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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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a63d8ee720
A flush request is done when set_format is called to empty internal bit buffer maintained by fdk-aac. When this happens, during the explicit call to handle_buffer, decodeFrame does not return a AAC_DEC_OK. This gets reported as a decoding error while no decoding error in fact took place. Since this can be confusing, just return a GST_FLOW_OK and log that an explicit flush was requested.
697 lines
19 KiB
C
697 lines
19 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 "gstfdkaac.h"
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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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GST_ELEMENT_REGISTER_DEFINE (fdkaacdec, "fdkaacdec", GST_RANK_MARGINAL,
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GST_TYPE_FDKAACDEC);
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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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gst_audio_info_init (&self->info);
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self->sample_rate = 0;
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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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/* 64 channels * 2048 samples * 2 bytes per sample */
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if (!self->decode_buffer) {
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self->decode_buffer_size = 64 * 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 gboolean
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gst_fdkaacdec_map_channels (GstFdkAacDec * self, const CStreamInfo * in,
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gboolean * updated)
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{
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GstAudioChannelPosition *positions = self->positions;
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AUDIO_CHANNEL_TYPE *channel_types = in->pChannelType;
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UCHAR *channel_indices = in->pChannelIndices;
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INT i, channels = in->numChannels;
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guint64 mask_mapped = 0;
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#define DEF_CHANSET(name, max) \
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GstAudioChannelPosition *set_ ## name[max] = {NULL}; \
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guint n_ ## name = 0, mapped_ ## name = 0
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#define PUSH_CHAN(name, index, pos_index) G_STMT_START { \
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if ((index) >= G_N_ELEMENTS (set_ ## name)) { \
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GST_WARNING_OBJECT (self, "Too many %s channels (%d)", \
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#name, (gint) (index)); \
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goto error; \
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} else if (set_ ## name[index] != NULL) { \
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GST_WARNING_OBJECT (self, "Channel %s[%d] already mapped", \
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#name, (gint) (index)); \
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goto error; \
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} else { \
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GST_DEBUG_OBJECT (self, "Mapping channel %s[%d] to %d", \
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#name, (gint) (index), (gint) pos_index); \
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set_ ## name[index] = &positions[pos_index]; \
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n_ ## name = MAX (n_ ## name, (index) + 1); \
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} \
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} G_STMT_END
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#define SHIFT_CHAN(name, pos) G_STMT_START { \
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if (mask_mapped & GST_AUDIO_CHANNEL_POSITION_MASK (pos)) { \
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GST_WARNING_OBJECT (self, "Position %s already mapped", #pos); \
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goto error; \
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} else if (set_ ## name[mapped_ ## name] == NULL) { \
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GST_WARNING_OBJECT (self, "Channel %s[%u] is a hole", \
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#name, mapped_ ## name); \
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goto error; \
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} else { \
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GST_DEBUG_OBJECT (self, "Mapping channel %s[%u] to %s", \
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#name, mapped_ ## name, #pos); \
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*set_ ## name[mapped_ ## name ++] = GST_AUDIO_CHANNEL_POSITION_ ## pos; \
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mask_mapped |= GST_AUDIO_CHANNEL_POSITION_MASK (pos); \
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} \
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} G_STMT_END
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DEF_CHANSET (front, 7);
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DEF_CHANSET (side, 2);
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DEF_CHANSET (rear, 3);
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DEF_CHANSET (lfe, 2);
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DEF_CHANSET (top_front, 3);
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DEF_CHANSET (top_center, 3);
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DEF_CHANSET (top_rear, 3);
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DEF_CHANSET (bottom_front, 3);
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if (self->channels == channels &&
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memcmp (self->channel_types, channel_types,
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channels * sizeof *channel_types) == 0 &&
