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479 lines
13 KiB
C
479 lines
13 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: Peter van Hardenberg <pvh@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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/* Based on ADPCM encoders 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/gstaudioencoder.h>
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#define GST_TYPE_ADPCM_ENC \
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(adpcmenc_get_type ())
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#define GST_TYPE_ADPCMENC_LAYOUT \
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(adpcmenc_layout_get_type ())
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#define GST_ADPCM_ENC(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_ADPCM_ENC, ADPCMEnc))
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#define GST_CAT_DEFAULT adpcmenc_debug
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GST_DEBUG_CATEGORY_STATIC (adpcmenc_debug);
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static GstStaticPadTemplate adpcmenc_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-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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static GstStaticPadTemplate adpcmenc_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-adpcm, "
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" layout=(string){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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#define MIN_ADPCM_BLOCK_SIZE 64
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#define MAX_ADPCM_BLOCK_SIZE 8192
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#define DEFAULT_ADPCM_BLOCK_SIZE 1024
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#define DEFAULT_ADPCM_LAYOUT LAYOUT_ADPCM_DVI
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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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enum adpcm_properties
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{
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ARG_0,
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ARG_BLOCK_SIZE,
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ARG_LAYOUT
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};
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enum adpcm_layout
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{
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LAYOUT_ADPCM_DVI
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};
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static GType
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adpcmenc_layout_get_type (void)
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{
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static GType adpcmenc_layout_type = 0;
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if (!adpcmenc_layout_type) {
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static GEnumValue layout_types[] = {
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{LAYOUT_ADPCM_DVI, "DVI/IMA APDCM", "dvi"},
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{0, NULL, NULL},
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};
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adpcmenc_layout_type = g_enum_register_static ("GstADPCMEncLayout",
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layout_types);
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}
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return adpcmenc_layout_type;
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}
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typedef struct _ADPCMEncClass
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{
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GstAudioEncoderClass parent_class;
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} ADPCMEncClass;
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typedef struct _ADPCMEnc
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{
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GstAudioEncoder 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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int samples_per_block;
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guint8 step_index[2];
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} ADPCMEnc;
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GType adpcmenc_get_type (void);
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G_DEFINE_TYPE (ADPCMEnc, adpcmenc, GST_TYPE_AUDIO_ENCODER);
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static gboolean
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adpcmenc_setup (ADPCMEnc * enc)
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{
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const int DVI_IMA_HEADER_SIZE = 4;
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const int ADPCM_SAMPLES_PER_BYTE = 2;
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guint64 sample_bytes;
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const char *layout;
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GstCaps *caps;
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switch (enc->layout) {
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case LAYOUT_ADPCM_DVI:
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layout = "dvi";
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/* IMA ADPCM includes a 4-byte header per channel, */
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sample_bytes = enc->blocksize - (DVI_IMA_HEADER_SIZE * enc->channels);
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/* two samples per byte, plus a single sample in the header. */
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enc->samples_per_block =
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((sample_bytes * ADPCM_SAMPLES_PER_BYTE) / enc->channels) + 1;
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break;
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default:
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GST_WARNING_OBJECT (enc, "Invalid layout");
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return FALSE;
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}
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caps = gst_caps_new_simple ("audio/x-adpcm",
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"rate", G_TYPE_INT, enc->rate,
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"channels", G_TYPE_INT, enc->channels,
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"layout", G_TYPE_STRING, layout,
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"block_align", G_TYPE_INT, enc->blocksize, NULL);
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gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps);
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gst_caps_unref (caps);
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/* Step index state is carried between blocks. */
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enc->step_index[0] = 0;
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enc->step_index[1] = 0;
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return TRUE;
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}
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static gboolean
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adpcmenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
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{
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ADPCMEnc *enc = (ADPCMEnc *) (benc);
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enc->rate = GST_AUDIO_INFO_RATE (info);
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enc->channels = GST_AUDIO_INFO_CHANNELS (info);
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if (!adpcmenc_setup (enc))
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return FALSE;
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/* report needs to base class */
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gst_audio_encoder_set_frame_samples_min (benc, enc->samples_per_block);
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gst_audio_encoder_set_frame_samples_max (benc, enc->samples_per_block);
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gst_audio_encoder_set_frame_max (benc, 1);
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return TRUE;
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}
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static void
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adpcmenc_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec)
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{
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ADPCMEnc *enc = GST_ADPCM_ENC (object);
