Import ok'd parts of ALAC encoder from GSoC repo.

Originally committed as revision 14798 to svn://svn.ffmpeg.org/ffmpeg/trunk

libavcodec/alacenc.c

@@ -0,0 +1,197 @@
+/**
+ * ALAC audio encoder
+ * Copyright (c) 2008  Jaikrishnan Menon <realityman@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)",
+};