Author: banan
Date: Thu Mar 20 14:40:49 2008
New Revision: 2023
Log:
dca encoder skeleton
Added:
dcaenc/
dcaenc/checkout.sh (contents, props changed)
dcaenc/dcaenc.c
dcaenc/dcaenc.h
dcaenc/ffmpeg.patch
dcaenc/readme
Added: dcaenc/checkout.sh
==============================================================================
--- (empty file)
+++ dcaenc/checkout.sh Thu Mar 20 14:40:49 2008
@@ -0,0 +1,11 @@
+echo "checking out ffmpeg svn"
+svn checkout svn://svn.mplayerhq.hu/ffmpeg/trunk/ ffmpeg -r 12510
+echo "Done, now just do a regular configure and make to build."
+echo "patching ffmpeg"
+cd ffmpeg
+patch -p0 <../ffmpeg.patch
+cd ..
+echo "linkling the dca files to ffmpeg/libavcodec"
+ln -s ../../dcaenc.c ffmpeg/libavcodec/dcaenc.c
+ln -s ../../dcaenc.h ffmpeg/libavcodec/dcaenc.h
+echo "all done recompile ffmpeg"
Added: dcaenc/dcaenc.c
==============================================================================
--- (empty file)
+++ dcaenc/dcaenc.c Thu Mar 20 14:40:49 2008
@@ -0,0 +1,379 @@
+/*
+ * DCA encoder
+ * Copyright (C) 2008 Alexander E. Patrakov
+ *
+ * This file is part of FFmpeg.
+ *
+ * When this file is part of FFmpeg it can be licensed under the LGPL 2.1
+ * currently it is a wip
+ */
+
+/* FFmpeg port by Benjamin Larsson */
+
+
+#include "avcodec.h"
+#include "bitstream.h"
+#include "dcaenc.h"
+
+#define MAX_CHANNELS (2)
+#define DCA_SUBBANDS_32 (32)
+#define DCA_MAX_FRAME_SIZE (16383)
+#define FIXED_FRAME_SIZE (7167)
+
+typedef struct {
+ PutBitContext pb;
+ int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
+ int start[MAX_CHANNELS];
+
+ int32_t pcm[DCA_SUBBANDS_32];
+ int32_t subband[64][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */
+ int16_t st_samples[4096];
+ int fill_samples;
+} DCAContext;
+
+static int32_t cos_table[128];
+
+
+static inline int32_t mul32(int32_t a, int32_t b)
+{
+ /* on >=i686, gcc compiles this into a single "imull" instruction */
+ int64_t r = (int64_t)a * b;
+ /* round the result before truncating - improves accuracy */
+ return (r + 0x80000000) >> 32;
+}
+
+/* Integer version of the cosine modulated Pseudo QMF */
+
+void qmf_init(void)
+{
+ int i;
+ int32_t c[17], s[17];
+ s[0] = 0; /* sin(index * PI / 64) * 0x7fffffff */
+ c[0] = 0x7fffffff; /* cos(index * PI / 64) * 0x7fffffff */
+
+ for (i = 1; i <= 16; i++) {
+ s[i] = 2 * (mul32(c[i-1], 105372028) + mul32(s[i-1], 2144896908));
+ c[i] = 2 * (mul32(c[i-1], 2144896908) - mul32(s[i-1], 105372028));
+ }
+
+ for (i = 0; i < 16; i++) {
+ cos_table[i] = c[i] >> 3; /* so that the output doesn't overflow */
+ cos_table[i+16] = s[16-i] >> 3;
+ cos_table[i+32] = -s[i] >> 3;
+ cos_table[i+48] = -c[16-i] >> 3;
+ cos_table[i+64] = -c[i] >> 3;
+ cos_table[i+80] = -s[16-i] >> 3;
+ cos_table[i+96] = s[i] >> 3;
+ cos_table[i+112] = c[16-i] >> 3;
+ }
+}
+
+static int32_t band_delta_factor(int band, int sample_num)
+{
+ int index = band * (2 * sample_num + 1);
