[FFmpeg-cvslog] r14733 - in trunk/libavcodec: Makefile mlp.c mlp.h mlp_parser.c mlpdec.c
ramiro
subversion
Wed Aug 13 20:47:03 CEST 2008
Author: ramiro
Date: Wed Aug 13 20:47:03 2008
New Revision: 14733
Log:
mlp: Split common code from parser and decoder to be used by encoder.
Added:
trunk/libavcodec/mlp.c
- copied, changed from r14728, /trunk/libavcodec/mlpdec.c
trunk/libavcodec/mlp.h
- copied, changed from r14728, /trunk/libavcodec/mlpdec.c
Modified:
trunk/libavcodec/Makefile
trunk/libavcodec/mlp_parser.c
trunk/libavcodec/mlpdec.c
Modified: trunk/libavcodec/Makefile
==============================================================================
--- trunk/libavcodec/Makefile (original)
+++ trunk/libavcodec/Makefile Wed Aug 13 20:47:03 2008
@@ -109,7 +109,7 @@ OBJS-$(CONFIG_MIMIC_DECODER) +
OBJS-$(CONFIG_MJPEG_DECODER) += mjpegdec.o mjpeg.o
OBJS-$(CONFIG_MJPEG_ENCODER) += mjpegenc.o mjpeg.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12data.o mpegvideo.o
OBJS-$(CONFIG_MJPEGB_DECODER) += mjpegbdec.o mjpegdec.o mjpeg.o
-OBJS-$(CONFIG_MLP_DECODER) += mlpdec.o
+OBJS-$(CONFIG_MLP_DECODER) += mlp.o mlpdec.o
OBJS-$(CONFIG_MMVIDEO_DECODER) += mmvideo.o
OBJS-$(CONFIG_MOTIONPIXELS_DECODER) += motionpixels.o
OBJS-$(CONFIG_MP2_DECODER) += mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
@@ -348,7 +348,7 @@ OBJS-$(CONFIG_H261_PARSER) +
OBJS-$(CONFIG_H263_PARSER) += h263_parser.o
OBJS-$(CONFIG_H264_PARSER) += h264_parser.o
OBJS-$(CONFIG_MJPEG_PARSER) += mjpeg_parser.o
-OBJS-$(CONFIG_MLP_PARSER) += mlp_parser.o
+OBJS-$(CONFIG_MLP_PARSER) += mlp.o mlp_parser.o
OBJS-$(CONFIG_MPEG4VIDEO_PARSER) += mpeg4video_parser.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEGAUDIO_PARSER) += mpegaudio_parser.o mpegaudiodecheader.o mpegaudiodata.o
OBJS-$(CONFIG_MPEGVIDEO_PARSER) += mpegvideo_parser.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
Copied: trunk/libavcodec/mlp.c (from r14728, /trunk/libavcodec/mlpdec.c)
==============================================================================
--- /trunk/libavcodec/mlpdec.c (original)
+++ trunk/libavcodec/mlp.c Wed Aug 13 20:47:03 2008
@@ -1,5 +1,5 @@
/*
- * MLP decoder
+ * MLP codec common code
* Copyright (c) 2007-2008 Ian Caulfield
*
* This file is part of FFmpeg.
@@ -19,180 +19,12 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-/**
- * @file mlpdec.c
- * MLP decoder
- */
-
#include <stdint.h>
-#include "avcodec.h"
-#include "libavutil/intreadwrite.h"
-#include "bitstream.h"
#include "libavutil/crc.h"
-#include "parser.h"
-#include "mlp_parser.h"
-
-/** Maximum number of channels that can be decoded. */
-#define MAX_CHANNELS 16
-
-/** Maximum number of matrices used in decoding; most streams have one matrix
- * per output channel, but some rematrix a channel (usually 0) more than once.
- */
-
-#define MAX_MATRICES 15
-
-/** Maximum number of substreams that can be decoded. This could also be set
- * higher, but I haven't seen any examples with more than two. */
-#define MAX_SUBSTREAMS 2
-
-/** maximum sample frequency seen in files */
-#define MAX_SAMPLERATE 192000
-
-/** maximum number of audio samples within one access unit */
-#define MAX_BLOCKSIZE (40 * (MAX_SAMPLERATE / 48000))
-/** next power of two greater than MAX_BLOCKSIZE */
-#define MAX_BLOCKSIZE_POW2 (64 * (MAX_SAMPLERATE / 48000))
-
-/** number of allowed filters */
-#define NUM_FILTERS 2
-
-/** The maximum number of taps in either the IIR or FIR filter;
- * I believe MLP actually specifies the maximum order for IIR filters as four,
- * and that the sum of the orders of both filters must be <= 8. */
-#define MAX_FILTER_ORDER 8
-
-/** number of bits used for VLC lookup - longest Huffman code is 9 */
-#define VLC_BITS 9
-
-
-static const char* sample_message =
- "Please file a bug report following the instructions at "
- "http://ffmpeg.mplayerhq.hu/bugreports.html and include "
- "a sample of this file.";
-
-typedef struct SubStream {
- //! Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
- uint8_t restart_seen;
-
- //@{
- /** restart header data */
- //! The type of noise to be used in the rematrix stage.
- uint16_t noise_type;
-
- //! The index of the first channel coded in this substream.
- uint8_t min_channel;
- //! The index of the last channel coded in this substream.
- uint8_t max_channel;
- //! The number of channels input into the rematrix stage.
- uint8_t max_matrix_channel;
-
- //! The left shift applied to random noise in 0x31ea substreams.
- uint8_t noise_shift;
- //! The current seed value for the pseudorandom noise generator(s).
- uint32_t noisegen_seed;
-
- //! Set if the substream contains extra info to check the size of VLC blocks.
- uint8_t data_check_present;
-
- //! Bitmask of which parameter sets are conveyed in a decoding parameter block.
- uint8_t param_presence_flags;
-#define PARAM_BLOCKSIZE (1 << 7)
-#define PARAM_MATRIX (1 << 6)
-#define PARAM_OUTSHIFT (1 << 5)
-#define PARAM_QUANTSTEP (1 << 4)
-#define PARAM_FIR (1 << 3)
-#define PARAM_IIR (1 << 2)
-#define PARAM_HUFFOFFSET (1 << 1)
- //@}
-
- //@{
- /** matrix data */
-
- //! Number of matrices to be applied.
