--- /dev/null
+/*
+ * Copyright (C) 2016 Sebastian Dröge <sebastian@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstfdkaacdec.h"
+
+#include <gst/pbutils/pbutils.h>
+
+#include <string.h>
+
+/* TODO:
+ * - LOAS / LATM support
+ * - Error concealment
+ */
+
+static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/mpeg, "
+ "mpegversion = (int) 4, "
+ "stream-format = (string) { adts, adif, raw }")
+ );
+
+static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " GST_AUDIO_NE (S16) ", "
+ "layout = (string) interleaved, "
+ "rate = (int) [8000, 96000], " "channels = (int) [1, 8]")
+ );
+
+GST_DEBUG_CATEGORY_STATIC (gst_fdkaacdec_debug);
+#define GST_CAT_DEFAULT gst_fdkaacdec_debug
+
+static gboolean gst_fdkaacdec_start (GstAudioDecoder * dec);
+static gboolean gst_fdkaacdec_stop (GstAudioDecoder * dec);
+static gboolean gst_fdkaacdec_set_format (GstAudioDecoder * dec,
+ GstCaps * caps);
+static GstFlowReturn gst_fdkaacdec_handle_frame (GstAudioDecoder * dec,
+ GstBuffer * in_buf);
+static void gst_fdkaacdec_flush (GstAudioDecoder * dec, gboolean hard);
+
+G_DEFINE_TYPE (GstFdkAacDec, gst_fdkaacdec, GST_TYPE_AUDIO_DECODER);
+
+static gboolean
+gst_fdkaacdec_start (GstAudioDecoder * dec)
+{
+ GstFdkAacDec *self = GST_FDKAACDEC (dec);
+
+ GST_DEBUG_OBJECT (self, "start");
+
+ return TRUE;
+}
+
+static gboolean
+gst_fdkaacdec_stop (GstAudioDecoder * dec)
+{
+ GstFdkAacDec *self = GST_FDKAACDEC (dec);
+
+ GST_DEBUG_OBJECT (self, "stop");
+
+ g_free (self->decode_buffer);
+ self->decode_buffer = NULL;
+
+ if (self->dec)
+ aacDecoder_Close (self->dec);
+ self->dec = NULL;
+
+ return TRUE;
+}
+
+static gboolean
+gst_fdkaacdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
+{
+ GstFdkAacDec *self = GST_FDKAACDEC (dec);
+ TRANSPORT_TYPE transport_format;
+ GstStructure *s;
+ const gchar *stream_format;
+ AAC_DECODER_ERROR err;
+
+ if (self->dec) {
+ /* drain */
+ gst_fdkaacdec_handle_frame (dec, NULL);
+ aacDecoder_Close (self->dec);
+ self->dec = NULL;
+ }
+
+ s = gst_caps_get_structure (caps, 0);
+ stream_format = gst_structure_get_string (s, "stream-format");
+ if (strcmp (stream_format, "raw") == 0) {
+ transport_format = TT_MP4_RAW;
+ } else if (strcmp (stream_format, "adif") == 0) {
+ transport_format = TT_MP4_ADIF;
+ } else if (strcmp (stream_format, "adts") == 0) {
+ transport_format = TT_MP4_ADTS;
+ } else {
+ g_assert_not_reached ();
+ }
+
+ self->dec = aacDecoder_Open (transport_format, 1);
+ if (!self->dec) {
+ GST_ERROR_OBJECT (self, "Failed to open decoder");
+ return FALSE;
+ }
+
+ if (transport_format == TT_MP4_RAW) {
+ GstBuffer *codec_data = NULL;
+ GstMapInfo map;
+ guint8 *data;
+ guint size;
+
+ gst_structure_get (s, "codec_data", GST_TYPE_BUFFER, &codec_data, NULL);
+
+ if (!codec_data) {
+ GST_ERROR_OBJECT (self, "Raw AAC without codec_data not supported");
+ return FALSE;
+ }
+
+ gst_buffer_map (codec_data, &map, GST_MAP_READ);
+ data = map.data;
+ size = map.size;
+
+ if ((err = aacDecoder_ConfigRaw (self->dec, &data, &size)) != AAC_DEC_OK) {
+ gst_buffer_unmap (codec_data, &map);
+ GST_ERROR_OBJECT (self, "Invalid codec_data: %d", err);
+ return FALSE;
+ }
+
+ gst_buffer_unmap (codec_data, &map);
+ }
+
+ if ((err =
+ aacDecoder_SetParam (self->dec, AAC_PCM_OUTPUT_CHANNEL_MAPPING,
+ 0)) != AAC_DEC_OK) {
+ GST_ERROR_OBJECT (self, "Failed to set output channel mapping: %d", err);
+ return FALSE;
+ }
+
+ if ((err =
+ aacDecoder_SetParam (self->dec, AAC_PCM_OUTPUT_INTERLEAVED,
+ 0)) != AAC_DEC_OK) {