+ if (band == 0)
+ return 0x07ffffff;
+ else
+ return cos_table[index & 127];
+}
+
+static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32], int channel)
+{
+ int band, i, j, k;
+ int32_t resp;
+ int32_t accum[DCA_SUBBANDS_32];
+
+ /* Place new samples into the history buffer */
+ for (i = 0; i < DCA_SUBBANDS_32; i++)
+ c->history[channel][c->start[channel] + i] = in[i];
+ c->start[channel] += DCA_SUBBANDS_32;
+ if (c->start[channel] == 512)
+ c->start[channel] = 0;
+
+ /* Calculate the dot product of the signal with the (possibly inverted)
+ reference decoder's response to this vector:
+ (0.0, 0.0, ..., 0.0, -1.0, 1.0, 0.0, ..., 0.0)
+ so that -1.0 cancels 1.0 from the previous step */
+
+ memset(accum,0,sizeof(int32_t));
+
+ for (k = 48, j = 0, i = c->start[channel]; i < 512; k++, j++, i++)
+ accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
+ for (i = 0; i < c->start[channel]; k++, j++, i++)
+ accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
+
+ resp = 0;
+ /* TODO: implement FFT instead of this naive calculation */
+ for (band = 0; band < DCA_SUBBANDS_32; band++) {
+ for (j = 0; j < 32; j++)
+ resp += mul32(accum[j], band_delta_factor(band, j));
+
+ out[band] = (band & 2) ? (-resp) : resp;
+ }
+}
+
+static void put_frame_header(DCAContext *c)
+{
+ /* SYNC */
+ put_bits(&c->pb, 16, 0x7ffe);
+ put_bits(&c->pb, 16, 0x8001);
+
+ /* Frame type: normal */
+ put_bits(&c->pb, 1, 1);
+
+ /* Deficit sample count: none */
+ put_bits(&c->pb, 5, 31);
+
+ /* CRC is not present */
+ put_bits(&c->pb, 1, 0);
+
+ /* Number of PCM sample blocks: 64
+ (larger values are unusable with 1:1 compression due to high bitrate
+ and frame size limitation) */
+ put_bits(&c->pb, 7, 63);
+
+ /* Primary frame byte size: 7168 */
+ put_bits(&c->pb, 14, FIXED_FRAME_SIZE);
+
+ /* Audio channel arrangement: L + R (stereo) */
+ put_bits(&c->pb, 6, 2);
+
+ /* Core audio sampling frequency: 44100 Hz */
+ put_bits(&c->pb, 4, 8);
+
+ /* Transmission bit rate: 1411.2 kbps */
+ put_bits(&c->pb, 5, 0x16);
+
+ /* Embedded down mix: disabled */
+ put_bits(&c->pb, 1, 0);
+
+ /* Embedded dynamic range flag: not present */
+ put_bits(&c->pb, 1, 0);
+
+ /* Embedded time stamp flag: not present */
+ put_bits(&c->pb, 1, 0);
+
+ /* Auxiliary data flag: not present */
+ put_bits(&c->pb, 1, 0);
+
+ /* HDCD source: no */
+ put_bits(&c->pb, 1, 0);
+
+ /* Extension audio ID: N/A */
+ put_bits(&c->pb, 3, 0);
+
+ /* Extended audio data: not present */
+ put_bits(&c->pb, 1, 0);
+
+ /* Audio sync word insertion flag: after each sub-frame */
+ put_bits(&c->pb, 1, 0);
+
+ /* Low frequency effects flag: not present */
+ put_bits(&c->pb, 2, 0);
+
+ /* Predictor history switch flag: on */
+ put_bits(&c->pb, 1, 1);
+
+ /* No CRC */
+ /* Multirate interpolator switch: non-perfect reconstruction */
+ put_bits(&c->pb, 1, 0);
+
+ /* Encoder software revision: 7 */
