;
|
|
GST_DEBUG_OBJECT (dec, "Using buffer of size %" G_GSIZE_FORMAT, size);
|
|
} else {
|
|
/* concealment data, pass NULL as the bits parameters */
|
|
GST_DEBUG_OBJECT (dec, "Using NULL buffer");
|
|
data = NULL;
|
|
size = 0;
|
|
}
|
|
|
|
if (gst_buffer_get_size (bufd) == 0) {
|
|
GstClockTime const opus_plc_alignment = 2500 * GST_USECOND;
|
|
GstClockTime aligned_missing_duration;
|
|
GstClockTime missing_duration = GST_BUFFER_DURATION (bufd);
|
|
|
|
if (!GST_CLOCK_TIME_IS_VALID (missing_duration) || missing_duration == 0) {
|
|
if (GST_CLOCK_TIME_IS_VALID (dec->last_known_buffer_duration)) {
|
|
missing_duration = dec->last_known_buffer_duration;
|
|
GST_WARNING_OBJECT (dec,
|
|
"Missing duration, using last duration %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (missing_duration));
|
|
} else {
|
|
GST_WARNING_OBJECT (dec,
|
|
"Missing buffer, but unknown duration, and no previously known duration, assuming 20 ms");
|
|
missing_duration = 20 * GST_MSECOND;
|
|
}
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (dec,
|
|
"missing buffer, doing PLC duration %" GST_TIME_FORMAT
|
|
" plus leftover %" GST_TIME_FORMAT, GST_TIME_ARGS (missing_duration),
|
|
GST_TIME_ARGS (dec->leftover_plc_duration));
|
|
|
|
/* add the leftover PLC duration to that of the buffer */
|
|
missing_duration += dec->leftover_plc_duration;
|
|
|
|
/* align the combined buffer and leftover PLC duration to multiples
|
|
* of 2.5ms, rounding to nearest, and store excess duration for later */
|
|
aligned_missing_duration =
|
|
((missing_duration +
|
|
opus_plc_alignment / 2) / opus_plc_alignment) * opus_plc_alignment;
|
|
dec->leftover_plc_duration = missing_duration - aligned_missing_duration;
|
|
|
|
/* Opus' PLC cannot operate with less than 2.5ms; skip PLC
|
|
* and accumulate the missing duration in the leftover_plc_duration
|
|
* for the next PLC attempt */
|
|
if (aligned_missing_duration < opus_plc_alignment) {
|
|
GST_DEBUG_OBJECT (dec,
|
|
"current duration %" GST_TIME_FORMAT
|
|
" of missing data not enough for PLC (minimum needed: %"
|
|
GST_TIME_FORMAT ") - skipping", GST_TIME_ARGS (missing_duration),
|
|
GST_TIME_ARGS (opus_plc_alignment));
|
|
goto done;
|
|
}
|
|
|
|
/* convert the duration (in nanoseconds) to sample count */
|
|
samples =
|
|
gst_util_uint64_scale_int (aligned_missing_duration, dec->sample_rate,
|
|
GST_SECOND);
|
|
|
|
GST_DEBUG_OBJECT (dec,
|
|
"calculated PLC frame length: %" GST_TIME_FORMAT
|
|
" num frame samples: %d new leftover: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (aligned_missing_duration), samples,
|
|
GST_TIME_ARGS (dec->leftover_plc_duration));
|
|
} else {
|
|
/* use maximum size (120 ms) as the number of returned samples is
|
|
not constant over the stream. */
|
|
samples = 120 * dec->sample_rate / 1000;
|
|
}
|
|
packet_size = samples * dec->n_channels * 2;
|
|
|
|
outbuf =
|
|
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec),
|
|
packet_size);
|
|
if (!outbuf) {
|
|
goto buffer_failed;
|
|
}
|
|
|
|
if (size > 0)
|
|
dec->last_known_buffer_duration = packet_duration_opus (data, size);
|
|
|
|
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
|
|
out_data = (gint16 *) omap.data;
|
|
|
|
do {
|
|
if (dec->use_inband_fec) {
|
|
if (gst_buffer_get_size (dec->last_buffer) > 0) {
|
|
/* normal delayed decode */
|
|
