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445 lines
13 KiB
C
445 lines
13 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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*
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, 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 -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 "gstopusdec.h"
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#include <string.h>
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#include <gst/tag/tag.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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#define DEC_MAX_FRAME_SIZE 2000
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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-int, "
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"rate = (int) { 8000, 12000, 16000, 24000, 48000 }, "
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"channels = (int) [ 1, 2 ], "
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"endianness = (int) BYTE_ORDER, "
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"signed = (boolean) true, " "width = (int) 16, " "depth = (int) 16")
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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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);
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GST_BOILERPLATE (GstOpusDec, gst_opus_dec, GstAudioDecoder,
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GST_TYPE_AUDIO_DECODER);
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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
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gst_opus_dec_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&opus_dec_src_factory));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&opus_dec_sink_factory));
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gst_element_class_set_details_simple (element_class, "Opus audio decoder",
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"Codec/Decoder/Audio",
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"decode opus streams to audio",
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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}
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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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GstAudioDecoderClass *adclass;
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GstElementClass *gstelement_class;
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adclass = (GstAudioDecoderClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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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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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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dec->frame_size = 0;
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dec->frame_samples = 960;
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dec->frame_duration = 0;
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if (dec->state) {
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opus_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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}
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static void
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gst_opus_dec_init (GstOpusDec * dec, GstOpusDecClass * g_class)
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{
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dec->sample_rate = 48000;
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dec->n_channels = 2;
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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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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 GstFlowReturn
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gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
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{
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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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static GstFlowReturn
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opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
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GstClockTime timestamp, GstClockTime duration)
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{
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GstFlowReturn res = GST_FLOW_OK;
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gint size;
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guint8 *data;
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GstBuffer *outbuf;
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gint16 *out_data;
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int n, err;
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if (dec->state == NULL) {
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GstCaps *caps;
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dec->state = opus_decoder_create (dec->sample_rate, dec->n_channels, &err);
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if (!dec->state || err != OPUS_OK)
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goto creation_failed;
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/* set caps */
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caps = gst_caps_new_simple ("audio/x-raw-int",
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"rate", G_TYPE_INT, dec->sample_rate,
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"channels", G_TYPE_INT, dec->n_channels,
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"signed", G_TYPE_BOOLEAN, TRUE,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
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GST_DEBUG_OBJECT (dec, "rate=%d channels=%d frame-size=%d",
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dec->sample_rate, dec->n_channels, dec->frame_size);
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if (!gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps))
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GST_ERROR ("nego failure");
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gst_caps_unref (caps);
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}
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if (buf) {
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data = GST_BUFFER_DATA (buf);
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size = GST_BUFFER_SIZE (buf);
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GST_DEBUG_OBJECT (dec, "received buffer of size %u", size);
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/* copy timestamp */
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} else {
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/* concealment data, pass NULL as the bits parameters */
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GST_DEBUG_OBJECT (dec, "creating concealment data");
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data = NULL;
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size = 0;
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}
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GST_DEBUG ("frames %d", opus_packet_get_nb_frames (data, size));
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GST_DEBUG ("bandwidth %d", opus_packet_get_bandwidth (data));
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GST_DEBUG ("samples_per_frame %d", opus_packet_get_samples_per_frame (data,
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48000));
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GST_DEBUG ("channels %d", opus_packet_get_nb_channels (data));
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res = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec),
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GST_BUFFER_OFFSET_NONE, dec->frame_samples * dec->n_channels * 2,
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GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
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if (res != GST_FLOW_OK) {
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GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
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return res;
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}
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out_data = (gint16 *) GST_BUFFER_DATA (outbuf);
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GST_LOG_OBJECT (dec, "decoding %d sample frame", dec->frame_samples);
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n = opus_decode (dec->state, data, size, out_data, dec->frame_samples, 0);
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if (n < 0) {
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GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
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return GST_FLOW_ERROR;
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}
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if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
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GST_WARNING_OBJECT (dec, "No timestamp in -> no timestamp out");
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}
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GST_DEBUG_OBJECT (dec, "timestamp=%" GST_TIME_FORMAT,
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GST_TIME_ARGS (timestamp));
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GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
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GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (buf);
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GST_LOG_OBJECT (dec, "pushing buffer with ts=%" GST_TIME_FORMAT ", dur=%"
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GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
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GST_TIME_ARGS (dec->frame_duration));
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res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
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if (res != GST_FLOW_OK)
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GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
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return res;
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creation_failed:
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GST_ERROR_OBJECT (dec, "Failed to create Opus decoder: %d", err);
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return GST_FLOW_ERROR;
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}
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static gint
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gst_opus_dec_get_frame_samples (GstOpusDec * dec)
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{
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gint frame_samples = 0;
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switch (dec->frame_size) {
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case 2:
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frame_samples = dec->sample_rate / 400;
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break;
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case 5:
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frame_samples = dec->sample_rate / 200;
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break;