+ put_bits(&c->pb, 4, 7);
+
+ /* Copy history: 0 */
+ put_bits(&c->pb, 2, 0);
+
+ /* Source PCM resolution: 16 bits, not DTS ES */
+ put_bits(&c->pb, 3, 0);
+
+ /* Front sum/difference coding: no */
+ put_bits(&c->pb, 1, 0);
+
+ /* Surrounds sum/difference coding: no */
+ put_bits(&c->pb, 1, 0);
+
+ /* Dialog normalization: 0 dB */
+ put_bits(&c->pb, 4, 0);
+}
+
+static void put_primary_audio_header(DCAContext *c)
+{
+ /* Number of subframes: 2 */
+ put_bits(&c->pb, 4, 1);
+
+ /* Number of primary audio channels: 2 */
+ put_bits(&c->pb, 3, 1);
+
+ /* Subband activity count: 27 + 27 */
+ put_bits(&c->pb, 5, 25);
+ put_bits(&c->pb, 5, 25);
+
+ /* High frequency VQ start subband: 27, 27 */
+ put_bits(&c->pb, 5, 26);
+ put_bits(&c->pb, 5, 26);
+
+ /* Joint intensity coding index: 0, 0 */
+ put_bits(&c->pb, 3, 0);
+ put_bits(&c->pb, 3, 0);
+
+ /* Transient mode codebook: A4, A4 (arbitrary) */
+ put_bits(&c->pb, 2, 0);
+ put_bits(&c->pb, 2, 0);
+
+ /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
+ put_bits(&c->pb, 3, 6);
+ put_bits(&c->pb, 3, 6);
+
+ /* Bit allocation quantizer select: linear 5-bit */
+ put_bits(&c->pb, 3, 6);
+ put_bits(&c->pb, 3, 6);
+
+ /* Quantization index codebook select: dummy data
+ to avoid transmission of scale factor adjustment */
+ put_bits(&c->pb, 1, 1); put_bits(&c->pb, 1, 1);
+ put_bits(&c->pb, 2, 3); put_bits(&c->pb, 2, 3);
+ put_bits(&c->pb, 2, 3); put_bits(&c->pb, 2, 3);
+ put_bits(&c->pb, 2, 3); put_bits(&c->pb, 2, 3);
+ put_bits(&c->pb, 2, 3); put_bits(&c->pb, 2, 3);
+ put_bits(&c->pb, 3, 7); put_bits(&c->pb, 3, 7);
+ put_bits(&c->pb, 3, 7); put_bits(&c->pb, 3, 7);
+ put_bits(&c->pb, 3, 7); put_bits(&c->pb, 3, 7);
+ put_bits(&c->pb, 3, 7); put_bits(&c->pb, 3, 7);
+ put_bits(&c->pb, 3, 7); put_bits(&c->pb, 3, 7);
+
+ /* Scale factor adjustment index: not transmitted */
+}
+
+/* TODO: don't hardcode 16-bit quantization */
+static uint32_t quantize(int32_t d)
+{
+ d = d >> 16;
+ return d & 0xffff;
+}
+
+
+static void put_subframe(DCAContext *c, int32_t subband_data[32][2][32])
+{
+ int i, sub, ss, ch;
+
+ /* Subsubframes count: 4 */
+ put_bits(&c->pb, 2, 3);
+
+ /* Partial subsubframe sample count: dummy */
+ put_bits(&c->pb, 3, 0);
+
+ /* Prediction mode: no ADPCM, in each channel and subband */
+ for (ch = 0; ch < 2; ch++)
+ for (sub = 0; sub < 27; sub++)
+ put_bits(&c->pb, 1, 0);
+
+ /* Prediction VQ addres: not transmitted */
+ /* Bit allocation index: 19 = "16 bits", for each channel and subband */
+ for (ch = 0; ch < 2; ch++)
+ for (sub = 0; sub < 27; sub++)
+ put_bits(&c->pb, 5, 19);
+
+ /* Transition mode: none for each channel and subband */
+ for (ch = 0; ch < 2; ch++)
+ for (sub = 0; sub < 27; sub++)
+ put_bits(&c->pb, 1, 0); /* according to Huffman codebook A4 */
+
+ /* Scale factors: the same for each channel and subband,
+ encoded according to Table D.1.2 */
+ for (ch = 0; ch < 2; ch++)