- uint8_t num_primitive_matrices;
-
- //! matrix output channel
- uint8_t matrix_out_ch[MAX_MATRICES];
-
- //! Whether the LSBs of the matrix output are encoded in the bitstream.
- uint8_t lsb_bypass[MAX_MATRICES];
- //! Matrix coefficients, stored as 2.14 fixed point.
- int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS+2];
- //! Left shift to apply to noise values in 0x31eb substreams.
- uint8_t matrix_noise_shift[MAX_MATRICES];
- //@}
-
- //! Left shift to apply to Huffman-decoded residuals.
- uint8_t quant_step_size[MAX_CHANNELS];
-
- //! number of PCM samples in current audio block
- uint16_t blocksize;
- //! Number of PCM samples decoded so far in this frame.
- uint16_t blockpos;
-
- //! Left shift to apply to decoded PCM values to get final 24-bit output.
- int8_t output_shift[MAX_CHANNELS];
-
- //! Running XOR of all output samples.
- int32_t lossless_check_data;
-
-} SubStream;
-
-#define FIR 0
-#define IIR 1
-
-/** filter data */
-typedef struct {
- uint8_t order; ///< number of taps in filter
- uint8_t shift; ///< Right shift to apply to output of filter.
-
- int32_t coeff[MAX_FILTER_ORDER];
- int32_t state[MAX_FILTER_ORDER];
-} FilterParams;
-
-/** sample data coding information */
-typedef struct {
- FilterParams filter_params[NUM_FILTERS];
-
- int16_t huff_offset; ///< Offset to apply to residual values.
- int32_t sign_huff_offset; ///< sign/rounding-corrected version of huff_offset
- uint8_t codebook; ///< Which VLC codebook to use to read residuals.
- uint8_t huff_lsbs; ///< Size of residual suffix not encoded using VLC.
-} ChannelParams;
-
-typedef struct MLPDecodeContext {
- AVCodecContext *avctx;
-
- //! Set if a valid major sync block has been read. Otherwise no decoding is possible.
- uint8_t params_valid;
-
- //! Number of substreams contained within this stream.
- uint8_t num_substreams;
-
- //! Index of the last substream to decode - further substreams are skipped.
- uint8_t max_decoded_substream;
-
- //! number of PCM samples contained in each frame
- int access_unit_size;
- //! next power of two above the number of samples in each frame
- int access_unit_size_pow2;
-
- SubStream substream[MAX_SUBSTREAMS];
-
- ChannelParams channel_params[MAX_CHANNELS];
-
- int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
- int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
- int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2];
-} MLPDecodeContext;
-
-/** Tables defining the Huffman codes.
- * There are three entropy coding methods used in MLP (four if you count
- * "none" as a method). These use the same sequences for codes starting with
- * 00 or 01, but have different codes starting with 1. */
+#include "mlp.h"
-static const uint8_t huffman_tables[3][18][2] = {
+const uint8_t ff_mlp_huffman_tables[3][18][2] = {
{ /* Huffman table 0, -7 - +10 */
{0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
{0x04, 3}, {0x05, 3}, {0x06, 3}, {0x07, 3},
@@ -208,27 +40,26 @@ static const uint8_t huffman_tables[3][1
}
};
-static VLC huff_vlc[3];
-
static int crc_init = 0;
static AVCRC crc_63[1024];
static AVCRC crc_1D[1024];
-/** Initialize static data, constant between all invocations of the codec. */
+static int crc_init_2D = 0;
+static AVCRC crc_2D[1024];
-static av_cold void init_static()
+int av_cold ff_mlp_init_crc2D(AVCodecParserContext *s)
{
- INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
- &huffman_tables[0][0][1], 2, 1,
- &huffman_tables[0][0][0], 2, 1, 512);
- INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
- &huffman_tables[1][0][1], 2, 1,
- &huffman_tables[1][0][0], 2, 1, 512);
- INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
- &huffman_tables[2][0][1], 2, 1,
- &huffman_tables[2][0][0], 2, 1, 512);
+ if (!crc_init_2D) {
+ av_crc_init(crc_2D, 0, 16, 0x002D, sizeof(crc_2D));
+ crc_init_2D = 1;
+ }
+ return 0;
+}
+
+void av_cold ff_mlp_init_crc()
+{
if (!crc_init) {
av_crc_init(crc_63, 0, 8, 0x63, sizeof(crc_63));
av_crc_init(crc_1D, 0, 8, 0x1D, sizeof(crc_1D));
@@ -236,23 +67,23 @@ static av_cold void init_static()
}
}
+uint16_t ff_mlp_checksum16(const uint8_t *buf, unsigned int buf_size)
+{
+ uint16_t crc;
-/** MLP uses checksums that seem to be based on the standard CRC algorithm, but
- * are not (in implementation terms, the table lookup and XOR are reversed).