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memcmp (self->channel_indices, channel_indices,
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channels * sizeof *channel_indices) == 0) {
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GST_TRACE_OBJECT (self, "Reusing cached positions for %d channels",
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channels);
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return TRUE;
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}
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self->channels = channels;
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memcpy (self->channel_types, channel_types, channels * sizeof *channel_types);
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memcpy (self->channel_indices, channel_indices,
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channels * sizeof *channel_indices);
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*updated = TRUE;
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for (i = 0; i < channels; i++) {
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guint8 type = in->pChannelType[i];
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guint8 index = in->pChannelIndices[i];
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switch (type) {
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case ACT_FRONT:
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PUSH_CHAN (front, index, i);
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break;
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case ACT_SIDE:
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PUSH_CHAN (side, index, i);
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break;
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case ACT_BACK:
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PUSH_CHAN (rear, index, i);
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break;
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case ACT_LFE:
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PUSH_CHAN (lfe, index, i);
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break;
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case ACT_FRONT_TOP:
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PUSH_CHAN (top_front, index, i);
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break;
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case ACT_SIDE_TOP:
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PUSH_CHAN (top_center, index, i);
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break;
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case ACT_BACK_TOP:
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PUSH_CHAN (top_rear, index, i);
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break;
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#ifdef HAVE_FDK_AAC_0_1_4
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case ACT_FRONT_BOTTOM:
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PUSH_CHAN (bottom_front, index, i);
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break;
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#endif
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case ACT_NONE:
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GST_INFO_OBJECT (self, "Channel %d is unpositioned", i);
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goto error;
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default:
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GST_ERROR_OBJECT (self, "Channel %d has unknown type %d", i, type);
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goto error;
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}
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}
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/* Outwards from the front center, following ISO/IEC 13818-7 8.5.2.2
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* "Explicit channel mapping using a program_config_element()" */
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switch (n_front) {
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case 7:
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SHIFT_CHAN (front, FRONT_CENTER);
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case 6:
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SHIFT_CHAN (front, FRONT_LEFT_OF_CENTER);
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SHIFT_CHAN (front, FRONT_RIGHT_OF_CENTER);
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SHIFT_CHAN (front, FRONT_LEFT);
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SHIFT_CHAN (front, FRONT_RIGHT);
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SHIFT_CHAN (front, WIDE_LEFT);
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SHIFT_CHAN (front, WIDE_RIGHT);
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break;
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case 5:
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SHIFT_CHAN (front, FRONT_CENTER);
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case 4:
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SHIFT_CHAN (front, FRONT_LEFT_OF_CENTER);
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SHIFT_CHAN (front, FRONT_RIGHT_OF_CENTER);
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SHIFT_CHAN (front, WIDE_LEFT);
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SHIFT_CHAN (front, WIDE_RIGHT);
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break;
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case 3:
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SHIFT_CHAN (front, FRONT_CENTER);
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case 2:
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SHIFT_CHAN (front, FRONT_LEFT);
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SHIFT_CHAN (front, FRONT_RIGHT);
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break;
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case 1:
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SHIFT_CHAN (front, FRONT_CENTER);
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break;
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}
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/* Front to rear */
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switch (n_side) {