+ for (sub = 0; sub < 27; sub++)
+ put_bits(&c->pb, 7, 110);
+
+ /* Joint subband scale factor codebook select: not transmitted */
+ /* Scale factors for joint subband coding: not transmitted */
+ /* Stereo down-mix coefficients: not transmitted */
+ /* Dynamic range coefficient: not transmitted */
+ /* Stde information CRC check word: not transmitted */
+ /* VQ encoded high frequency subbands: not transmitted */
+ /* LFE data: none */
+ /* Audio data: 4 subsubframes */
+
+ for (ss = 0; ss < 4 ; ss++)
+ for (ch = 0; ch < 2; ch++)
+ for (sub = 0; sub < 27; sub++)
+ for (i = 0; i < 8; i++)
+ put_bits(&c->pb, 16, quantize(subband_data[ss * 8 + i][ch][sub]));
+ /* DSYNC */
+ align_put_bits(&c->pb);
+ put_bits(&c->pb, 0xffff, 16);
+}
+
+void put_frame(DCAContext *c, int32_t subband_data[64][2][32], uint8_t *frame)
+{
+ int channel;
+ init_put_bits(&c->pb, frame, DCA_MAX_FRAME_SIZE);
+
+ put_frame_header(c);
+ put_primary_audio_header(c);
+ for (channel=0 ; channel<2; channel++)
+ put_subframe(c, &subband_data[32 * channel]);
+
+ flush_put_bits(&c->pb);
+}
+
+static int DCA_encode_frame(AVCodecContext *avctx,
+ uint8_t *frame, int buf_size, void *data)
+{
+ int i,k,channel;
+ DCAContext *c = avctx->priv_data;
+// int16_t *samples = data;
+
+// if (buf_size < MAX_CHANNELS*2048*sizeof(int16_t))
+// return -1;
+
+ // We always get 2048 16bit samples per call so save then until next call
+
+ if (c->fill_samples) {
+ memcpy(c->st_samples, data, 2048*sizeof(int16_t));
+ c->fill_samples = 0;
+ return 0;
+ }
+ memcpy(&c->st_samples[2048], data, 2048*sizeof(int16_t));
+ c->fill_samples = 0;
+
+ for (i = 0; i < 64; i ++) /* i is the decimated sample number */
+ for (channel=0; channel<2 ; channel++) {
+ /* Get 32 PCM samples */
+ for (k = 0; k < 32; k++) { /* k is the sample number in a 32-sample block */
+ c->pcm[k] = c->st_samples[4 * (32*i+k) + 2 * channel] << 16;
+ }
+ /* Put subband samples into the proper place */
+ qmf_decompose(c, c->pcm, &c->subband[i][channel][0], channel);
+ }
+
+ put_frame(c, c->subband, frame);
+
+ return 4000;
+}
+
+static int DCA_encode_init(AVCodecContext *avctx) {
+ DCAContext *c = avctx->priv_data;
+
+ if(avctx->channels != 2 || avctx->sample_rate != 44100) {
+ av_log(avctx, AV_LOG_ERROR, "Only 44.1 kHz stereo is supported at the moment!\n");
+ return -1;
+ }
+
+ // Make sure that the first call to encode is a fill call
+ c->fill_samples = 1;
+
+ qmf_init();
+ return 0;
+}
+
+AVCodec dca_encoder = {
+ "dca",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_DTS,
+ sizeof(DCAContext),
+ DCA_encode_init,
+ DCA_encode_frame,
+ NULL,
+ NULL,
+};
Added: dcaenc/dcaenc.h
==============================================================================
--- (empty file)
+++ dcaenc/dcaenc.h Thu Mar 20 14:40:49 2008
@@ -0,0 +1,539 @@
+/*
+ * DCA encoder tables
+ * Copyright (C) 2008 Alexander E. Patrakov
+ *
