[FFmpeg-cvslog] r14798 - trunk/libavcodec/alacenc.c
ramiro
subversion
Sun Aug 17 06:36:06 CEST 2008
Author: ramiro
Date: Sun Aug 17 06:36:06 2008
New Revision: 14798
Log:
Import ok'd parts of ALAC encoder from GSoC repo.
Added:
trunk/libavcodec/alacenc.c
Added: trunk/libavcodec/alacenc.c
==============================================================================
--- (empty file)
+++ trunk/libavcodec/alacenc.c Sun Aug 17 06:36:06 2008
@@ -0,0 +1,197 @@
+/**
+ * ALAC audio encoder
+ * Copyright (c) 2008 Jaikrishnan Menon <realityman at gmx.net>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "avcodec.h"
+#include "bitstream.h"
+#include "dsputil.h"
+#include "lpc.h"
+
+#define DEFAULT_FRAME_SIZE 4096
+#define DEFAULT_SAMPLE_SIZE 16
+#define MAX_CHANNELS 8
+#define ALAC_EXTRADATA_SIZE 36
+#define ALAC_FRAME_HEADER_SIZE 55
+#define ALAC_FRAME_FOOTER_SIZE 3
+
+#define ALAC_ESCAPE_CODE 0x1FF
+#define ALAC_MAX_LPC_ORDER 30
+
+ int interlacing_shift;
+ int interlacing_leftweight;
+ PutBitContext pbctx;
+ DSPContext dspctx;
+ AVCodecContext *avctx;
+} AlacEncodeContext;
+
+
+static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)
+{
+ int divisor, q, r;
+
+ k = FFMIN(k, s->rc.k_modifier);
+ divisor = (1<<k) - 1;
+ q = x / divisor;
+ r = x % divisor;
+
+ if(q > 8) {
+ // write escape code and sample value directly
+ put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE);
+ put_bits(&s->pbctx, write_sample_size, x);
+ } else {
+ if(q)
+ put_bits(&s->pbctx, q, (1<<q) - 1);
+ put_bits(&s->pbctx, 1, 0);
+
+ if(k != 1) {
+ if(r > 0)
+ put_bits(&s->pbctx, k, r+1);
+ else
+ put_bits(&s->pbctx, k-1, 0);
+ }
+ }
+}
+
+static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
+{
+ put_bits(&s->pbctx, 3, s->channels-1); // No. of channels -1
+ put_bits(&s->pbctx, 16, 0); // Seems to be zero
+ put_bits(&s->pbctx, 1, 1); // Sample count is in the header
+ put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field
+ put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim
+ put_bits(&s->pbctx, 32, s->avctx->frame_size); // No. of samples in the frame
+}
+
+static void write_compressed_frame(AlacEncodeContext *s)
+{
+ int i, j;
+
+ /* only simple mid/side decorrelation supported as of now */
+ alac_stereo_decorrelation(s);
+ put_bits(&s->pbctx, 8, s->interlacing_shift);
+ put_bits(&s->pbctx, 8, s->interlacing_leftweight);
+
+ for(i=0;i<s->channels;i++) {
+
+ calc_predictor_params(s, i);
+
+ put_bits(&s->pbctx, 4, 0); // prediction type : currently only type 0 has been RE'd
+ put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant);
+
+ put_bits(&s->pbctx, 3, s->rc.rice_modifier);
+ put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);
+ // predictor coeff. table
+ for(j=0;j<s->lpc[i].lpc_order;j++) {
+ put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
+ }
+ }
+
+ // apply lpc and entropy coding to audio samples
+
+ for(i=0;i<s->channels;i++) {
+ alac_linear_predictor(s, i);
+ alac_entropy_coder(s);
+ }
+}
+
+static av_cold int alac_encode_init(AVCodecContext *avctx)
+{
+ AlacEncodeContext *s = avctx->priv_data;
+ uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1);
+
+ avctx->frame_size = DEFAULT_FRAME_SIZE;
+ avctx->bits_per_sample = DEFAULT_SAMPLE_SIZE;
+ s->channels = avctx->channels;
+ s->samplerate = avctx->sample_rate;
+
+ if(avctx->sample_fmt != SAMPLE_FMT_S16) {
+ av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
+ return -1;
+ }
+
+ // Set default compression level
+ if(avctx->compression_level == FF_COMPRESSION_DEFAULT)
+ s->compression_level = 1;
+ else
+ s->compression_level = av_clip(avctx->compression_level, 0, 1);
+
+ // Initialize default Rice parameters
+ s->rc.history_mult = 40;
+ s->rc.initial_history = 10;
+ s->rc.k_modifier = 14;
+ s->rc.rice_modifier = 4;
+
+ s->max_coded_frame_size = (ALAC_FRAME_HEADER_SIZE + ALAC_FRAME_FOOTER_SIZE +
+ avctx->frame_size*s->channels*avctx->bits_per_sample)>>3;
+
+ s->write_sample_size = avctx->bits_per_sample + s->channels - 1; // FIXME: consider wasted_bytes
+
+ AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE);
+ AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c'));
+ AV_WB32(alac_extradata+12, avctx->frame_size);
+ AV_WB8 (alac_extradata+17, avctx->bits_per_sample);
+ AV_WB8 (alac_extradata+21, s->channels);
+ AV_WB32(alac_extradata+24, s->max_coded_frame_size);
+ AV_WB32(alac_extradata+28, s->samplerate*s->channels*avctx->bits_per_sample); // average bitrate
+ AV_WB32(alac_extradata+32, s->samplerate);
+
+ // Set relevant extradata fields
+ if(s->compression_level > 0) {
+ AV_WB8(alac_extradata+18, s->rc.history_mult);
+ AV_WB8(alac_extradata+19, s->rc.initial_history);
+ AV_WB8(alac_extradata+20, s->rc.k_modifier);
+ }
+
+ avctx->extradata = alac_extradata;
+ avctx->extradata_size = ALAC_EXTRADATA_SIZE;
+
+ avctx->coded_frame = avcodec_alloc_frame();
+ avctx->coded_frame->key_frame = 1;
+
+ s->avctx = avctx;
+ dsputil_init(&s->dspctx, avctx);
+
+ allocate_sample_buffers(s);
+
+ return 0;
+}
+
+static av_cold int alac_encode_close(AVCodecContext *avctx)
+{
+ AlacEncodeContext *s = avctx->priv_data;
+
+ av_freep(&avctx->extradata);
+ avctx->extradata_size = 0;
+ av_freep(&avctx->coded_frame);
+ free_sample_buffers(s);
+ return 0;
+}
+
+AVCodec alac_encoder = {
+ "alac",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_ALAC,
+ sizeof(AlacEncodeContext),
+ alac_encode_init,
+ alac_encode_frame,
+ alac_encode_close,
+ .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
+ .long_name = "ALAC (Apple Lossless Audio Codec)",
+};
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