GST_LOG_OBJECT (dec, "FEC enabled, decoding last delayed buffer");
|
|
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
|
|
0);
|
|
} else {
|
|
/* FEC reconstruction decode */
|
|
GST_LOG_OBJECT (dec, "FEC enabled, reconstructing last buffer");
|
|
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
|
|
1);
|
|
}
|
|
} else {
|
|
/* normal decode */
|
|
GST_LOG_OBJECT (dec, "FEC disabled, decoding buffer");
|
|
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
|
|
0);
|
|
}
|
|
if (n == OPUS_BUFFER_TOO_SMALL) {
|
|
/* if too small, add 2.5 milliseconds and try again, up to the
|
|
* Opus max size of 120 milliseconds */
|
|
if (samples >= 120 * dec->sample_rate / 1000)
|
|
break;
|
|
samples += 25 * dec->sample_rate / 10000;
|
|
packet_size = samples * dec->n_channels * 2;
|
|
gst_buffer_unmap (outbuf, &omap);
|
|
gst_buffer_unref (outbuf);
|
|
outbuf =
|
|
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec),
|
|
packet_size);
|
|
if (!outbuf) {
|
|
goto buffer_failed;
|
|
}
|
|
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
|
|
out_data = (gint16 *) omap.data;
|
|
}
|
|
} while (n == OPUS_BUFFER_TOO_SMALL);
|
|
gst_buffer_unmap (outbuf, &omap);
|
|
if (data != NULL)
|
|
gst_buffer_unmap (buf, &map);
|
|
|
|
if (n < 0) {
|
|
GstFlowReturn ret = GST_FLOW_ERROR;
|
|
|
|
gst_buffer_unref (outbuf);
|
|
GST_AUDIO_DECODER_ERROR (dec, 1, STREAM, DECODE, (NULL),
|
|
("Decoding error (%d): %s", n, opus_strerror (n)), ret);
|
|
return ret;
|
|
}
|
|
GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
|
|
gst_buffer_set_size (outbuf, n * 2 * dec->n_channels);
|
|
GST_BUFFER_DURATION (outbuf) = samples * GST_SECOND / dec->sample_rate;
|
|
samples = n;
|
|
|
|
cmeta = gst_buffer_get_audio_clipping_meta (buf);
|
|
|
|
g_assert (!cmeta || cmeta->format == GST_FORMAT_DEFAULT);
|
|
|
|
/* Skip any samples that need skipping */
|
|
if (cmeta && cmeta->start) {
|
|
guint pre_skip = cmeta->start;
|
|
guint scaled_pre_skip = pre_skip * dec->sample_rate / 48000;
|
|
guint skip = scaled_pre_skip > n ? n : scaled_pre_skip;
|
|
guint scaled_skip = skip * 48000 / dec->sample_rate;
|
|
|
|
gst_buffer_resize (outbuf, skip * 2 * dec->n_channels, -1);
|
|
|
|
GST_INFO_OBJECT (dec,
|
|
"Skipping %u samples at the beginning (%u at 48000 Hz)",
|
|
skip, scaled_skip);
|
|
}
|
|
|
|
if (cmeta && cmeta->end) {
|
|
guint post_skip = cmeta->end;
|
|
guint scaled_post_skip = post_skip * dec->sample_rate / 48000;
|
|
guint skip = scaled_post_skip > n ? n : scaled_post_skip;
|
|
guint scaled_skip = skip * 48000 / dec->sample_rate;
|
|
guint outsize = gst_buffer_get_size (outbuf);
|
|
guint skip_bytes = skip * 2 * dec->n_channels;
|
|
|
|
if (outsize > skip_bytes)
|
|
outsize -= skip_bytes;
|
|
else
|
|
outsize = 0;
|
|
|
|
gst_buffer_resize (outbuf, 0, outsize);
|
|
|
|
GST_INFO_OBJECT (dec,
|
|
"Skipping %u samples at the end (%u at 48000 Hz)", skip, scaled_skip);
|
|
}
|
|
|
|
if (gst_buffer_get_size (outbuf) == 0) {
|
|
gst_buffer_unref (outbuf);
|
|
outbuf = NULL;
|
|
} else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) {
|
|
gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16,
|
|
dec->n_channels, dec->opus_pos, dec->info.position);
|
|
}
|
|
|
|
/* Apply gain */
|
|
/* Would be better off leaving this to a volume element, as this is
|
|
a naive conversion that does too many int/float conversions.