- * We can implement this behavior using a standard av_crc on all but the
- * last element, then XOR that with the last element. */
+ crc = av_crc(crc_2D, 0, buf, buf_size - 2);
+ crc ^= AV_RL16(buf + buf_size - 2);
+ return crc;
+}
-static uint8_t mlp_checksum8(const uint8_t *buf, unsigned int buf_size)
+uint8_t ff_mlp_checksum8(const uint8_t *buf, unsigned int buf_size)
{
uint8_t checksum = av_crc(crc_63, 0x3c, buf, buf_size - 1); // crc_63[0xa2] == 0x3c
checksum ^= buf[buf_size-1];
return checksum;
}
-/** Calculate an 8-bit checksum over a restart header -- a non-multiple-of-8
- * number of bits, starting two bits into the first byte of buf. */
-
-static uint8_t mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size)
+uint8_t ff_mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size)
{
int i;
int num_bytes = (bit_size + 2) / 8;
@@ -271,714 +102,7 @@ static uint8_t mlp_restart_checksum(cons
return crc;
}
-static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
- unsigned int substr, unsigned int ch)
-{
- ChannelParams *cp = &m->channel_params[ch];
- SubStream *s = &m->substream[substr];
- int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
- int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
- int32_t sign_huff_offset = cp->huff_offset;
-
- if (cp->codebook > 0)
- sign_huff_offset -= 7 << lsb_bits;
-
- if (sign_shift >= 0)
- sign_huff_offset -= 1 << sign_shift;
-
- return sign_huff_offset;
-}
-
-/** Read a sample, consisting of either, both or neither of entropy-coded MSBs
- * and plain LSBs. */
-
-static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
- unsigned int substr, unsigned int pos)
-{
- SubStream *s = &m->substream[substr];
- unsigned int mat, channel;
-
- for (mat = 0; mat < s->num_primitive_matrices; mat++)
- if (s->lsb_bypass[mat])
- m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
-
- for (channel = s->min_channel; channel <= s->max_channel; channel++) {
- ChannelParams *cp = &m->channel_params[channel];
- int codebook = cp->codebook;
- int quant_step_size = s->quant_step_size[channel];
- int lsb_bits = cp->huff_lsbs - quant_step_size;
- int result = 0;
-
- if (codebook > 0)
- result = get_vlc2(gbp, huff_vlc[codebook-1].table,
- VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
-
- if (result < 0)
- return -1;
-
- if (lsb_bits > 0)
- result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
-
- result += cp->sign_huff_offset;
- result <<= quant_step_size;
-
- m->sample_buffer[pos + s->blockpos][channel] = result;
- }
-
- return 0;
-}
-
-static av_cold int mlp_decode_init(AVCodecContext *avctx)
-{
- MLPDecodeContext *m = avctx->priv_data;
- int substr;
-
- init_static();
- m->avctx = avctx;
- for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
- m->substream[substr].lossless_check_data = 0xffffffff;
- avctx->sample_fmt = SAMPLE_FMT_S16;
- return 0;
-}
-
-/** Read a major sync info header - contains high level information about
- * the stream - sample rate, channel arrangement etc. Most of this
- * information is not actually necessary for decoding, only for playback.
- */
-
-static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
-{
- MLPHeaderInfo mh;
- int substr;
-
- if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
- return -1;
-
- if (mh.group1_bits == 0) {
- av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
- return -1;
- }
- if (mh.group2_bits > mh.group1_bits) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Channel group 2 cannot have more bits per sample than group 1.\n");
- return -1;
- }
-
- if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Channel groups with differing sample rates are not currently supported.\n");
- return -1;
- }
-
- if (mh.group1_samplerate == 0) {
- av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
- return -1;
- }
- if (mh.group1_samplerate > MAX_SAMPLERATE) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Sampling rate %d is greater than the supported maximum (%d).\n",
- mh.group1_samplerate, MAX_SAMPLERATE);
- return -1;
- }
- if (mh.access_unit_size > MAX_BLOCKSIZE) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Block size %d is greater than the supported maximum (%d).\n",
- mh.access_unit_size, MAX_BLOCKSIZE);
- return -1;
- }
- if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Block size pow2 %d is greater than the supported maximum (%d).\n",
- mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
- return -1;
- }
-
- if (mh.num_substreams == 0)
- return -1;
- if (mh.num_substreams > MAX_SUBSTREAMS) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Number of substreams %d is larger than the maximum supported "
- "by the decoder. %s\n", mh.num_substreams, sample_message);
- return -1;
- }
-
- m->access_unit_size = mh.access_unit_size;
- m->access_unit_size_pow2 = mh.access_unit_size_pow2;
-
- m->num_substreams = mh.num_substreams;
- m->max_decoded_substream = m->num_substreams - 1;
-
- m->avctx->sample_rate = mh.group1_samplerate;
- m->avctx->frame_size = mh.access_unit_size;
-
-#ifdef CONFIG_AUDIO_NONSHORT
- m->avctx->bits_per_sample = mh.group1_bits;
- if (mh.group1_bits > 16) {
- m->avctx->sample_fmt = SAMPLE_FMT_S32;
- }
-#endif
-
- m->params_valid = 1;
- for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
- m->substream[substr].restart_seen = 0;
-
- return 0;
-}
-
-/** Read a restart header from a block in a substream. This contains parameters
- * required to decode the audio that do not change very often. Generally
- * (always) present only in blocks following a major sync. */
-
-static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
- const uint8_t *buf, unsigned int substr)
-{
- SubStream *s = &m->substream[substr];
- unsigned int ch;
- int sync_word, tmp;
- uint8_t checksum;
- uint8_t lossless_check;
- int start_count = get_bits_count(gbp);
-
- sync_word = get_bits(gbp, 13);
-
- if (sync_word != 0x31ea >> 1) {
- av_log(m->avctx, AV_LOG_ERROR,
- "restart header sync incorrect (got 0x%04x)\n", sync_word);
- return -1;
- }
- s->noise_type = get_bits1(gbp);
-
- skip_bits(gbp, 16); /* Output timestamp */
-
- s->min_channel = get_bits(gbp, 4);
- s->max_channel = get_bits(gbp, 4);
- s->max_matrix_channel = get_bits(gbp, 4);
-
- if (s->min_channel > s->max_channel) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Substream min channel cannot be greater than max channel.\n");
- return -1;
- }
-
- if (m->avctx->request_channels > 0
- && s->max_channel + 1 >= m->avctx->request_channels
- && substr < m->max_decoded_substream) {