+ GST_ERROR_OBJECT (self, "Failed to set output channel mapping: %d", err);
+ return FALSE;
+ }
+
+ /* 8 channels * 2 bytes per sample * 2048 samples */
+ if (!self->decode_buffer) {
+ self->decode_buffer_size = 8 * 2 * 2048;
+ self->decode_buffer = g_malloc (self->decode_buffer_size);
+ }
+
+ 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;
+ guint size, valid;
+ CStreamInfo *stream_info;
+ GstAudioInfo info;
+ guint flags = 0, i;
+ GstAudioChannelPosition pos[64], gst_pos[64];
+ gboolean need_reorder;
+
+ if (inbuf) {
+ gst_buffer_map (inbuf, &imap, GST_MAP_READ);
+ valid = size = imap.size;
+
+ if ((err =
+ aacDecoder_Fill (self->dec, (guint8 **) & imap.data, &size,
+ &valid)) != AAC_DEC_OK) {
+ GST_AUDIO_DECODER_ERROR (self, 1, STREAM, DECODE, (NULL),
+ ("filling error: %d", err), ret);
+ gst_buffer_unmap (inbuf, &imap);
+ goto out;
+ }
+ gst_buffer_unmap (inbuf, &imap);
+
+ if (GST_BUFFER_IS_DISCONT (inbuf))
+ flags |= AACDEC_INTR;
+ } else {
+ flags |= AACDEC_FLUSH;
+ }
+
+ if ((err =
+ aacDecoder_DecodeFrame (self->dec, (gint16 *) self->decode_buffer,
+ self->decode_buffer_size, flags)) != 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;
+ }
+
+ /* FIXME: Don't recalculate this on every buffer */
+ if (stream_info->numChannels == 1) {
+ pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
+ } else {
+ gint n_front = 0, n_side = 0, n_back = 0, n_lfe = 0;
+
+ /* FIXME: Can this be simplified somehow? */
+ for (i = 0; i < stream_info->numChannels; i++) {
+ if (stream_info->pChannelType[i] == ACT_FRONT) {
+ n_front++;
+ } else if (stream_info->pChannelType[i] == ACT_SIDE) {
+ n_side++;
+ } else if (stream_info->pChannelType[i] == ACT_BACK) {
+ n_back++;
+ } else if (stream_info->pChannelType[i] == ACT_LFE) {
+ n_lfe++;
+ } else {
+ GST_ERROR_OBJECT (self, "Channel type %d not supported",
+ stream_info->pChannelType[i]);
+ ret = GST_FLOW_NOT_NEGOTIATED;
+ goto out;
+ }
+ }
+
+ for (i = 0; i < stream_info->numChannels; i++) {
+ if (stream_info->pChannelType[i] == ACT_FRONT) {
+ if (stream_info->pChannelIndices[i] == 0) {
+ if (n_front & 1)
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
+ else if (n_front > 2)
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
+ else
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ } else if (stream_info->pChannelIndices[i] == 1) {
+ if ((n_front & 1) && n_front > 3)
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
+ else if (n_front & 1)
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ else if (n_front > 2)
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
+ else
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ } else if (stream_info->pChannelIndices[i] == 2) {
+ if ((n_front & 1) && n_front > 3)
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
+ else if (n_front & 1)
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ else if (n_front > 2)
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ else
+ g_assert_not_reached ();
+ } else if (stream_info->pChannelIndices[i] == 3) {
+ if ((n_front & 1) && n_front > 3)
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ else if (n_front & 1)
+ g_assert_not_reached ();
+ else if (n_front > 2)
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ else
+ g_assert_not_reached ();
+ } else if (stream_info->pChannelIndices[i] == 4) {
+ if ((n_front & 1) && n_front > 2)
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ else if (n_front & 1)
+ g_assert_not_reached ();
+ else if (n_front > 2)
+ g_assert_not_reached ();
