mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-21 07:46:38 +00:00
56b822f9f6
Avoids useless check of downstream caps when handling an
accept-caps query
Elements: dtsdec, faad, gsmdec, mpg123audiodec, opusdec,
sbcdec, adpcmdec, sirendec
499 lines
14 KiB
C
499 lines
14 KiB
C
/* GStreamer
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* Copyright (C) 2009 Pioneers of the Inevitable <songbird@songbirdnest.com>
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*
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* Authors: Michael Smith <msmith@songbirdnest.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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/* Based on ADPCM decoders in libsndfile,
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Copyright (C) 1999-2002 Erik de Castro Lopo <erikd@zip.com.au
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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 <gst/gst.h>
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#include <gst/audio/gstaudiodecoder.h>
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#define GST_TYPE_ADPCM_DEC \
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(adpcmdec_get_type ())
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#define GST_ADPCM_DEC(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_ADPCM_DEC, ADPCMDec))
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#define GST_CAT_DEFAULT adpcmdec_debug
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GST_DEBUG_CATEGORY_STATIC (adpcmdec_debug);
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static GstStaticPadTemplate adpcmdec_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-adpcm, "
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" layout=(string){microsoft, dvi}, "
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" block_align = (int) [64, 8192], "
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" rate = (int)[ 1, MAX ], " "channels = (int)[1,2];")
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);
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static GstStaticPadTemplate adpcmdec_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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"rate = (int) [1, MAX], channels = (int) [1,2]")
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);
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enum adpcm_layout
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{
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LAYOUT_ADPCM_MICROSOFT,
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LAYOUT_ADPCM_DVI
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};
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typedef struct _ADPCMDecClass
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{
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GstAudioDecoderClass parent_class;
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} ADPCMDecClass;
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typedef struct _ADPCMDec
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{
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GstAudioDecoder parent;
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enum adpcm_layout layout;
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int rate;
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int channels;
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int blocksize;
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} ADPCMDec;
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GType adpcmdec_get_type (void);
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G_DEFINE_TYPE (ADPCMDec, adpcmdec, GST_TYPE_AUDIO_DECODER);
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static gboolean
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adpcmdec_set_format (GstAudioDecoder * bdec, GstCaps * in_caps)
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{
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ADPCMDec *dec = (ADPCMDec *) (bdec);
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GstStructure *structure = gst_caps_get_structure (in_caps, 0);
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const gchar *layout;
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GstAudioInfo info;
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layout = gst_structure_get_string (structure, "layout");
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if (!layout)
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return FALSE;
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if (g_str_equal (layout, "microsoft"))
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dec->layout = LAYOUT_ADPCM_MICROSOFT;
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else if (g_str_equal (layout, "dvi"))
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dec->layout = LAYOUT_ADPCM_DVI;
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else
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return FALSE;
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if (!gst_structure_get_int (structure, "block_align", &dec->blocksize))
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dec->blocksize = -1; /* Not provided */
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if (!gst_structure_get_int (structure, "rate", &dec->rate))
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return FALSE;
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if (!gst_structure_get_int (structure, "channels", &dec->channels))
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return FALSE;
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gst_audio_info_init (&info);
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gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, dec->rate,
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dec->channels, NULL);
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gst_audio_decoder_set_output_format (bdec, &info);
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return TRUE;
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}
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/*=====================================================================
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* From libsndfile:
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*
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* MS ADPCM Block Layout.
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* ======================
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* Block is usually 256, 512 or 1024 bytes depending on sample rate.
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* For a mono file, the block is laid out as follows:
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* byte purpose
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* 0 block predictor [0..6]
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* 1,2 initial idelta (positive)
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* 3,4 sample 1
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* 5,6 sample 0
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* 7..n packed bytecodes
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*
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* For a stereo file, the block is laid out as follows:
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* byte purpose
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* 0 block predictor [0..6] for left channel
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* 1 block predictor [0..6] for right channel
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* 2,3 initial idelta (positive) for left channel
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* 4,5 initial idelta (positive) for right channel
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* 6,7 sample 1 for left channel
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* 8,9 sample 1 for right channel
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* 10,11 sample 0 for left channel
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* 12,13 sample 0 for right channel
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* 14..n packed bytecodes
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*
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*=====================================================================
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*/
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static const int AdaptationTable[] = {
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230, 230, 230, 230, 307, 409, 512, 614,
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768, 614, 512, 409, 307, 230, 230, 230
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};
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static const int AdaptCoeff1[] = {
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256, 512, 0, 192, 240, 460, 392
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};
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static const int AdaptCoeff2[] = {
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0, -256, 0, 64, 0, -208, -232
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};
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static gint16
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read_sample (const guint8 * data)
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{
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guint16 val = data[0] | (data[1] << 8);
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return *((gint16 *) & val);
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}
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/* Decode a single block of data from 'data', storing 'n_samples' decoded 16 bit
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samples in 'samples'.