+ * 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
+ */
+
+/* This is a scaled version of the response of the reference decoder to
+ this vector of subband samples: ( 1.0 0.0 0.0 ... 0.0 )
+ */
+
+static const int32_t UnQMF[512] = {
+7,
+4,
+-961,
+-2844,
+-8024,
+-18978,
+-32081,
+-15635,
+-16582,
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+5,
+7
+};
Added: dcaenc/ffmpeg.patch
==============================================================================
--- (empty file)
+++ dcaenc/ffmpeg.patch Thu Mar 20 14:40:49 2008
@@ -0,0 +1,63 @@
+Index: libavcodec/Makefile
+===================================================================
+--- libavcodec/Makefile (revision 12509)
++++ libavcodec/Makefile (working copy)
+@@ -53,6 +53,7 @@
+ OBJS-$(CONFIG_CSCD_DECODER) += cscd.o
+ OBJS-$(CONFIG_CYUV_DECODER) += cyuv.o
+ OBJS-$(CONFIG_DCA_DECODER) += dca.o
++OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o
+ OBJS-$(CONFIG_DNXHD_DECODER) += dnxhddec.o dnxhddata.o
+ OBJS-$(CONFIG_DNXHD_ENCODER) += dnxhdenc.o dnxhddata.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12data.o mpegvideo.o
+ OBJS-$(CONFIG_DSICINAUDIO_DECODER) += dsicinav.o
+Index: libavcodec/allcodecs.c
+===================================================================
+--- libavcodec/allcodecs.c (revision 12509)
++++ libavcodec/allcodecs.c (working copy)
+@@ -179,7 +179,7 @@
+ REGISTER_DECODER (APE, ape);
+ REGISTER_DECODER (ATRAC3, atrac3);
+ REGISTER_DECODER (COOK, cook);
+- REGISTER_DECODER (DCA, dca);
++ REGISTER_ENCDEC (DCA, dca);
+ REGISTER_DECODER (DSICINAUDIO, dsicinaudio);
+ REGISTER_ENCDEC (FLAC, flac);
+ REGISTER_DECODER (IMC, imc);
+Index: libavformat/raw.c
+===================================================================
+--- libavformat/raw.c (revision 12509)
++++ libavformat/raw.c (working copy)
+@@ -513,6 +513,20 @@
+ raw_write_packet,
+ .flags= AVFMT_NOTIMESTAMPS,
+ };
++
++AVOutputFormat dts_muxer = {
++ "dts",
++ "raw dts",
++ "audio/x-dca",
++ "dts",
++ 0,
++ CODEC_ID_DTS,
++ 0,
++ NULL,
++ raw_write_packet,
++ .flags= AVFMT_NOTIMESTAMPS,
++};
++
+ #endif //CONFIG_MUXERS
+
+ AVInputFormat dts_demuxer = {
+Index: libavformat/allformats.c
+===================================================================
+--- libavformat/allformats.c (revision 12509)
++++ libavformat/allformats.c (working copy)
+@@ -69,7 +69,7 @@
+ REGISTER_MUXER (CRC, crc);
+ REGISTER_DEMUXER (DAUD, daud);
+ REGISTER_DEMUXER (DSICIN, dsicin);
+- REGISTER_DEMUXER (DTS, dts);
++ REGISTER_MUXDEMUX (DTS, dts);
+ REGISTER_MUXDEMUX (DV, dv);
+ REGISTER_DEMUXER (DXA, dxa);
+ REGISTER_DEMUXER (EA, ea);
Added: dcaenc/readme
==============================================================================
--- (empty file)
+++ dcaenc/readme Thu Mar 20 14:40:49 2008
@@ -0,0 +1,5 @@
+This source is based on code from here:
+
+http://roundup.mplayerhq.hu/roundup/ffmpeg/file153/edts-20080207.tar.gz
+
+