- av_log(m->avctx, AV_LOG_INFO,
- "Extracting %d channel downmix from substream %d. "
- "Further substreams will be skipped.\n",
- s->max_channel + 1, substr);
- m->max_decoded_substream = substr;
- }
-
- s->noise_shift = get_bits(gbp, 4);
- s->noisegen_seed = get_bits(gbp, 23);
-
- skip_bits(gbp, 19);
-
- s->data_check_present = get_bits1(gbp);
- lossless_check = get_bits(gbp, 8);
- if (substr == m->max_decoded_substream
- && s->lossless_check_data != 0xffffffff) {
- tmp = s->lossless_check_data;
- tmp ^= tmp >> 16;
- tmp ^= tmp >> 8;
- tmp &= 0xff;
- if (tmp != lossless_check)
- av_log(m->avctx, AV_LOG_WARNING,
- "Lossless check failed - expected %02x, calculated %02x.\n",
- lossless_check, tmp);
- else
- dprintf(m->avctx, "Lossless check passed for substream %d (%x).\n",
- substr, tmp);
- }
-
- skip_bits(gbp, 16);
-
- for (ch = 0; ch <= s->max_matrix_channel; ch++) {
- int ch_assign = get_bits(gbp, 6);
- dprintf(m->avctx, "ch_assign[%d][%d] = %d\n", substr, ch,
- ch_assign);
- if (ch_assign != ch) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Non-1:1 channel assignments are used in this stream. %s\n",
- sample_message);
- return -1;
- }
- }
-
- checksum = mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
-
- if (checksum != get_bits(gbp, 8))
- av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
-
- /* Set default decoding parameters. */
- s->param_presence_flags = 0xff;
- s->num_primitive_matrices = 0;
- s->blocksize = 8;
- s->lossless_check_data = 0;
-
- memset(s->output_shift , 0, sizeof(s->output_shift ));
- memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
-
- for (ch = s->min_channel; ch <= s->max_channel; ch++) {
- ChannelParams *cp = &m->channel_params[ch];
- cp->filter_params[FIR].order = 0;
- cp->filter_params[IIR].order = 0;
- cp->filter_params[FIR].shift = 0;
- cp->filter_params[IIR].shift = 0;
-
- /* Default audio coding is 24-bit raw PCM. */
- cp->huff_offset = 0;
- cp->sign_huff_offset = (-1) << 23;
- cp->codebook = 0;
- cp->huff_lsbs = 24;
- }
-
- if (substr == m->max_decoded_substream) {
- m->avctx->channels = s->max_channel + 1;
- }
-
- return 0;
-}
-
-/** Read parameters for one of the prediction filters. */
-
-static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
- unsigned int channel, unsigned int filter)
-{
- FilterParams *fp = &m->channel_params[channel].filter_params[filter];
- const char fchar = filter ? 'I' : 'F';
- int i, order;
-
- // Filter is 0 for FIR, 1 for IIR.
- assert(filter < 2);
-
- order = get_bits(gbp, 4);
- if (order > MAX_FILTER_ORDER) {
- av_log(m->avctx, AV_LOG_ERROR,
- "%cIR filter order %d is greater than maximum %d.\n",
- fchar, order, MAX_FILTER_ORDER);
- return -1;
- }
- fp->order = order;
-
- if (order > 0) {
- int coeff_bits, coeff_shift;
-
- fp->shift = get_bits(gbp, 4);
-
- coeff_bits = get_bits(gbp, 5);
- coeff_shift = get_bits(gbp, 3);
- if (coeff_bits < 1 || coeff_bits > 16) {
- av_log(m->avctx, AV_LOG_ERROR,
- "%cIR filter coeff_bits must be between 1 and 16.\n",
- fchar);
- return -1;
- }
- if (coeff_bits + coeff_shift > 16) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
- fchar);
- return -1;
- }
-
- for (i = 0; i < order; i++)
- fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
-
- if (get_bits1(gbp)) {
- int state_bits, state_shift;
-
- if (filter == FIR) {
- av_log(m->avctx, AV_LOG_ERROR,
- "FIR filter has state data specified.\n");
- return -1;
- }
-
- state_bits = get_bits(gbp, 4);
- state_shift = get_bits(gbp, 4);
-
- /* TODO: Check validity of state data. */
-
- for (i = 0; i < order; i++)
- fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
- }
- }
-
- return 0;
-}
-
-/** Read decoding parameters that change more often than those in the restart
- * header. */
-
-static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
- unsigned int substr)
-{
- SubStream *s = &m->substream[substr];
- unsigned int mat, ch;
-
- if (get_bits1(gbp))
- s->param_presence_flags = get_bits(gbp, 8);
-
- if (s->param_presence_flags & PARAM_BLOCKSIZE)
- if (get_bits1(gbp)) {
- s->blocksize = get_bits(gbp, 9);
- if (s->blocksize > MAX_BLOCKSIZE) {
- av_log(m->avctx, AV_LOG_ERROR, "block size too large\n");
- s->blocksize = 0;
- return -1;
- }
- }
-
- if (s->param_presence_flags & PARAM_MATRIX)
- if (get_bits1(gbp)) {
- s->num_primitive_matrices = get_bits(gbp, 4);
-
- for (mat = 0; mat < s->num_primitive_matrices; mat++) {
- int frac_bits, max_chan;
- s->matrix_out_ch[mat] = get_bits(gbp, 4);
- frac_bits = get_bits(gbp, 4);
- s->lsb_bypass [mat] = get_bits1(gbp);
-
- if (s->matrix_out_ch[mat] > s->max_channel) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Invalid channel %d specified as output from matrix.\n",
- s->matrix_out_ch[mat]);
- return -1;
- }
- if (frac_bits > 14) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Too many fractional bits specified.\n");
- return -1;
- }
-
- max_chan = s->max_matrix_channel;
- if (!s->noise_type)
- max_chan+=2;
-
- for (ch = 0; ch <= max_chan; ch++) {
- int coeff_val = 0;
- if (get_bits1(gbp))
- coeff_val = get_sbits(gbp, frac_bits + 2);
-
- s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
- }
-
- if (s->noise_type)
- s->matrix_noise_shift[mat] = get_bits(gbp, 4);
- else
- s->matrix_noise_shift[mat] = 0;
- }
- }
-
- if (s->param_presence_flags & PARAM_OUTSHIFT)
- if (get_bits1(gbp))
- for (ch = 0; ch <= s->max_matrix_channel; ch++) {
- s->output_shift[ch] = get_bits(gbp, 4);
- dprintf(m->avctx, "output shift[%d] = %d\n",
- ch, s->output_shift[ch]);
- /* TODO: validate */
- }
-
- if (s->param_presence_flags & PARAM_QUANTSTEP)
- if (get_bits1(gbp))
- for (ch = 0; ch <= s->max_channel; ch++) {
- ChannelParams *cp = &m->channel_params[ch];
-
- s->quant_step_size[ch] = get_bits(gbp, 4);
- /* TODO: validate */
-
- cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
- }
-
- for (ch = s->min_channel; ch <= s->max_channel; ch++)
- if (get_bits1(gbp)) {
- ChannelParams *cp = &m->channel_params[ch];
- FilterParams *fir = &cp->filter_params[FIR];
- FilterParams *iir = &cp->filter_params[IIR];
-
- if (s->param_presence_flags & PARAM_FIR)
- if (get_bits1(gbp))
- if (read_filter_params(m, gbp, ch, FIR) < 0)
- return -1;
-
- if (s->param_presence_flags & PARAM_IIR)
- if (get_bits1(gbp))
- if (read_filter_params(m, gbp, ch, IIR) < 0)
- return -1;
-
- if (fir->order && iir->order &&
- fir->shift != iir->shift) {
- av_log(m->avctx, AV_LOG_ERROR,
- "FIR and IIR filters must use the same precision.\n");
- return -1;
- }
- /* The FIR and IIR filters must have the same precision.