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All buffer lengths have been verified by the caller
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*/
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static gboolean
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adpcmdec_decode_ms_block (ADPCMDec * dec, int n_samples, const guint8 * data,
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gint16 * samples)
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{
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gint16 pred[2];
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gint16 idelta[2];
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int idx; /* Current byte offset in 'data' */
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int i; /* Current sample index in 'samples' */
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/* Read the block header, verify for sanity */
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if (dec->channels == 1) {
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pred[0] = data[0];
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idelta[0] = read_sample (data + 1);
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samples[1] = read_sample (data + 3);
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samples[0] = read_sample (data + 5);
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idx = 7;
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i = 2;
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if (pred[0] < 0 || pred[0] > 6) {
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GST_WARNING_OBJECT (dec, "Invalid block predictor");
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return FALSE;
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}
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}
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else {
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pred[0] = data[0];
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pred[1] = data[1];
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idelta[0] = read_sample (data + 2);
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idelta[1] = read_sample (data + 4);
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samples[2] = read_sample (data + 6);
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samples[3] = read_sample (data + 8);
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samples[0] = read_sample (data + 10);
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samples[1] = read_sample (data + 12);
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idx = 14;
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i = 4;
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if (pred[0] < 0 || pred[0] > 6 || pred[1] < 0 || pred[1] > 6) {
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GST_WARNING_OBJECT (dec, "Invalid block predictor");
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return FALSE;
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}
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}
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for (; i < n_samples; i++) {
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int chan = i % dec->channels;
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int bytecode;
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int delta;
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int current;
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int predict;
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if (i % 2 == 0) {
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bytecode = (data[idx] >> 4) & 0x0F;
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} else {
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bytecode = data[idx] & 0x0F;
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idx++;
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}
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delta = idelta[chan];
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idelta[chan] = (AdaptationTable[bytecode] * delta) >> 8;
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if (idelta[chan] < 16)
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idelta[chan] = 16;
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/* Bytecode is used above as an index into the table. Below, it's used
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as a signed 4-bit value; convert appropriately */
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if (bytecode & 0x8)
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bytecode -= 0x10;
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predict = ((samples[i - dec->channels] * AdaptCoeff1[pred[chan]]) +
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(samples[i - 2 * dec->channels] * AdaptCoeff2[pred[chan]])
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) >> 8;
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current = (bytecode * delta) + predict;
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/* Clamp to 16 bits, store decoded sample */
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samples[i] = CLAMP (current, G_MININT16, G_MAXINT16);
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}
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return TRUE;
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}
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static const int ima_indx_adjust[16] = {
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-1, -1, -1, -1, 2, 4, 6, 8, -1, -1, -1, -1, 2, 4, 6, 8,
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};
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static const int ima_step_size[89] = {
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7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
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50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, 143, 157, 173, 190, 209, 230,
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253, 279, 307, 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, 876, 963,
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1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327,
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3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, 9493, 10442,
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11487, 12635, 13899, 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794,
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32767
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};
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/* Decode a single block of data from 'data', storing 'n_samples' decoded 16 bit
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samples in 'samples'.
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All buffer lengths have been verified by the caller
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*/
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static gboolean
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adpcmdec_decode_ima_block (ADPCMDec * dec, int n_samples, const guint8 * data,
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gint16 * samples)
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{
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gint16 stepindex[2];
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int channel;
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int idx;
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int i, j;
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int sample;
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if ((n_samples - dec->channels) % 8 != 0) {
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GST_WARNING_OBJECT (dec, "Input not correct size");
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return FALSE;
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}
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for (channel = 0; channel < dec->channels; channel++) {
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samples[channel] = read_sample (data + channel * 4);
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stepindex[channel] = MIN (data[channel * 4 + 2], 88);
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if (data[channel * 4 + 3] != 0) {
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GST_WARNING_OBJECT (dec, "Synchronisation error");
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return FALSE;
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}
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}
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i = dec->channels;
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idx = 4 * dec->channels;
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while (i < n_samples) {
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for (channel = 0; channel < dec->channels; channel++) {
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sample = i + channel;
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for (j = 0; j < 8; j++) {
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int bytecode;
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int step;
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int diff;
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if (j % 2 == 0) {
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bytecode = data[idx] & 0x0F;
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} else {
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bytecode = (data[idx] >> 4) & 0x0F;
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idx++;
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}
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step = ima_step_size[stepindex[channel]];
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diff = (2 * (bytecode & 0x7) * step + step) / 8;
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if (bytecode & 8)
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diff = -diff;
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samples[sample] =
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CLAMP (samples[sample - dec->channels] + diff, G_MININT16,
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G_MAXINT16);
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stepindex[channel] =
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CLAMP (stepindex[channel] + ima_indx_adjust[bytecode], 0, 88);
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sample += dec->channels;
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}
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}
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i += 8 * dec->channels;
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}
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return TRUE;
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}
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static GstBuffer *
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adpcmdec_decode_block (ADPCMDec * dec, const guint8 * data, int blocksize)
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{
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gboolean res = FALSE;
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GstBuffer *outbuf = NULL;
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int outsize;
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int samples;
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GstMapInfo omap;
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if (dec->layout == LAYOUT_ADPCM_MICROSOFT) {
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/* Each block has a 3 byte header per channel, plus 4 bytes per channel to
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give two initial sample values per channel. Then the remainder gives
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two samples per byte */
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if (blocksize < 7 * dec->channels)
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goto exit;
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samples = (blocksize - 7 * dec->channels) * 2 + 2 * dec->channels;
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outsize = 2 * samples;
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outbuf = gst_buffer_new_and_alloc (outsize);
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gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
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res = adpcmdec_decode_ms_block (dec, samples, data, (gint16 *) omap.data);
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gst_buffer_unmap (outbuf, &omap);
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} else if (dec->layout == LAYOUT_ADPCM_DVI) {
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/* Each block has a 4 byte header per channel, include an initial sample.