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case 2:
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SHIFT_CHAN (side, SIDE_LEFT);
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SHIFT_CHAN (side, SIDE_RIGHT);
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break;
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case 1:
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GST_ERROR_OBJECT (self, "Single side channel not supported");
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goto error;
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}
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/* Inwards to the rear center */
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switch (n_rear) {
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case 5:
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SHIFT_CHAN (rear, SURROUND_LEFT);
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SHIFT_CHAN (rear, SURROUND_RIGHT);
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SHIFT_CHAN (rear, REAR_LEFT);
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SHIFT_CHAN (rear, REAR_RIGHT);
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SHIFT_CHAN (rear, REAR_CENTER);
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break;
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case 4:
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SHIFT_CHAN (rear, SURROUND_LEFT);
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SHIFT_CHAN (rear, SURROUND_RIGHT);
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SHIFT_CHAN (rear, REAR_LEFT);
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SHIFT_CHAN (rear, REAR_RIGHT);
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break;
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case 3:
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SHIFT_CHAN (rear, SURROUND_LEFT);
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SHIFT_CHAN (rear, SURROUND_RIGHT);
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SHIFT_CHAN (rear, REAR_CENTER);
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break;
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case 2:
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SHIFT_CHAN (rear, SURROUND_LEFT);
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SHIFT_CHAN (rear, SURROUND_RIGHT);
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break;
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case 1:
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SHIFT_CHAN (rear, REAR_CENTER);
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break;
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}
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switch (n_lfe) {
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case 2:
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SHIFT_CHAN (lfe, LFE1);
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SHIFT_CHAN (lfe, LFE2);
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break;
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case 1:
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SHIFT_CHAN (lfe, LFE1);
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break;
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}
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switch (n_top_front) {
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case 3:
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SHIFT_CHAN (top_front, TOP_FRONT_CENTER);
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case 2:
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SHIFT_CHAN (top_front, TOP_FRONT_LEFT);
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SHIFT_CHAN (top_front, TOP_FRONT_RIGHT);
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break;
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case 1:
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SHIFT_CHAN (top_front, TOP_FRONT_CENTER);
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break;
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}
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switch (n_top_center) {
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case 3:
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SHIFT_CHAN (top_center, TOP_CENTER);
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case 2:
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SHIFT_CHAN (top_center, TOP_SIDE_LEFT);
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SHIFT_CHAN (top_center, TOP_SIDE_RIGHT);
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break;
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case 1:
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SHIFT_CHAN (top_center, TOP_CENTER);
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break;
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}
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switch (n_top_rear) {
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case 3:
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SHIFT_CHAN (top_rear, TOP_REAR_LEFT);
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SHIFT_CHAN (top_rear, TOP_REAR_RIGHT);
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SHIFT_CHAN (top_rear, TOP_REAR_CENTER);
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break;
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case 2:
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SHIFT_CHAN (top_rear, TOP_REAR_LEFT);
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SHIFT_CHAN (top_rear, TOP_REAR_RIGHT);
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break;
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case 1:
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SHIFT_CHAN (top_rear, TOP_REAR_CENTER);
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break;
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}
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switch (n_bottom_front) {
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case 3:
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SHIFT_CHAN (bottom_front, BOTTOM_FRONT_CENTER);
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case 2:
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SHIFT_CHAN (bottom_front, BOTTOM_FRONT_LEFT);