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switch (prop_id) {
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case ARG_BLOCK_SIZE:
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enc->blocksize = g_value_get_int (value);
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break;
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case ARG_LAYOUT:
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enc->layout = g_value_get_enum (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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adpcmenc_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec)
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{
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ADPCMEnc *enc = GST_ADPCM_ENC (object);
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switch (prop_id) {
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case ARG_BLOCK_SIZE:
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g_value_set_int (value, enc->blocksize);
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break;
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case ARG_LAYOUT:
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g_value_set_enum (value, enc->layout);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static guint8
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adpcmenc_encode_ima_sample (gint16 sample, gint16 * prev_sample,
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guint8 * stepindex)
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{
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const int NEGATIVE_SIGN_BIT = 0x8;
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int diff, vpdiff, mask, step;
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int bytecode = 0x0;
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diff = sample - *prev_sample;
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step = ima_step_size[*stepindex];
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vpdiff = step >> 3;
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if (diff < 0) {
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diff = -diff;
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bytecode = NEGATIVE_SIGN_BIT;
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}
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mask = 0x4;
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while (mask > 0) {
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if (diff >= step) {
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bytecode |= mask;
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diff -= step;
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vpdiff += step;
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}
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step >>= 1;
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mask >>= 1;
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}
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if (bytecode & 8) {
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vpdiff = -vpdiff;
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}
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*prev_sample = CLAMP (*prev_sample + vpdiff, G_MININT16, G_MAXINT16);
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*stepindex = CLAMP (*stepindex + ima_indx_adjust[bytecode], 0, 88);
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return bytecode;
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}
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static gboolean
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adpcmenc_encode_ima_block (ADPCMEnc * enc, const gint16 * samples,
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guint8 * outbuf)
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{
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const int HEADER_SIZE = 4;
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gint16 prev_sample[2] = { 0, 0 };
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guint32 write_pos = 0;
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guint32 read_pos = 0;
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guint8 channel = 0;
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/* Write a header for each channel.
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* The header consists of a sixteen-bit predicted sound value,
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* and an eight bit step_index, carried forward from any previous block.
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* These allow seeking within the file.
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*/
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for (channel = 0; channel < enc->channels; channel++) {
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write_pos = channel * HEADER_SIZE;
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outbuf[write_pos + 0] = (samples[channel] & 0xFF);
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outbuf[write_pos + 1] = (samples[channel] >> 8) & 0xFF;
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outbuf[write_pos + 2] = enc->step_index[channel];
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outbuf[write_pos + 3] = 0;
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prev_sample[channel] = samples[channel];
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}
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/* raw-audio looks like this for a stereo stream:
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* [ L, R, L, R, L, R ... ]
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* encoded audio is in eight-sample blocks, two samples to a byte thusly:
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* [ LL, LL, LL, LL, RR, RR, RR, RR ... ]
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*/
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write_pos = HEADER_SIZE * enc->channels;
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read_pos = enc->channels; /* the first sample is in the header. */
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while (write_pos < enc->blocksize) {
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gint8 CHANNEL_CHUNK_SIZE = 8;
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for (channel = 0; channel < enc->channels; channel++) {
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/* convert eight samples (four bytes) per channel, then swap */
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guint32 channel_chunk_base = read_pos + channel;
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gint8 chunk;
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for (chunk = 0; chunk < CHANNEL_CHUNK_SIZE; chunk++) {
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guint8 packed_byte = 0, encoded_sample;
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encoded_sample =
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adpcmenc_encode_ima_sample (samples[channel_chunk_base +
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(chunk * enc->channels)], &prev_sample[channel],
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&enc->step_index[channel]);
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packed_byte |= encoded_sample & 0x0F;
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chunk++;
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encoded_sample =
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adpcmenc_encode_ima_sample (samples[channel_chunk_base +
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(chunk * enc->channels)], &prev_sample[channel],
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&enc->step_index[channel]);
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packed_byte |= encoded_sample << 4 & 0xF0;
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outbuf[write_pos++] = packed_byte;
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}
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}
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/* advance to the next block of 8 samples per channel */
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read_pos += CHANNEL_CHUNK_SIZE * enc->channels;
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if (read_pos > enc->samples_per_block * enc->channels) {
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GST_LOG ("Ran past the end. (Reading %i of %i.)", read_pos,
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enc->samples_per_block);
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}
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}
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return TRUE;
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}
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static GstBuffer *
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adpcmenc_encode_block (ADPCMEnc * enc, const gint16 * samples, int blocksize)