- * To simplify the filtering code, only the precision of the
- * FIR filter is considered. If only the IIR filter is employed,
- * the FIR filter precision is set to that of the IIR filter, so
- * that the filtering code can use it. */
- if (!fir->order && iir->order)
- fir->shift = iir->shift;
-
- if (s->param_presence_flags & PARAM_HUFFOFFSET)
- if (get_bits1(gbp))
- cp->huff_offset = get_sbits(gbp, 15);
-
- cp->codebook = get_bits(gbp, 2);
- cp->huff_lsbs = get_bits(gbp, 5);
-
- cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
-
- /* TODO: validate */
- }
-
- return 0;
-}
-
-#define MSB_MASK(bits) (-1u << bits)
-
-/** Generate PCM samples using the prediction filters and residual values
- * read from the data stream, and update the filter state. */
-
-static void filter_channel(MLPDecodeContext *m, unsigned int substr,
- unsigned int channel)
-{
- SubStream *s = &m->substream[substr];
- int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER];
- FilterParams *fp[NUM_FILTERS] = { &m->channel_params[channel].filter_params[FIR],
- &m->channel_params[channel].filter_params[IIR], };
- unsigned int filter_shift = fp[FIR]->shift;
- int32_t mask = MSB_MASK(s->quant_step_size[channel]);
- int index = MAX_BLOCKSIZE;
- int j, i;
-
- for (j = 0; j < NUM_FILTERS; j++) {
- memcpy(&filter_state_buffer[j][MAX_BLOCKSIZE], &fp[j]->state[0],
- MAX_FILTER_ORDER * sizeof(int32_t));
- }
-
- for (i = 0; i < s->blocksize; i++) {
- int32_t residual = m->sample_buffer[i + s->blockpos][channel];
- unsigned int order;
- int64_t accum = 0;
- int32_t result;
-
- /* TODO: Move this code to DSPContext? */
-
- for (j = 0; j < NUM_FILTERS; j++)
- for (order = 0; order < fp[j]->order; order++)
- accum += (int64_t)filter_state_buffer[j][index + order] *
- fp[j]->coeff[order];
-
- accum = accum >> filter_shift;
- result = (accum + residual) & mask;
-
- --index;
-
- filter_state_buffer[FIR][index] = result;
- filter_state_buffer[IIR][index] = result - accum;
-
- m->sample_buffer[i + s->blockpos][channel] = result;
- }
-
- for (j = 0; j < NUM_FILTERS; j++) {
- memcpy(&fp[j]->state[0], &filter_state_buffer[j][index],
- MAX_FILTER_ORDER * sizeof(int32_t));
- }
-}
-
-/** Read a block of PCM residual data (or actual if no filtering active). */
-
-static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
- unsigned int substr)
-{
- SubStream *s = &m->substream[substr];
- unsigned int i, ch, expected_stream_pos = 0;
-
- if (s->data_check_present) {
- expected_stream_pos = get_bits_count(gbp);
- expected_stream_pos += get_bits(gbp, 16);
- av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
- "we have not tested yet. %s\n", sample_message);
- }
-
- if (s->blockpos + s->blocksize > m->access_unit_size) {
- av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
- return -1;
- }
-
- memset(&m->bypassed_lsbs[s->blockpos][0], 0,
- s->blocksize * sizeof(m->bypassed_lsbs[0]));
-
- for (i = 0; i < s->blocksize; i++) {
- if (read_huff_channels(m, gbp, substr, i) < 0)
- return -1;
- }
-
- for (ch = s->min_channel; ch <= s->max_channel; ch++) {
- filter_channel(m, substr, ch);
- }
-
- s->blockpos += s->blocksize;
-
- if (s->data_check_present) {
- if (get_bits_count(gbp) != expected_stream_pos)
- av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
- skip_bits(gbp, 8);
- }
-
- return 0;
-}
-
-/** Data table used for TrueHD noise generation function. */
-
-static const int8_t noise_table[256] = {
- 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
- 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
- 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
- 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
- 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
- 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
- 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
- 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
- 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
- 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
- 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
- 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
- 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
- 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
- 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
- -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
-};
-
-/** Noise generation functions.