|
|
However, we don't have control over the pipeline...
|
|
So make it optional if the user program wants to use a volume,
|
|
but do it by default so the correct volume goes out by default */
|
|
if (dec->apply_gain && outbuf && dec->r128_gain) {
|
|
gsize rsize;
|
|
unsigned int i, nsamples;
|
|
double volume = dec->r128_gain_volume;
|
|
gint16 *samples;
|
|
|
|
gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE);
|
|
samples = (gint16 *) omap.data;
|
|
rsize = omap.size;
|
|
GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
|
|
nsamples = rsize / 2;
|
|
for (i = 0; i < nsamples; ++i) {
|
|
int sample = (int) (samples[i] * volume + 0.5);
|
|
samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
|
|
}
|
|
gst_buffer_unmap (outbuf, &omap);
|
|
}
|
|
|
|
if (dec->use_inband_fec) {
|
|
gst_buffer_replace (&dec->last_buffer, buffer);
|
|
}
|
|
|
|
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
|
|
|
|
if (res != GST_FLOW_OK)
|
|
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
|
|
|
|
done:
|
|
return res;
|
|
|
|
creation_failed:
|
|
GST_ELEMENT_ERROR (dec, LIBRARY, INIT, ("Failed to create Opus decoder"),
|
|
("Failed to create Opus decoder (%d): %s", err, opus_strerror (err)));
|
|
return GST_FLOW_ERROR;
|
|
|
|
buffer_failed:
|
|
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
|
|
("Failed to create %u byte buffer", packet_size));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
|
|
static gboolean
|
|
gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
|
|
{
|
|
GstOpusDec *dec = GST_OPUS_DEC (bdec);
|
|
gboolean ret = TRUE;
|
|
GstStructure *s;
|
|
const GValue *streamheader;
|
|
GstCaps *old_caps;
|
|
|
|
GST_DEBUG_OBJECT (dec, "set_format: %" GST_PTR_FORMAT, caps);
|
|
|
|
if ((old_caps = gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (bdec)))) {
|
|
if (gst_caps_is_equal (caps, old_caps)) {
|
|
gst_caps_unref (old_caps);
|
|
GST_DEBUG_OBJECT (dec, "caps didn't change");
|
|
goto done;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (dec, "caps have changed, resetting decoder");
|
|
gst_opus_dec_reset (dec);
|
|
gst_caps_unref (old_caps);
|
|
}
|
|
|
|
s = gst_caps_get_structure (caps, 0);
|
|
if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
|
|
G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
|
|
gst_value_array_get_size (streamheader) >= 2) {
|
|
const GValue *header, *vorbiscomment;
|
|
GstBuffer *buf;
|
|
GstFlowReturn res = GST_FLOW_OK;
|
|
|
|
header = gst_value_array_get_value (streamheader, 0);
|
|
if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
|
|
buf = gst_value_get_buffer (header);
|
|
res = gst_opus_dec_parse_header (dec, buf);
|
|
if (res != GST_FLOW_OK) {
|
|
ret = FALSE;
|
|
goto done;
|
|
}
|
|
gst_buffer_replace (&dec->streamheader, buf);
|
|
}
|
|
|
|
vorbiscomment = gst_value_array_get_value (streamheader, 1);
|
|
if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
|
|
buf = gst_value_get_buffer (vorbiscomment);
