[FFmpeg-soc] [soc]: r2262 - alacenc/alacenc.c

[jai_menon]

Author: jai_menon
Date: Fri May 30 12:06:38 2008
New Revision: 2262

Log:
initial commit: basic encoder framework, supports alac verbatim mode

Added:
   alacenc/alacenc.c

Added: alacenc/alacenc.c
==============================================================================
--- (empty file)
+++ alacenc/alacenc.c	Fri May 30 12:06:38 2008
@@ -0,0 +1,145 @@
+/**
+ * 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"
+
+#define DEFAULT_FRAME_SIZE        4096
+#define DEFAULT_SAMPLE_SIZE       16
+
+#define ALAC_EXTRADATA_SIZE       36
+#define ALAC_FRAME_HEADER_SIZE    55
+#define ALAC_FRAME_FOOTER_SIZE    3
+
+typedef struct AlacEncodeContext {
+    int channels;
+    int samplerate;
+    int compression_level;
+    int max_coded_frame_size;
+    AVCodecContext *avctx;
+} AlacEncodeContext;
+
+/**
+ * put_sbits
+ * @param pb PutBitContext pointer
+ * @param bits Number of bits to be output
+ * @param val Data Bits
+ */
+static void put_sbits(PutBitContext *pb, int bits, int32_t val)
+{
+    put_bits(pb, bits, val & ((1<<bits)-1));
+}
+
+static void write_frame_header(AlacEncodeContext *s, PutBitContext *pbctx)
+{
+    put_bits(pbctx, 3,  s->channels-1);         // No. of channels -1
+    put_bits(pbctx, 16, 0);                     // Seems to be zero
+    put_bits(pbctx, 1,  1);                     // Sample count is in the header
+    put_bits(pbctx, 2,  0);                     // FIXME: Wasted bytes field
+    put_bits(pbctx, 1,  1);                     // Audio block is verbatim
+    put_bits(pbctx, 32, s->avctx->frame_size);  // No. of samples in the frame
+}
+
+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;
+    }
+
+    s->max_coded_frame_size = (ALAC_FRAME_HEADER_SIZE + ALAC_FRAME_FOOTER_SIZE +
+                               avctx->frame_size*s->channels*avctx->bits_per_sample)>>3;
+
+    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);
+
+    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;
+    return 0;
+}
+
+static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
+                             int buf_size, void *data)
+{
+    PutBitContext pb;
+    AlacEncodeContext *s = avctx->priv_data;
+    int16_t *samples;
+    int i, ch;
+
+    if(buf_size < s->max_coded_frame_size) {
+        av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n");
+        return -1;
+    }
+
+    init_put_bits(&pb, frame, buf_size);
+    write_frame_header(s, &pb);
+
+    for(ch=0; ch<s->channels; ch++) {
+        samples = (int16_t *)data + ch;
+        for(i=0; i<avctx->frame_size; i++) {
+            put_sbits(&pb, 16, *samples);
+            samples += s->channels;
+        }
+    }
+
+    put_bits(&pb, 3, 7);
+    flush_put_bits(&pb);
+    return(put_bits_count(&pb)>>3);
+}
+
+static av_cold int alac_encode_close(AVCodecContext *avctx)
+{
+    av_freep(&avctx->extradata);
+    avctx->extradata_size = 0;
+    av_freep(&avctx->coded_frame);
+    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)",
+};