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SHIFT_CHAN (bottom_front, BOTTOM_FRONT_RIGHT);
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break;
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case 1:
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SHIFT_CHAN (bottom_front, BOTTOM_FRONT_CENTER);
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break;
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}
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if (mask_mapped != 0) {
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GST_INFO_OBJECT (self, "Mapped %d front, %d side, %d rear, %d lfe,"
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" %d top front, %d top center, %d top rear, %d bottom front channels",
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mapped_front, mapped_side, mapped_rear, mapped_lfe, mapped_top_front,
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mapped_top_center, mapped_top_rear, mapped_bottom_front);
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return TRUE;
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}
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if (channels == 1) {
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GST_INFO_OBJECT (self, "Mapped a mono channel");
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positions[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
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return TRUE;
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}
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error:
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if (channels > 0) {
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GST_WARNING_OBJECT (self, "Mapped %d channels, without positions",
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channels);
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for (i = 0; i < channels; i++)
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positions[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
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return TRUE;
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}
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GST_ERROR_OBJECT (self, "No channels to map");
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return FALSE;
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#undef DEF_CHANSET
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#undef PUSH_CHAN
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#undef SHIFT_CHAN
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}
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static gboolean
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gst_fdkaacdec_map_channel_config (GstFdkAacDec * self, const CStreamInfo * in,
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gboolean * updated)
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{
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const GstFdkAacChannelLayout *layout;
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CHANNEL_MODE config = in->channelConfig;
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INT channels = in->numChannels;
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if (config == 0) {
|
|
return gst_fdkaacdec_map_channels (self, in, updated);
|
|
}
|
|
|
|
if (self->config == config && self->channels == channels) {
|
|
GST_TRACE_OBJECT (self,
|
|
"Reusing cached positions for channelConfig %d (%d channels)",
|
|
config, channels);
|
|
return TRUE;
|
|
}
|
|
|
|
self->config = config;
|
|
self->channels = channels;
|
|
*updated = TRUE;
|
|
|
|
for (layout = channel_layouts; layout->channels; layout++) {
|
|
if (layout->mode == config && layout->channels == channels)
|
|
break;
|
|
}
|
|
|
|
if (!layout->channels) {
|
|
GST_WARNING_OBJECT (self, "Unknown channelConfig %d (%d channels)",
|
|
config, channels);
|
|
return gst_fdkaacdec_map_channels (self, in, updated);
|
|
}
|
|
|
|
GST_INFO_OBJECT (self, "Known channelConfig %d (%d channels)",
|
|
config, channels);
|
|
memcpy (self->positions, layout->positions,
|
|
channels * sizeof *self->positions);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_fdkaacdec_update_info (GstFdkAacDec * self)
|
|
{
|
|
GstAudioChannelPosition positions[64];
|
|
GstAudioInfo *info = &self->info;
|
|
gint channels = self->channels;
|
|
|
|
memcpy (positions, self->positions, channels * sizeof *positions);
|
|
|
|
if (!gst_audio_channel_positions_to_valid_order (positions, channels)) {
|
|
GST_ERROR_OBJECT (self, "Failed to reorder channels");
|
|
return FALSE;
|
|
}
|
|
|
|
gst_audio_info_set_format (info, GST_AUDIO_FORMAT_S16, self->sample_rate,
|
|
channels, positions);
|
|
|
|
if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (self), info)) {
|
|
GST_ERROR_OBJECT (self, "Failed to set output format");
|
|
return FALSE;
|
|
}
|
|
|
|
self->need_reorder = memcmp (positions, self->positions,
|
|
channels * sizeof *positions) != 0;
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_fdkaacdec_handle_frame (GstAudioDecoder * dec, GstBuffer * inbuf)
|
|
{
|
|
GstFdkAacDec *self = GST_FDKAACDEC (dec);
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstBuffer *outbuf;
|
|
GstMapInfo imap;
|
|
AAC_DECODER_ERROR err;
|
|
UINT flags = 0;
|
|
guint size, valid;
|
|
CStreamInfo *stream_info;
|
|
gboolean updated = FALSE;
|
|
|
|
if (inbuf) {
|
|
gst_buffer_ref (inbuf);
|
|
gst_buffer_map (inbuf, &imap, GST_MAP_READ);
|
|
valid = size = imap.size;
|
|
|
|
err = aacDecoder_Fill (self->dec, (guint8 **) & imap.data, &size, &valid);
|
|
if (err != AAC_DEC_OK) {
|
|
GST_AUDIO_DECODER_ERROR (self, 1, STREAM, DECODE, (NULL),
|
|
("filling error: %d", err), ret);
|
|
goto out;
|
|
}
|
|
|
|
if (GST_BUFFER_IS_DISCONT (inbuf)) {
|
|
flags |= AACDEC_INTR;
|
|
}
|
|
} else {
|
|
flags |= AACDEC_FLUSH;
|
|
}
|
|
|
|
err = aacDecoder_DecodeFrame (self->dec, self->decode_buffer,
|
|
self->decode_buffer_size, flags);
|
|
if (err == AAC_DEC_TRANSPORT_SYNC_ERROR) {
|
|
ret = GST_FLOW_OK;
|
|
outbuf = NULL;
|
|
goto finish;
|
|
} else if ((err != AAC_DEC_OK) && (flags & AACDEC_FLUSH)) {
|
|
/*
|
|
* A flush/drain was requested when set_format got called. When a flush
|
|
* gets requested, aacDecoder_DecodeFrame may not return AAC_DEC_OK. Do
|
|
* not report a decoding error with GST_AUDIO_DECODER_ERROR for this case.
