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468 lines
15 KiB
C
468 lines
15 KiB
C
/*
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* Copyright (C) 2016 Sebastian Dröge <sebastian@centricular.com>
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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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstfdkaacdec.h"
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#include <gst/pbutils/pbutils.h>
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#include <string.h>
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/* TODO:
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* - LOAS / LATM support
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* - Error concealment
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*/
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#ifndef HAVE_FDK_AAC_0_1_4
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#define AAC_PCM_MAX_OUTPUT_CHANNELS AAC_PCM_OUTPUT_CHANNELS
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#define CHANNELS_CAPS_STR "channels = (int) [1, 6]"
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#else
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#define CHANNELS_CAPS_STR "channels = (int) [1, 8]"
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#endif
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static GstStaticPadTemplate sink_template = 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/mpeg, "
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"mpegversion = (int) {2, 4}, "
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"stream-format = (string) { adts, adif, raw }, " CHANNELS_CAPS_STR)
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);
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static GstStaticPadTemplate src_template = 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) [8000, 96000], " CHANNELS_CAPS_STR)
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);
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GST_DEBUG_CATEGORY_STATIC (gst_fdkaacdec_debug);
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#define GST_CAT_DEFAULT gst_fdkaacdec_debug
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static gboolean gst_fdkaacdec_start (GstAudioDecoder * dec);
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static gboolean gst_fdkaacdec_stop (GstAudioDecoder * dec);
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static gboolean gst_fdkaacdec_set_format (GstAudioDecoder * dec,
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GstCaps * caps);
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static GstFlowReturn gst_fdkaacdec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * in_buf);
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static void gst_fdkaacdec_flush (GstAudioDecoder * dec, gboolean hard);
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G_DEFINE_TYPE (GstFdkAacDec, gst_fdkaacdec, GST_TYPE_AUDIO_DECODER);
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static gboolean
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gst_fdkaacdec_start (GstAudioDecoder * dec)
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{
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GstFdkAacDec *self = GST_FDKAACDEC (dec);
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GST_DEBUG_OBJECT (self, "start");
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return TRUE;
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}
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static gboolean
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gst_fdkaacdec_stop (GstAudioDecoder * dec)
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{
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GstFdkAacDec *self = GST_FDKAACDEC (dec);
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GST_DEBUG_OBJECT (self, "stop");
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g_free (self->decode_buffer);
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self->decode_buffer = NULL;
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if (self->dec)
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aacDecoder_Close (self->dec);
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self->dec = NULL;
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return TRUE;
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}
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static gboolean
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gst_fdkaacdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
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{
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GstFdkAacDec *self = GST_FDKAACDEC (dec);
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TRANSPORT_TYPE transport_format;
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GstStructure *s;
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const gchar *stream_format;
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AAC_DECODER_ERROR err;
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if (self->dec) {
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/* drain */
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gst_fdkaacdec_handle_frame (dec, NULL);
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aacDecoder_Close (self->dec);
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self->dec = NULL;
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}
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s = gst_caps_get_structure (caps, 0);
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stream_format = gst_structure_get_string (s, "stream-format");
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if (strcmp (stream_format, "raw") == 0) {
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transport_format = TT_MP4_RAW;
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} else if (strcmp (stream_format, "adif") == 0) {
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transport_format = TT_MP4_ADIF;
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} else if (strcmp (stream_format, "adts") == 0) {
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transport_format = TT_MP4_ADTS;
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} else {
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g_assert_not_reached ();
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}
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self->dec = aacDecoder_Open (transport_format, 1);
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if (!self->dec) {
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GST_ERROR_OBJECT (self, "Failed to open decoder");
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return FALSE;
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}
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if (transport_format == TT_MP4_RAW) {
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GstBuffer *codec_data = NULL;
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GstMapInfo map;
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guint8 *data;
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guint size;
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gst_structure_get (s, "codec_data", GST_TYPE_BUFFER, &codec_data, NULL);
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if (!codec_data) {
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GST_ERROR_OBJECT (self, "Raw AAC without codec_data not supported");
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return FALSE;
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}
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gst_buffer_map (codec_data, &map, GST_MAP_READ);
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data = map.data;
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size = map.size;
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if ((err = aacDecoder_ConfigRaw (self->dec, &data, &size)) != AAC_DEC_OK) {
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gst_buffer_unmap (codec_data, &map);
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gst_buffer_unref (codec_data);
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GST_ERROR_OBJECT (self, "Invalid codec_data: %d", err);
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return FALSE;
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}
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gst_buffer_unmap (codec_data, &map);
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gst_buffer_unref (codec_data);
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}
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err = aacDecoder_SetParam (self->dec, AAC_PCM_MAX_OUTPUT_CHANNELS, 0);
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if (err != AAC_DEC_OK) {
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GST_ERROR_OBJECT (self, "Failed to disable downmixing: %d", err);
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return FALSE;
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}
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/* Choose WAV channel mapping to get interleaving even with libfdk-aac 2.0.0
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* The pChannelIndices retain the indices from the standard MPEG mapping so
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* we're agnostic to the actual order. */
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err = aacDecoder_SetParam (self->dec, AAC_PCM_OUTPUT_CHANNEL_MAPPING, 1);
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if (err != AAC_DEC_OK) {
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GST_ERROR_OBJECT (self, "Failed to set output channel mapping: %d", err);
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return FALSE;
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}
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/* 8 channels * 2 bytes per sample * 2048 samples */
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if (!self->decode_buffer) {
