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gstreamer/ext/fdkaac/gstfdkaacdec.c
Stéphane Cerveau 4b72e8cad5 fdkaacdec: add support for mpegversion=2 
Fix for #1199
2020-02-04 07:52:22 +00:00

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/*
* 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
*/
#ifndef HAVE_FDK_AAC_0_1_4
#define AAC_PCM_MAX_OUTPUT_CHANNELS AAC_PCM_OUTPUT_CHANNELS
#define CHANNELS_CAPS_STR "channels = (int) [1, 6]"
#else
#define CHANNELS_CAPS_STR "channels = (int) [1, 8]"
#endif
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg, "
"mpegversion = (int) {2, 4}, "
"stream-format = (string) { adts, adif, raw }, " CHANNELS_CAPS_STR)
);
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_CAPS_STR)
);
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_buffer_unref (codec_data);
GST_ERROR_OBJECT (self, "Invalid codec_data: %d", err);
return FALSE;
}
gst_buffer_unmap (codec_data, &map);
gst_buffer_unref (codec_data);
}
err = aacDecoder_SetParam (self->dec, AAC_PCM_MAX_OUTPUT_CHANNELS, 0);
if (err != AAC_DEC_OK) {
GST_ERROR_OBJECT (self, "Failed to disable downmixing: %d", err);
return FALSE;
}
/* Choose WAV channel mapping to get interleaving even with libfdk-aac 2.0.0
* The pChannelIndices retain the indices from the standard MPEG mapping so
* we're agnostic to the actual order. */
err = aacDecoder_SetParam (self->dec, AAC_PCM_OUTPUT_CHANNEL_MAPPING, 1);
if (err != 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 * 2048;
self->decode_buffer = g_new (gint16, 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_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) {
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_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_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_BACK) {
if (stream_info->pChannelIndices[i] == 0) {
if (n_back & 1)
pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
else if (n_back > 2)
pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
else
pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
} else if (stream_info->pChannelIndices[i] == 1) {
if ((n_back & 1) && n_back > 3)
pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
else if (n_back & 1)
pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
else if (n_back > 2)
pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
else
pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
} else if (stream_info->pChannelIndices[i] == 2) {
if ((n_back & 1) && n_back > 3)
pos[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
else if (n_back & 1)
pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
else if (n_back > 2)
pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
else
g_assert_not_reached ();
} else if (stream_info->pChannelIndices[i] == 3) {
if ((n_back & 1) && n_back > 3)
pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
else if (n_back & 1)
g_assert_not_reached ();
else if (n_back > 2)
pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
else
g_assert_not_reached ();
} else if (stream_info->pChannelIndices[i] == 4) {
if ((n_back & 1) && n_back > 3)
pos[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
else if (n_back & 1)
g_assert_not_reached ();
else if (n_back > 2)
g_assert_not_reached ();
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);
}
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");
}
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