- * I'm not sure what these are for - they seem to be some kind of pseudorandom
- * sequence generators, used to generate noise data which is used when the
- * channels are rematrixed. I'm not sure if they provide a practical benefit
- * to compression, or just obfuscate the decoder. Are they for some kind of
- * dithering? */
-
-/** Generate two channels of noise, used in the matrix when
- * restart sync word == 0x31ea. */
-
-static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
-{
- SubStream *s = &m->substream[substr];
- unsigned int i;
- uint32_t seed = s->noisegen_seed;
- unsigned int maxchan = s->max_matrix_channel;
-
- for (i = 0; i < s->blockpos; i++) {
- uint16_t seed_shr7 = seed >> 7;
- m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
- m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
-
- seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
- }
-
- s->noisegen_seed = seed;
-}
-
-/** Generate a block of noise, used when restart sync word == 0x31eb. */
-
-static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
-{
- SubStream *s = &m->substream[substr];
- unsigned int i;
- uint32_t seed = s->noisegen_seed;
-
- for (i = 0; i < m->access_unit_size_pow2; i++) {
- uint8_t seed_shr15 = seed >> 15;
- m->noise_buffer[i] = noise_table[seed_shr15];
- seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
- }
-
- s->noisegen_seed = seed;
-}
-
-
-/** Apply the channel matrices in turn to reconstruct the original audio
- * samples. */
-
-static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
-{
- SubStream *s = &m->substream[substr];
- unsigned int mat, src_ch, i;
- unsigned int maxchan;
-
- maxchan = s->max_matrix_channel;
- if (!s->noise_type) {
- generate_2_noise_channels(m, substr);
- maxchan += 2;
- } else {
- fill_noise_buffer(m, substr);
- }
-
- for (mat = 0; mat < s->num_primitive_matrices; mat++) {
- int matrix_noise_shift = s->matrix_noise_shift[mat];
- unsigned int dest_ch = s->matrix_out_ch[mat];
- int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
-
- /* TODO: DSPContext? */
-
- for (i = 0; i < s->blockpos; i++) {
- int64_t accum = 0;
- for (src_ch = 0; src_ch <= maxchan; src_ch++) {
- accum += (int64_t)m->sample_buffer[i][src_ch]
- * s->matrix_coeff[mat][src_ch];
- }
- if (matrix_noise_shift) {
- uint32_t index = s->num_primitive_matrices - mat;
- index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
- accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
- }
- m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
- + m->bypassed_lsbs[i][mat];
- }
- }
-}
-
-/** Write the audio data into the output buffer. */
-
-static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
- uint8_t *data, unsigned int *data_size, int is32)
-{
- SubStream *s = &m->substream[substr];
- unsigned int i, ch = 0;
- int32_t *data_32 = (int32_t*) data;
- int16_t *data_16 = (int16_t*) data;
-
- if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
- return -1;
-
- for (i = 0; i < s->blockpos; i++) {
- for (ch = 0; ch <= s->max_channel; ch++) {
- int32_t sample = m->sample_buffer[i][ch] << s->output_shift[ch];
- s->lossless_check_data ^= (sample & 0xffffff) << ch;
- if (is32) *data_32++ = sample << 8;
- else *data_16++ = sample >> 8;
- }
- }
-
- *data_size = i * ch * (is32 ? 4 : 2);
-
- return 0;
-}
-
-static int output_data(MLPDecodeContext *m, unsigned int substr,
- uint8_t *data, unsigned int *data_size)
-{
- if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
- return output_data_internal(m, substr, data, data_size, 1);
- else
- return output_data_internal(m, substr, data, data_size, 0);
-}
-
-
-/** XOR together all the bytes of a buffer.
- * Does this belong in dspcontext? */
-
-static uint8_t calculate_parity(const uint8_t *buf, unsigned int buf_size)
+uint8_t ff_mlp_calculate_parity(const uint8_t *buf, unsigned int buf_size)
{
uint32_t scratch = 0;
const uint8_t *buf_end = buf + buf_size;
@@ -994,198 +118,3 @@ static uint8_t calculate_parity(const ui
return scratch;
}
-
-/** Read an access unit from the stream.
- * Returns < 0 on error, 0 if not enough data is present in the input stream
- * otherwise returns the number of bytes consumed. */
-
-static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
- const uint8_t *buf, int buf_size)
-{
- MLPDecodeContext *m = avctx->priv_data;
- GetBitContext gb;
- unsigned int length, substr;
- unsigned int substream_start;
- unsigned int header_size = 4;
- unsigned int substr_header_size = 0;
- uint8_t substream_parity_present[MAX_SUBSTREAMS];
- uint16_t substream_data_len[MAX_SUBSTREAMS];
- uint8_t parity_bits;
-
- if (buf_size < 4)
- return 0;
-
- length = (AV_RB16(buf) & 0xfff) * 2;
-
- if (length > buf_size)
- return -1;
-
- init_get_bits(&gb, (buf + 4), (length - 4) * 8);
-
- if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
- dprintf(m->avctx, "Found major sync.\n");
- if (read_major_sync(m, &gb) < 0)
- goto error;
- header_size += 28;
- }
-
- if (!m->params_valid) {
- av_log(m->avctx, AV_LOG_WARNING,
- "Stream parameters not seen; skipping frame.\n");
- *data_size = 0;
- return length;
- }
-
- substream_start = 0;
-
- for (substr = 0; substr < m->num_substreams; substr++) {
- int extraword_present, checkdata_present, end;
-
- extraword_present = get_bits1(&gb);
- skip_bits1(&gb);
- checkdata_present = get_bits1(&gb);
- skip_bits1(&gb);
-
- end = get_bits(&gb, 12) * 2;
-
- substr_header_size += 2;
-
- if (extraword_present) {
- skip_bits(&gb, 16);
- substr_header_size += 2;
- }
-
- if (end + header_size + substr_header_size > length) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Indicated length of substream %d data goes off end of "
- "packet.\n", substr);
- end = length - header_size - substr_header_size;
- }
-
- if (end < substream_start) {
- av_log(avctx, AV_LOG_ERROR,
- "Indicated end offset of substream %d data "
- "is smaller than calculated start offset.\n",
- substr);
- goto error;
- }
-
- if (substr > m->max_decoded_substream)
- continue;
-
- substream_parity_present[substr] = checkdata_present;
- substream_data_len[substr] = end - substream_start;
- substream_start = end;
- }
-
- parity_bits = calculate_parity(buf, 4);
- parity_bits ^= calculate_parity(buf + header_size, substr_header_size);
-
- if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
- av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