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self->decode_buffer_size = 8 * 2048;
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self->decode_buffer = g_new (gint16, self->decode_buffer_size);
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}
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return TRUE;
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}
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static GstFlowReturn
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gst_fdkaacdec_handle_frame (GstAudioDecoder * dec, GstBuffer * inbuf)
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{
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GstFdkAacDec *self = GST_FDKAACDEC (dec);
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GstFlowReturn ret = GST_FLOW_OK;
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GstBuffer *outbuf;
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GstMapInfo imap;
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AAC_DECODER_ERROR err;
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guint size, valid;
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CStreamInfo *stream_info;
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GstAudioInfo info;
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guint flags = 0, i;
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GstAudioChannelPosition pos[64], gst_pos[64];
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gboolean need_reorder;
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if (inbuf) {
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gst_buffer_ref (inbuf);
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gst_buffer_map (inbuf, &imap, GST_MAP_READ);
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valid = size = imap.size;
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err = aacDecoder_Fill (self->dec, (guint8 **) & imap.data, &size, &valid);
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if (err != AAC_DEC_OK) {
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GST_AUDIO_DECODER_ERROR (self, 1, STREAM, DECODE, (NULL),
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("filling error: %d", err), ret);
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goto out;
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}
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if (GST_BUFFER_IS_DISCONT (inbuf)) {
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flags |= AACDEC_INTR;
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}
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} else {
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flags |= AACDEC_FLUSH;
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}
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err = aacDecoder_DecodeFrame (self->dec, self->decode_buffer,
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self->decode_buffer_size, flags);
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if (err == AAC_DEC_TRANSPORT_SYNC_ERROR) {
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ret = GST_FLOW_OK;
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outbuf = NULL;
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goto finish;
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} else if (err != AAC_DEC_OK) {
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GST_AUDIO_DECODER_ERROR (self, 1, STREAM, DECODE, (NULL),
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("decoding error: %d", err), ret);
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goto out;
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}
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stream_info = aacDecoder_GetStreamInfo (self->dec);
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if (!stream_info) {
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GST_AUDIO_DECODER_ERROR (self, 1, STREAM, DECODE, (NULL),
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("failed to get stream info"), ret);
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goto out;
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}
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/* FIXME: Don't recalculate this on every buffer */
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if (stream_info->numChannels == 1) {
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pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
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} else {
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gint n_front = 0, n_back = 0, n_lfe = 0;
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/* FIXME: Can this be simplified somehow? */
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for (i = 0; i < stream_info->numChannels; i++) {
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if (stream_info->pChannelType[i] == ACT_FRONT) {
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n_front++;
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} else if (stream_info->pChannelType[i] == ACT_BACK) {
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n_back++;
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} else if (stream_info->pChannelType[i] == ACT_LFE) {
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n_lfe++;
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} else {
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GST_ERROR_OBJECT (self, "Channel type %d not supported",
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stream_info->pChannelType[i]);
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ret = GST_FLOW_NOT_NEGOTIATED;
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goto out;
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}
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}
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for (i = 0; i < stream_info->numChannels; i++) {
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if (stream_info->pChannelType[i] == ACT_FRONT) {
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if (stream_info->pChannelIndices[i] == 0) {
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if (n_front & 1)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
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else if (n_front > 2)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
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else
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pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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} else if (stream_info->pChannelIndices[i] == 1) {
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if ((n_front & 1) && n_front > 3)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
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else if (n_front & 1)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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else if (n_front > 2)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
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else
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pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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} else if (stream_info->pChannelIndices[i] == 2) {
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if ((n_front & 1) && n_front > 3)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
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else if (n_front & 1)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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else if (n_front > 2)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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else
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g_assert_not_reached ();
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} else if (stream_info->pChannelIndices[i] == 3) {
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if ((n_front & 1) && n_front > 3)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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else if (n_front & 1)
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g_assert_not_reached ();
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else if (n_front > 2)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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else
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g_assert_not_reached ();
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} else if (stream_info->pChannelIndices[i] == 4) {
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if ((n_front & 1) && n_front > 2)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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else if (n_front & 1)
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g_assert_not_reached ();
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else if (n_front > 2)
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g_assert_not_reached ();
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else
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g_assert_not_reached ();
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} else {
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GST_ERROR_OBJECT (self, "Front channel index %d not supported",
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stream_info->pChannelIndices[i]);
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ret = GST_FLOW_NOT_NEGOTIATED;
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goto out;
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}
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} else if (stream_info->pChannelType[i] == ACT_BACK) {
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if (stream_info->pChannelIndices[i] == 0) {
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if (n_back & 1)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
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else if (n_back > 2)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
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else