|
|
res = gst_opus_dec_parse_comments (dec, buf);
|
|
if (res != GST_FLOW_OK) {
|
|
ret = FALSE;
|
|
goto done;
|
|
}
|
|
gst_buffer_replace (&dec->vorbiscomment, buf);
|
|
}
|
|
} else {
|
|
const GstAudioChannelPosition *posn = NULL;
|
|
|
|
if (!gst_codec_utils_opus_parse_caps (caps, &dec->sample_rate,
|
|
&dec->n_channels, &dec->channel_mapping_family, &dec->n_streams,
|
|
&dec->n_stereo_streams, dec->channel_mapping)) {
|
|
ret = FALSE;
|
|
goto done;
|
|
}
|
|
|
|
if (dec->channel_mapping_family == 1 && dec->n_channels <= 8)
|
|
posn = gst_opus_channel_positions[dec->n_channels - 1];
|
|
|
|
if (!gst_opus_dec_negotiate (dec, posn))
|
|
return FALSE;
|
|
}
|
|
|
|
done:
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2)
|
|
{
|
|
gsize size1, size2;
|
|
gboolean res;
|
|
GstMapInfo map;
|
|
|
|
size1 = gst_buffer_get_size (buf1);
|
|
size2 = gst_buffer_get_size (buf2);
|
|
|
|
if (size1 != size2)
|
|
return FALSE;
|
|
|
|
gst_buffer_map (buf1, &map, GST_MAP_READ);
|
|
res = gst_buffer_memcmp (buf2, 0, map.data, map.size) == 0;
|
|
gst_buffer_unmap (buf1, &map);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf)
|
|
{
|
|
GstFlowReturn res;
|
|
GstOpusDec *dec;
|
|
|
|
/* no fancy draining */
|
|
if (G_UNLIKELY (!buf))
|
|
return GST_FLOW_OK;
|
|
|
|
dec = GST_OPUS_DEC (adec);
|
|
GST_LOG_OBJECT (dec,
|
|
"Got buffer ts %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
|
|
|
|
/* If we have the streamheader and vorbiscomment from the caps already
|
|
* ignore them here */
|
|
if (dec->streamheader && dec->vorbiscomment) {
|
|
if (memcmp_buffers (dec->streamheader, buf)) {
|
|
GST_DEBUG_OBJECT (dec, "found streamheader");
|
|
gst_audio_decoder_finish_frame (adec, NULL, 1);
|
|
res = GST_FLOW_OK;
|
|
} else if (memcmp_buffers (dec->vorbiscomment, buf)) {
|
|
GST_DEBUG_OBJECT (dec, "found vorbiscomments");
|
|
gst_audio_decoder_finish_frame (adec, NULL, 1);
|
|
res = GST_FLOW_OK;
|
|
} else {
|
|
res = opus_dec_chain_parse_data (dec, buf);
|
|
}
|
|
} else {
|
|
/* Otherwise fall back to packet counting and assume that the
|
|
* first two packets might be the headers, checking magic. */
|
|
switch (dec->packetno) {
|
|
case 0:
|
|
if (gst_opus_header_is_header (buf, "OpusHead", 8)) {
|
|
GST_DEBUG_OBJECT (dec, "found streamheader");
|
|
res = gst_opus_dec_parse_header (dec, buf);
|
|
gst_audio_decoder_finish_frame (adec, NULL, 1);
|
|
} else {
|
|
res = opus_dec_chain_parse_data (dec, buf);
|
|
}
|
|
break;
|
|
case 1:
|
|
if (gst_opus_header_is_header (buf, "OpusTags", 8)) {
|
|
GST_DEBUG_OBJECT (dec, "counted vorbiscomments");
|
|
res = gst_opus_dec_parse_comments (dec, buf);
|
|
gst_audio_decoder_finish_frame (adec, NULL, 1);
|
|
} else {
|
|
res = opus_dec_chain_parse_data (dec, buf);
|
|
}
|
|
break;
|
|
default:
|
|
{
|
|
res = opus_dec_chain_parse_data (dec, buf);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
dec->packetno++;