- goto error;
- }
-
- buf += header_size + substr_header_size;
-
- for (substr = 0; substr <= m->max_decoded_substream; substr++) {
- SubStream *s = &m->substream[substr];
- init_get_bits(&gb, buf, substream_data_len[substr] * 8);
-
- s->blockpos = 0;
- do {
- if (get_bits1(&gb)) {
- if (get_bits1(&gb)) {
- /* A restart header should be present. */
- if (read_restart_header(m, &gb, buf, substr) < 0)
- goto next_substr;
- s->restart_seen = 1;
- }
-
- if (!s->restart_seen) {
- av_log(m->avctx, AV_LOG_ERROR,
- "No restart header present in substream %d.\n",
- substr);
- goto next_substr;
- }
-
- if (read_decoding_params(m, &gb, substr) < 0)
- goto next_substr;
- }
-
- if (!s->restart_seen) {
- av_log(m->avctx, AV_LOG_ERROR,
- "No restart header present in substream %d.\n",
- substr);
- goto next_substr;
- }
-
- if (read_block_data(m, &gb, substr) < 0)
- return -1;
-
- } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)
- && get_bits1(&gb) == 0);
-
- skip_bits(&gb, (-get_bits_count(&gb)) & 15);
- if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32 &&
- (show_bits_long(&gb, 32) == 0xd234d234 ||
- show_bits_long(&gb, 20) == 0xd234e)) {
- skip_bits(&gb, 18);
- if (substr == m->max_decoded_substream)
- av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
-
- if (get_bits1(&gb)) {
- int shorten_by = get_bits(&gb, 13);
- shorten_by = FFMIN(shorten_by, s->blockpos);
- s->blockpos -= shorten_by;
- } else
- skip_bits(&gb, 13);
- }
- if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 16 &&
- substream_parity_present[substr]) {
- uint8_t parity, checksum;
-
- parity = calculate_parity(buf, substream_data_len[substr] - 2);
- if ((parity ^ get_bits(&gb, 8)) != 0xa9)
- av_log(m->avctx, AV_LOG_ERROR,
- "Substream %d parity check failed.\n", substr);
-
- checksum = mlp_checksum8(buf, substream_data_len[substr] - 2);
- if (checksum != get_bits(&gb, 8))
- av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n",
- substr);
- }
- if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
- av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n",
- substr);
- return -1;
- }
-
-next_substr:
- buf += substream_data_len[substr];
- }
-
- rematrix_channels(m, m->max_decoded_substream);
-
- if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
- return -1;
-
- return length;
-
-error:
- m->params_valid = 0;
- return -1;
-}
-
-AVCodec mlp_decoder = {
- "mlp",
- CODEC_TYPE_AUDIO,
- CODEC_ID_MLP,
- sizeof(MLPDecodeContext),
- mlp_decode_init,
- NULL,
- NULL,
- read_access_unit,
- .long_name = NULL_IF_CONFIG_SMALL("Meridian Lossless Packing"),
-};
-
Copied: trunk/libavcodec/mlp.h (from r14728, /trunk/libavcodec/mlpdec.c)
==============================================================================
--- /trunk/libavcodec/mlpdec.c (original)
+++ trunk/libavcodec/mlp.h Wed Aug 13 20:47:03 2008
@@ -1,5 +1,5 @@
/*
- * MLP decoder
+ * MLP codec common header file
* Copyright (c) 2007-2008 Ian Caulfield
*
* This file is part of FFmpeg.
@@ -19,19 +19,12 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-/**
- * @file mlpdec.c
- * MLP decoder
- */
+#ifndef FFMPEG_MLP_H
+#define FFMPEG_MLP_H
#include <stdint.h>
#include "avcodec.h"
-#include "libavutil/intreadwrite.h"
-#include "bitstream.h"
-#include "libavutil/crc.h"
-#include "parser.h"
-#include "mlp_parser.h"
/** Maximum number of channels that can be decoded. */
#define MAX_CHANNELS 16
@@ -62,83 +55,6 @@
* and that the sum of the orders of both filters must be <= 8. */
#define MAX_FILTER_ORDER 8
-/** number of bits used for VLC lookup - longest Huffman code is 9 */
-#define VLC_BITS 9
-
-
-static const char* sample_message =
- "Please file a bug report following the instructions at "
- "http://ffmpeg.mplayerhq.hu/bugreports.html and include "
- "a sample of this file.";
-
-typedef struct SubStream {
- //! Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
- uint8_t restart_seen;
-
- //@{
- /** restart header data */
- //! The type of noise to be used in the rematrix stage.
- uint16_t noise_type;
-
- //! The index of the first channel coded in this substream.
- uint8_t min_channel;
- //! The index of the last channel coded in this substream.
- uint8_t max_channel;
- //! The number of channels input into the rematrix stage.
- uint8_t max_matrix_channel;
-
- //! The left shift applied to random noise in 0x31ea substreams.
- uint8_t noise_shift;
- //! The current seed value for the pseudorandom noise generator(s).
- uint32_t noisegen_seed;
-
- //! Set if the substream contains extra info to check the size of VLC blocks.
- uint8_t data_check_present;
-
- //! Bitmask of which parameter sets are conveyed in a decoding parameter block.
- uint8_t param_presence_flags;
-#define PARAM_BLOCKSIZE (1 << 7)
-#define PARAM_MATRIX (1 << 6)
-#define PARAM_OUTSHIFT (1 << 5)
-#define PARAM_QUANTSTEP (1 << 4)
-#define PARAM_FIR (1 << 3)
-#define PARAM_IIR (1 << 2)
-#define PARAM_HUFFOFFSET (1 << 1)
- //@}
-
- //@{
- /** matrix data */
-
- //! Number of matrices to be applied.
- uint8_t num_primitive_matrices;
-
- //! matrix output channel
- uint8_t matrix_out_ch[MAX_MATRICES];
-
- //! Whether the LSBs of the matrix output are encoded in the bitstream.
- uint8_t lsb_bypass[MAX_MATRICES];
- //! Matrix coefficients, stored as 2.14 fixed point.
- int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS+2];
- //! Left shift to apply to noise values in 0x31eb substreams.
- uint8_t matrix_noise_shift[MAX_MATRICES];
- //@}
-
- //! Left shift to apply to Huffman-decoded residuals.
- uint8_t quant_step_size[MAX_CHANNELS];
-
- //! number of PCM samples in current audio block
- uint16_t blocksize;
- //! Number of PCM samples decoded so far in this frame.
- uint16_t blockpos;
-
- //! Left shift to apply to decoded PCM values to get final 24-bit output.
- int8_t output_shift[MAX_CHANNELS];
-
- //! Running XOR of all output samples.
- int32_t lossless_check_data;
-
-} SubStream;
-
#define FIR 0
#define IIR 1
@@ -161,1031 +77,33 @@ typedef struct {
uint8_t huff_lsbs; ///< Size of residual suffix not encoded using VLC.
} ChannelParams;
-typedef struct MLPDecodeContext {
- AVCodecContext *avctx;
-
- //! Set if a valid major sync block has been read. Otherwise no decoding is possible.
- uint8_t params_valid;
-
- //! Number of substreams contained within this stream.