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pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
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} else if (stream_info->pChannelIndices[i] == 1) {
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if ((n_back & 1) && n_back > 3)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
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else if (n_back & 1)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
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else if (n_back > 2)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
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else
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pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
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} else if (stream_info->pChannelIndices[i] == 2) {
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if ((n_back & 1) && n_back > 3)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
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else if (n_back & 1)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
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else if (n_back > 2)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
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else
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g_assert_not_reached ();
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} else if (stream_info->pChannelIndices[i] == 3) {
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if ((n_back & 1) && n_back > 3)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
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else if (n_back & 1)
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g_assert_not_reached ();
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else if (n_back > 2)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
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else
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g_assert_not_reached ();
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} else if (stream_info->pChannelIndices[i] == 4) {
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if ((n_back & 1) && n_back > 3)
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pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
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else if (n_back & 1)
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g_assert_not_reached ();
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else if (n_back > 2)
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g_assert_not_reached ();
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else
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g_assert_not_reached ();
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} else {
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GST_ERROR_OBJECT (self, "Side channel index %d not supported",
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stream_info->pChannelIndices[i]);
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ret = GST_FLOW_NOT_NEGOTIATED;
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goto out;
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}
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} else if (stream_info->pChannelType[i] == ACT_LFE) {
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if (stream_info->pChannelIndices[i] == 0) {
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pos[i] = GST_AUDIO_CHANNEL_POSITION_LFE1;
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} else {
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GST_ERROR_OBJECT (self, "LFE channel index %d not supported",
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stream_info->pChannelIndices[i]);
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ret = GST_FLOW_NOT_NEGOTIATED;
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goto out;
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}
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} else {
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GST_ERROR_OBJECT (self, "Channel type %d not supported",
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stream_info->pChannelType[i]);
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ret = GST_FLOW_NOT_NEGOTIATED;
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goto out;
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}
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}
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}
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memcpy (gst_pos, pos,
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sizeof (GstAudioChannelPosition) * stream_info->numChannels);
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if (!gst_audio_channel_positions_to_valid_order (gst_pos,
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stream_info->numChannels)) {
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ret = GST_FLOW_NOT_NEGOTIATED;
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goto out;
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}
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need_reorder =
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memcmp (pos, gst_pos,
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sizeof (GstAudioChannelPosition) * stream_info->numChannels) != 0;
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gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16,
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stream_info->sampleRate, stream_info->numChannels, gst_pos);
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if (!gst_audio_decoder_set_output_format (dec, &info)) {
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GST_ERROR_OBJECT (self, "Failed to set output format");
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ret = GST_FLOW_NOT_NEGOTIATED;
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goto out;
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}
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outbuf =
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gst_audio_decoder_allocate_output_buffer (dec,
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stream_info->frameSize * GST_AUDIO_INFO_BPF (&info));
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gst_buffer_fill (outbuf, 0, self->decode_buffer,
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gst_buffer_get_size (outbuf));
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if (need_reorder) {
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gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_INFO_FORMAT (&info),
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GST_AUDIO_INFO_CHANNELS (&info), pos, gst_pos);
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}
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finish:
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ret = gst_audio_decoder_finish_frame (dec, outbuf, 1);
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out:
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if (inbuf) {
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gst_buffer_unmap (inbuf, &imap);
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gst_buffer_unref (inbuf);
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}
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return ret;
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}
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static void
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gst_fdkaacdec_flush (GstAudioDecoder * dec, gboolean hard)
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{
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GstFdkAacDec *self = GST_FDKAACDEC (dec);
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if (self->dec) {
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AAC_DECODER_ERROR err;
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err = aacDecoder_DecodeFrame (self->dec, self->decode_buffer,
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self->decode_buffer_size, AACDEC_FLUSH);
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if (err != AAC_DEC_OK) {
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GST_ERROR_OBJECT (self, "flushing error: %d", err);
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}
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}
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}
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static void
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gst_fdkaacdec_init (GstFdkAacDec * self)
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{
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self->dec = NULL;
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gst_audio_decoder_set_drainable (GST_AUDIO_DECODER (self), TRUE);
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gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (self), TRUE);
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}
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static void
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gst_fdkaacdec_class_init (GstFdkAacDecClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
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base_class->start = GST_DEBUG_FUNCPTR (gst_fdkaacdec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_fdkaacdec_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_fdkaacdec_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_fdkaacdec_handle_frame);
|
|
base_class->flush = GST_DEBUG_FUNCPTR (gst_fdkaacdec_flush);
|
|
|
|
gst_element_class_add_static_pad_template (element_class, &sink_template);
|
|
gst_element_class_add_static_pad_template (element_class, &src_template);
|
|
|
|
gst_element_class_set_static_metadata (element_class, "FDK AAC audio decoder",
|
|
"Codec/Decoder/Audio", "FDK AAC audio decoder",
|
|
"Sebastian Dröge <sebastian@centricular.com>");
|
|
|
|
GST_DEBUG_CATEGORY_INIT (gst_fdkaacdec_debug, "fdkaacdec", 0,
|
|
"fdkaac decoder");
|
|
}
|