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_opus_dec_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstOpusDec *dec = GST_OPUS_DEC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_USE_INBAND_FEC:
|
|
g_value_set_boolean (value, dec->use_inband_fec);
|
|
break;
|
|
case PROP_APPLY_GAIN:
|
|
g_value_set_boolean (value, dec->apply_gain);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_opus_dec_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstOpusDec *dec = GST_OPUS_DEC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_USE_INBAND_FEC:
|
|
dec->use_inband_fec = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_APPLY_GAIN:
|
|
dec->apply_gain = g_value_get_boolean (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* caps must be writable */
|
|
static void
|
|
gst_opus_dec_caps_extend_channels_options (GstCaps * caps)
|
|
{
|
|
unsigned n;
|
|
int channels;
|
|
|
|
for (n = 0; n < gst_caps_get_size (caps); ++n) {
|
|
GstStructure *s = gst_caps_get_structure (caps, n);
|
|
if (gst_structure_get_int (s, "channels", &channels)) {
|
|
if (channels == 1 || channels == 2) {
|
|
GValue v = { 0 };
|
|
g_value_init (&v, GST_TYPE_INT_RANGE);
|
|
gst_value_set_int_range (&v, 1, 2);
|
|
gst_structure_set_value (s, "channels", &v);
|
|
g_value_unset (&v);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_opus_dec_value_list_append_int (GValue * list, gint i)
|
|
{
|
|
GValue v = { 0 };
|
|
|
|
g_value_init (&v, G_TYPE_INT);
|
|
g_value_set_int (&v, i);
|
|
gst_value_list_append_value (list, &v);
|
|
g_value_unset (&v);
|
|
}
|
|
|
|
static void
|
|
gst_opus_dec_caps_extend_rate_options (GstCaps * caps)
|
|
{
|
|
unsigned n;
|
|
GValue v = { 0 };
|
|
|
|
g_value_init (&v, GST_TYPE_LIST);
|
|
gst_opus_dec_value_list_append_int (&v, 48000);
|
|
gst_opus_dec_value_list_append_int (&v, 24000);
|
|
gst_opus_dec_value_list_append_int (&v, 16000);
|
|
gst_opus_dec_value_list_append_int (&v, 12000);
|
|
gst_opus_dec_value_list_append_int (&v, 8000);
|
|
|
|
for (n = 0; n < gst_caps_get_size (caps); ++n) {
|
|
GstStructure *s = gst_caps_get_structure (caps, n);
|
|
|
|
gst_structure_set_value (s, "rate", &v);
|
|
}
|
|
g_value_unset (&v);
|
|
}
|
|
|
|
GstCaps *
|
|
gst_opus_dec_getcaps (GstAudioDecoder * dec, GstCaps * filter)
|
|
{
|
|
GstCaps *caps, *proxy_filter = NULL, *ret;
|
|
|
|
if (filter) {
|
|
proxy_filter = gst_caps_copy (filter);
|
|
gst_opus_dec_caps_extend_channels_options (proxy_filter);
|
|
gst_opus_dec_caps_extend_rate_options (proxy_filter);
|
|
}
|
|
caps = gst_audio_decoder_proxy_getcaps (dec, NULL, proxy_filter);
|
|
if (proxy_filter)
|
|
gst_caps_unref (proxy_filter);
|
|
if (caps) {
|
|
caps = gst_caps_make_writable (caps);
|
|
gst_opus_dec_caps_extend_channels_options (caps);
|
|
gst_opus_dec_caps_extend_rate_options (caps);
|
|
}
|
|
|
|
if (filter) {
|
|
ret = gst_caps_intersect (caps, filter);
|
|
gst_caps_unref (caps);
|
|
} else {
|
|
ret = caps;
|
|
}
|
|
return ret;
|
|
}
|