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{
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gboolean res = FALSE;
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GstBuffer *outbuf = NULL;
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GstMapInfo omap;
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if (enc->layout == LAYOUT_ADPCM_DVI) {
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outbuf = gst_buffer_new_and_alloc (enc->blocksize);
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gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
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res = adpcmenc_encode_ima_block (enc, samples, omap.data);
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gst_buffer_unmap (outbuf, &omap);
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} else {
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/* should not happen afaics */
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g_assert_not_reached ();
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GST_WARNING_OBJECT (enc, "Unknown layout");
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res = FALSE;
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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 (enc, "Encode of block failed");
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}
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return outbuf;
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}
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static GstFlowReturn
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adpcmenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
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{
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ADPCMEnc *enc = (ADPCMEnc *) (benc);
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GstFlowReturn ret = GST_FLOW_OK;
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gint16 *samples;
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GstBuffer *outbuf;
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int input_bytes_per_block;
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const int BYTES_PER_SAMPLE = 2;
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GstMapInfo map;
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/* we don't deal with squeezing remnants, so simply discard those */
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if (G_UNLIKELY (buffer == NULL)) {
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GST_DEBUG_OBJECT (benc, "no data");
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goto done;
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}
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input_bytes_per_block =
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enc->samples_per_block * BYTES_PER_SAMPLE * enc->channels;
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gst_buffer_map (buffer, &map, GST_MAP_READ);
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if (G_UNLIKELY (map.size < input_bytes_per_block)) {
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GST_DEBUG_OBJECT (enc, "discarding trailing data %d", (gint) map.size);
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gst_buffer_unmap (buffer, &map);
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ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
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goto done;
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}
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samples = (gint16 *) map.data;
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outbuf = adpcmenc_encode_block (enc, samples, enc->blocksize);
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gst_buffer_unmap (buffer, &map);
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ret = gst_audio_encoder_finish_frame (benc, outbuf, enc->samples_per_block);
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done:
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return ret;
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}
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static gboolean
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adpcmenc_start (GstAudioEncoder * enc)
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{
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GST_DEBUG_OBJECT (enc, "start");
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return TRUE;
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}
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static gboolean
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adpcmenc_stop (GstAudioEncoder * enc)
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{
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GST_DEBUG_OBJECT (enc, "stop");
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return TRUE;
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}
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static void
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adpcmenc_init (ADPCMEnc * enc)
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{
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/* Set defaults. */
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enc->blocksize = DEFAULT_ADPCM_BLOCK_SIZE;
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enc->layout = DEFAULT_ADPCM_LAYOUT;
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}
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static void
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adpcmenc_class_init (ADPCMEncClass * klass)
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{
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GObjectClass *gobjectclass = (GObjectClass *) klass;
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GstElementClass *element_class = (GstElementClass *) klass;
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GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) klass;
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gobjectclass->set_property = adpcmenc_set_property;
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gobjectclass->get_property = adpcmenc_get_property;
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&adpcmenc_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&adpcmenc_src_template));
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gst_element_class_set_details_simple (element_class, "ADPCM encoder",
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"Codec/Encoder/Audio",
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"Encode ADPCM audio",
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"Pioneers of the Inevitable <songbird@songbirdnest.com>");
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base_class->start = GST_DEBUG_FUNCPTR (adpcmenc_start);
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base_class->stop = GST_DEBUG_FUNCPTR (adpcmenc_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (adpcmenc_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (adpcmenc_handle_frame);
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g_object_class_install_property (gobjectclass, ARG_LAYOUT,
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g_param_spec_enum ("layout", "Layout",
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"Layout for output stream",
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GST_TYPE_ADPCMENC_LAYOUT, DEFAULT_ADPCM_LAYOUT,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobjectclass, ARG_BLOCK_SIZE,
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g_param_spec_int ("blockalign", "Block Align",
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"Block size for output stream",
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MIN_ADPCM_BLOCK_SIZE, MAX_ADPCM_BLOCK_SIZE,
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DEFAULT_ADPCM_BLOCK_SIZE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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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 (adpcmenc_debug, "adpcmenc", 0, "ADPCM Encoders");
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if (!gst_element_register (plugin, "adpcmenc", GST_RANK_PRIMARY,
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GST_TYPE_ADPCM_ENC)) {
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return FALSE;
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}
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return TRUE;
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}
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GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, adpcmenc,
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"ADPCM encoder", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
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GST_PACKAGE_ORIGIN);
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