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Then the remainder gives two samples per byte */
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if (blocksize < 4 * dec->channels)
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goto exit;
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samples = (blocksize - 4 * dec->channels) * 2 + dec->channels;
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outsize = 2 * samples;
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outbuf = gst_buffer_new_and_alloc (outsize);
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gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
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res = adpcmdec_decode_ima_block (dec, samples, data, (gint16 *) omap.data);
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gst_buffer_unmap (outbuf, &omap);
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} else {
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GST_WARNING_OBJECT (dec, "Unknown layout");
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}
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if (!res) {
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if (outbuf)
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gst_buffer_unref (outbuf);
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outbuf = NULL;
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GST_WARNING_OBJECT (dec, "Decode of block failed");
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}
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exit:
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return outbuf;
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}
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static GstFlowReturn
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adpcmdec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
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gint * offset, gint * length)
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{
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ADPCMDec *dec = (ADPCMDec *) (bdec);
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guint size;
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size = gst_adapter_available (adapter);
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g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
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if (dec->blocksize < 0) {
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/* No explicit blocksize; we just process one input buffer at a time */
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*offset = 0;
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*length = size;
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} else {
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if (size >= dec->blocksize) {
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*offset = 0;
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*length = dec->blocksize;
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} else {
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return GST_FLOW_EOS;
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}
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}
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return GST_FLOW_OK;
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}
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static GstFlowReturn
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adpcmdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
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{
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ADPCMDec *dec = (ADPCMDec *) (bdec);
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GstFlowReturn ret = GST_FLOW_OK;
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GstMapInfo map;
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GstBuffer *outbuf = NULL;
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/* no fancy draining */
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if (G_UNLIKELY (!buffer))
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return GST_FLOW_OK;
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if (!dec->blocksize)
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return GST_FLOW_NOT_NEGOTIATED;
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gst_buffer_map (buffer, &map, GST_MAP_READ);
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outbuf = adpcmdec_decode_block (dec, map.data, dec->blocksize);
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gst_buffer_unmap (buffer, &map);
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if (outbuf == NULL) {
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GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
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("frame decode failed"), ret);
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}
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if (ret == GST_FLOW_OK)
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ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
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return ret;
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}
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static gboolean
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adpcmdec_start (GstAudioDecoder * bdec)
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{
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ADPCMDec *dec = (ADPCMDec *) bdec;
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GST_DEBUG_OBJECT (dec, "start");
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dec->blocksize = 0;
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dec->rate = 0;
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dec->channels = 0;
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return TRUE;
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}
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static gboolean
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adpcmdec_stop (GstAudioDecoder * dec)
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{
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GST_DEBUG_OBJECT (dec, "stop");
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return TRUE;
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}
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static void
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adpcmdec_init (ADPCMDec * dec)
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{
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gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
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gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
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(dec), TRUE);
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
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}
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static void
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adpcmdec_class_init (ADPCMDecClass * klass)
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{
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GstElementClass *element_class = (GstElementClass *) klass;
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GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) klass;
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&adpcmdec_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&adpcmdec_src_template));
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gst_element_class_set_static_metadata (element_class, "ADPCM decoder",
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"Codec/Decoder/Audio",
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"Decode MS and IMA ADPCM audio",
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"Pioneers of the Inevitable <songbird@songbirdnest.com>");
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base_class->start = GST_DEBUG_FUNCPTR (adpcmdec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (adpcmdec_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (adpcmdec_set_format);
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base_class->parse = GST_DEBUG_FUNCPTR (adpcmdec_parse);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (adpcmdec_handle_frame);
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}
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static gboolean
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plugin_init (GstPlugin * plugin)
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{
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GST_DEBUG_CATEGORY_INIT (adpcmdec_debug, "adpcmdec", 0, "ADPCM Decoders");
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if (!gst_element_register (plugin, "adpcmdec", GST_RANK_PRIMARY,
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GST_TYPE_ADPCM_DEC)) {
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return FALSE;
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}
|
|
return TRUE;
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, adpcmdec,
|
|
"ADPCM decoder", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
|
|
GST_PACKAGE_ORIGIN);
|