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case 10:
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frame_samples = dec->sample_rate / 100;
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break;
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case 20:
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frame_samples = dec->sample_rate / 50;
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break;
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case 40:
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frame_samples = dec->sample_rate / 25;
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break;
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case 60:
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frame_samples = 3 * dec->sample_rate / 50;
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break;
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default:
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GST_WARNING_OBJECT (dec, "Unsupported frame size: %d", dec->frame_size);
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frame_samples = 0;
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break;
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}
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return frame_samples;
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}
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static gboolean
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gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
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{
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GstOpusDec *dec = GST_OPUS_DEC (bdec);
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gboolean ret = TRUE;
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GstStructure *s;
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const GValue *streamheader;
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GST_DEBUG_OBJECT (dec, "set_format: %" GST_PTR_FORMAT, caps);
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s = gst_caps_get_structure (caps, 0);
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if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
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G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
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gst_value_array_get_size (streamheader) >= 2) {
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const GValue *header, *vorbiscomment;
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GstBuffer *buf;
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GstFlowReturn res = GST_FLOW_OK;
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header = gst_value_array_get_value (streamheader, 0);
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if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
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buf = gst_value_get_buffer (header);
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res = gst_opus_dec_parse_header (dec, buf);
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if (res != GST_FLOW_OK)
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goto done;
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gst_buffer_replace (&dec->streamheader, buf);
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}
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vorbiscomment = gst_value_array_get_value (streamheader, 1);
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if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
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buf = gst_value_get_buffer (vorbiscomment);
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res = gst_opus_dec_parse_comments (dec, buf);
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if (res != GST_FLOW_OK)
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goto done;
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gst_buffer_replace (&dec->vorbiscomment, buf);
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}
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}
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if (!gst_structure_get_int (s, "frame-size", &dec->frame_size)) {
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GST_WARNING_OBJECT (dec, "Frame size not included in caps");
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}
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if (!gst_structure_get_int (s, "channels", &dec->n_channels)) {
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GST_WARNING_OBJECT (dec, "Number of channels not included in caps");
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}
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if (!gst_structure_get_int (s, "rate", &dec->sample_rate)) {
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GST_WARNING_OBJECT (dec, "Sample rate not included in caps");
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}
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dec->frame_samples = gst_opus_dec_get_frame_samples (dec);
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dec->frame_duration = gst_util_uint64_scale_int (dec->frame_samples,
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GST_SECOND, dec->sample_rate);
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GST_INFO_OBJECT (dec,
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"Got frame size %d, %d channels, %d Hz, giving %d samples per frame, frame duration %"
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GST_TIME_FORMAT, dec->frame_size, dec->n_channels, dec->sample_rate,
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dec->frame_samples, GST_TIME_ARGS (dec->frame_duration));
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caps = gst_caps_new_simple ("audio/x-raw-int",
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"rate", G_TYPE_INT, dec->sample_rate,
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"channels", G_TYPE_INT, dec->n_channels,
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"signed", G_TYPE_BOOLEAN, TRUE,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
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gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
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gst_caps_unref (caps);
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done:
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return ret;
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}
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static gboolean
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memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2)
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{
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gsize size1, size2;
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size1 = GST_BUFFER_SIZE (buf1);
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size2 = GST_BUFFER_SIZE (buf2);
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if (size1 != size2)
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return FALSE;
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return !memcmp (GST_BUFFER_DATA (buf1), GST_BUFFER_DATA (buf2), size1);
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}
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static GstFlowReturn
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gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf)
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{
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GstFlowReturn res;
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GstOpusDec *dec;
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/* no fancy draining */
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if (G_UNLIKELY (!buf))
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return GST_FLOW_OK;
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dec = GST_OPUS_DEC (adec);
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GST_LOG_OBJECT (dec,
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"Got buffer ts %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT,
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
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GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
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/* If we have the streamheader and vorbiscomment from the caps already
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* ignore them here */
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if (dec->streamheader && dec->vorbiscomment) {
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if (memcmp_buffers (dec->streamheader, buf)) {
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GST_DEBUG_OBJECT (dec, "found streamheader");
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gst_audio_decoder_finish_frame (adec, NULL, 1);
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res = GST_FLOW_OK;
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} else if (memcmp_buffers (dec->vorbiscomment, buf)) {
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GST_DEBUG_OBJECT (dec, "found vorbiscomments");
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gst_audio_decoder_finish_frame (adec, NULL, 1);
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res = GST_FLOW_OK;
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} else {
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res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
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GST_BUFFER_DURATION (buf));
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}
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} else {
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/* Otherwise fall back to packet counting and assume that the
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* first two packets are the headers. */
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switch (dec->packetno) {
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case 0:
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GST_DEBUG_OBJECT (dec, "counted streamheader");
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res = gst_opus_dec_parse_header (dec, buf);
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gst_audio_decoder_finish_frame (adec, NULL, 1);
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break;
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case 1:
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GST_DEBUG_OBJECT (dec, "counted vorbiscomments");
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res = gst_opus_dec_parse_comments (dec, buf);
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gst_audio_decoder_finish_frame (adec, NULL, 1);
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break;
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default:
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{
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res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
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GST_BUFFER_DURATION (buf));
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break;
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}
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}
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}
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dec->packetno++;
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return res;
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}
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