|
|
*/
|
|
GST_LOG_OBJECT (self, "Decoder flush was requested");
|
|
ret = GST_FLOW_OK;
|
|
goto out;
|
|
} else if (err != AAC_DEC_OK) {
|
|
GST_AUDIO_DECODER_ERROR (self, 1, STREAM, DECODE, (NULL),
|
|
("decoding error: %d", err), ret);
|
|
goto out;
|
|
}
|
|
|
|
stream_info = aacDecoder_GetStreamInfo (self->dec);
|
|
if (!stream_info) {
|
|
GST_AUDIO_DECODER_ERROR (self, 1, STREAM, DECODE, (NULL),
|
|
("failed to get stream info"), ret);
|
|
goto out;
|
|
}
|
|
|
|
if (stream_info->sampleRate != self->sample_rate) {
|
|
self->sample_rate = stream_info->sampleRate;
|
|
updated = TRUE;
|
|
}
|
|
|
|
if (!gst_fdkaacdec_map_channel_config (self, stream_info, &updated)) {
|
|
ret = GST_FLOW_NOT_NEGOTIATED;
|
|
goto out;
|
|
}
|
|
|
|
if (updated && !gst_fdkaacdec_update_info (self)) {
|
|
ret = GST_FLOW_NOT_NEGOTIATED;
|
|
goto out;
|
|
}
|
|
|
|
outbuf =
|
|
gst_audio_decoder_allocate_output_buffer (dec,
|
|
stream_info->frameSize * GST_AUDIO_INFO_BPF (&self->info));
|
|
|
|
gst_buffer_fill (outbuf, 0, self->decode_buffer,
|
|
gst_buffer_get_size (outbuf));
|
|
|
|
if (self->need_reorder) {
|
|
gst_audio_buffer_reorder_channels (outbuf,
|
|
GST_AUDIO_INFO_FORMAT (&self->info),
|
|
GST_AUDIO_INFO_CHANNELS (&self->info),
|
|
self->positions, self->info.position);
|
|
}
|
|
|
|
finish:
|
|
ret = gst_audio_decoder_finish_frame (dec, outbuf, 1);
|
|
|
|
out:
|
|
|
|
if (inbuf) {
|
|
gst_buffer_unmap (inbuf, &imap);
|
|
gst_buffer_unref (inbuf);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_fdkaacdec_flush (GstAudioDecoder * dec, gboolean hard)
|
|
{
|
|
GstFdkAacDec *self = GST_FDKAACDEC (dec);
|
|
|
|
if (self->dec) {
|
|
AAC_DECODER_ERROR err;
|
|
err = aacDecoder_DecodeFrame (self->dec, self->decode_buffer,
|
|
self->decode_buffer_size, AACDEC_FLUSH);
|
|
if (err != AAC_DEC_OK) {
|
|
GST_ERROR_OBJECT (self, "flushing error: %d", err);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_fdkaacdec_init (GstFdkAacDec * self)
|
|
{
|
|
self->dec = NULL;
|
|
|
|
gst_audio_decoder_set_drainable (GST_AUDIO_DECODER (self), TRUE);
|
|
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (self), TRUE);
|
|
}
|
|
|
|
static void
|
|
gst_fdkaacdec_class_init (GstFdkAacDecClass * klass)
|
|
{
|
|
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
|
|
GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
|
|
|
|
base_class->start = GST_DEBUG_FUNCPTR (gst_fdkaacdec_start);
|
|
base_class->stop = GST_DEBUG_FUNCPTR (gst_fdkaacdec_stop);
|
|
base_class->set_format = GST_DEBUG_FUNCPTR (gst_fdkaacdec_set_format);
|
|
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");
|
|
}
|