+ else
+ g_assert_not_reached ();
+ } else {
+ GST_ERROR_OBJECT (self, "Front channel index %d not supported",
+ stream_info->pChannelIndices[i]);
+ ret = GST_FLOW_NOT_NEGOTIATED;
+ goto out;
+ }
+ } else if (stream_info->pChannelType[i] == ACT_SIDE) {
+ if (n_side & 1) {
+ GST_ERROR_OBJECT (self, "Odd number of side channels not supported");
+ ret = GST_FLOW_NOT_NEGOTIATED;
+ goto out;
+ } else if (stream_info->pChannelIndices[i] == 0) {
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
+ } else if (stream_info->pChannelIndices[i] == 1) {
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
+ } else {
+ GST_ERROR_OBJECT (self, "Side channel index %d not supported",
+ stream_info->pChannelIndices[i]);
+ ret = GST_FLOW_NOT_NEGOTIATED;
+ goto out;
+ }
+ } else if (stream_info->pChannelType[i] == ACT_BACK) {
+ if (stream_info->pChannelIndices[i] == 0) {
+ if (n_back & 1)
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
+ else
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
+ } else if (stream_info->pChannelIndices[i] == 1) {
+ if (n_back & 1)
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
+ else
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
+ } else if (stream_info->pChannelIndices[i] == 2) {
+ if (n_back & 1)
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
+ else
+ g_assert_not_reached ();
+ } else {
+ GST_ERROR_OBJECT (self, "Side channel index %d not supported",
+ stream_info->pChannelIndices[i]);
+ ret = GST_FLOW_NOT_NEGOTIATED;
+ goto out;
+ }
+ } else if (stream_info->pChannelType[i] == ACT_LFE) {
+ if (stream_info->pChannelIndices[i] == 0) {
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_LFE1;
+ } else {
+ GST_ERROR_OBJECT (self, "LFE channel index %d not supported",
+ stream_info->pChannelIndices[i]);
+ ret = GST_FLOW_NOT_NEGOTIATED;
+ goto out;
+ }
+ } else {
+ GST_ERROR_OBJECT (self, "Channel type %d not supported",
+ stream_info->pChannelType[i]);
+ ret = GST_FLOW_NOT_NEGOTIATED;
+ goto out;
+ }
+ }
+ }
+
+ memcpy (gst_pos, pos,
+ sizeof (GstAudioChannelPosition) * stream_info->numChannels);
+ if (!gst_audio_channel_positions_to_valid_order (gst_pos,
+ stream_info->numChannels)) {
+ ret = GST_FLOW_NOT_NEGOTIATED;
+ goto out;
+ }
+
+ need_reorder =
+ memcmp (pos, gst_pos,
+ sizeof (GstAudioChannelPosition) * stream_info->numChannels) != 0;
+
+ gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16,
+ stream_info->sampleRate, stream_info->numChannels, gst_pos);
+ if (!gst_audio_decoder_set_output_format (dec, &info)) {
+ GST_ERROR_OBJECT (self, "Failed to set output format");
+ ret = GST_FLOW_NOT_NEGOTIATED;
+ goto out;
+ }
+
+ outbuf =
+ gst_audio_decoder_allocate_output_buffer (dec,
+ stream_info->frameSize * GST_AUDIO_INFO_BPF (&info));
+ gst_buffer_fill (outbuf, 0, self->decode_buffer,
+ gst_buffer_get_size (outbuf));
+
+ if (need_reorder) {
+ gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_INFO_FORMAT (&info),
+ GST_AUDIO_INFO_CHANNELS (&info), pos, gst_pos);
+ }
+
+ ret = gst_audio_decoder_finish_frame (dec, outbuf, 1);
+
+out:
+
+ return ret;
+}
+
+static void
+gst_fdkaacdec_flush (GstAudioDecoder * dec, gboolean hard)
+{
+ GstFdkAacDec *self = GST_FDKAACDEC (dec);
+
+ if (self->dec) {
+ AAC_DECODER_ERROR err;
+ if ((err =
+ aacDecoder_DecodeFrame (self->dec, (gint16 *) self->decode_buffer,
+ self->decode_buffer_size, AACDEC_FLUSH)) != 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");
+}
projects / platform / upstream / gst-plugins-bad.git / commitdiff