- uint8_t num_substreams;
-
- //! Index of the last substream to decode - further substreams are skipped.
- uint8_t max_decoded_substream;
-
- //! number of PCM samples contained in each frame
- int access_unit_size;
- //! next power of two above the number of samples in each frame
- int access_unit_size_pow2;
-
- SubStream substream[MAX_SUBSTREAMS];
-
- ChannelParams channel_params[MAX_CHANNELS];
-
- int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
- int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
- int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2];
-} MLPDecodeContext;
-
/** Tables defining the Huffman codes.
* There are three entropy coding methods used in MLP (four if you count
* "none" as a method). These use the same sequences for codes starting with
* 00 or 01, but have different codes starting with 1. */
-static const uint8_t huffman_tables[3][18][2] = {
- { /* Huffman table 0, -7 - +10 */
- {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
- {0x04, 3}, {0x05, 3}, {0x06, 3}, {0x07, 3},
- {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
- }, { /* Huffman table 1, -7 - +8 */
- {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
- {0x02, 2}, {0x03, 2},
- {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
- }, { /* Huffman table 2, -7 - +7 */
- {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
- {0x01, 1},
- {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
- }
-};
-
-static VLC huff_vlc[3];
-
-static int crc_init = 0;
-static AVCRC crc_63[1024];
-static AVCRC crc_1D[1024];
-
-
-/** Initialize static data, constant between all invocations of the codec. */
-
-static av_cold void init_static()
-{
- INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
- &huffman_tables[0][0][1], 2, 1,
- &huffman_tables[0][0][0], 2, 1, 512);
- INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
- &huffman_tables[1][0][1], 2, 1,
- &huffman_tables[1][0][0], 2, 1, 512);
- INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
- &huffman_tables[2][0][1], 2, 1,
- &huffman_tables[2][0][0], 2, 1, 512);
-
- if (!crc_init) {
- av_crc_init(crc_63, 0, 8, 0x63, sizeof(crc_63));
- av_crc_init(crc_1D, 0, 8, 0x1D, sizeof(crc_1D));
- crc_init = 1;
- }
-}
-
+extern const uint8_t ff_mlp_huffman_tables[3][18][2];
/** MLP uses checksums that seem to be based on the standard CRC algorithm, but
* are not (in implementation terms, the table lookup and XOR are reversed).
* We can implement this behavior using a standard av_crc on all but the
* last element, then XOR that with the last element. */
-static uint8_t mlp_checksum8(const uint8_t *buf, unsigned int buf_size)
-{
- uint8_t checksum = av_crc(crc_63, 0x3c, buf, buf_size - 1); // crc_63[0xa2] == 0x3c
- checksum ^= buf[buf_size-1];
- return checksum;
-}
+uint8_t ff_mlp_checksum8 (const uint8_t *buf, unsigned int buf_size);
+uint16_t ff_mlp_checksum16(const uint8_t *buf, unsigned int buf_size);
/** Calculate an 8-bit checksum over a restart header -- a non-multiple-of-8
* number of bits, starting two bits into the first byte of buf. */
-static uint8_t mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size)
-{
- int i;
- int num_bytes = (bit_size + 2) / 8;
-
- int crc = crc_1D[buf[0] & 0x3f];
- crc = av_crc(crc_1D, crc, buf + 1, num_bytes - 2);
- crc ^= buf[num_bytes - 1];
-
- for (i = 0; i < ((bit_size + 2) & 7); i++) {
- crc <<= 1;
- if (crc & 0x100)
- crc ^= 0x11D;
- crc ^= (buf[num_bytes] >> (7 - i)) & 1;
- }
-
- return crc;
-}
-
-static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
- unsigned int substr, unsigned int ch)
-{
- ChannelParams *cp = &m->channel_params[ch];
- SubStream *s = &m->substream[substr];
- int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
- int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
- int32_t sign_huff_offset = cp->huff_offset;
-
- if (cp->codebook > 0)
- sign_huff_offset -= 7 << lsb_bits;
-
- if (sign_shift >= 0)
- sign_huff_offset -= 1 << sign_shift;
-
- return sign_huff_offset;
-}
-
-/** Read a sample, consisting of either, both or neither of entropy-coded MSBs
- * and plain LSBs. */
-
-static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
- unsigned int substr, unsigned int pos)
-{
- SubStream *s = &m->substream[substr];
- unsigned int mat, channel;
-
- for (mat = 0; mat < s->num_primitive_matrices; mat++)
- if (s->lsb_bypass[mat])
- m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
-
- for (channel = s->min_channel; channel <= s->max_channel; channel++) {
- ChannelParams *cp = &m->channel_params[channel];
- int codebook = cp->codebook;
- int quant_step_size = s->quant_step_size[channel];
- int lsb_bits = cp->huff_lsbs - quant_step_size;
- int result = 0;
-
- if (codebook > 0)
- result = get_vlc2(gbp, huff_vlc[codebook-1].table,
- VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
-
- if (result < 0)
- return -1;
-
- if (lsb_bits > 0)
- result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
-
- result += cp->sign_huff_offset;
- result <<= quant_step_size;
-
- m->sample_buffer[pos + s->blockpos][channel] = result;
- }
-
- return 0;
-}
-
-static av_cold int mlp_decode_init(AVCodecContext *avctx)
-{
- MLPDecodeContext *m = avctx->priv_data;
- int substr;
-
- init_static();
- m->avctx = avctx;
- for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
- m->substream[substr].lossless_check_data = 0xffffffff;
- avctx->sample_fmt = SAMPLE_FMT_S16;
- return 0;
-}
-
-/** Read a major sync info header - contains high level information about
- * the stream - sample rate, channel arrangement etc. Most of this
- * information is not actually necessary for decoding, only for playback.
- */
-
-static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
-{
- MLPHeaderInfo mh;
- int substr;
-
- if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
- return -1;
-
- if (mh.group1_bits == 0) {
- av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
- return -1;
- }
- if (mh.group2_bits > mh.group1_bits) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Channel group 2 cannot have more bits per sample than group 1.\n");
- return -1;
- }
-
- if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Channel groups with differing sample rates are not currently supported.\n");
- return -1;
- }
-
- if (mh.group1_samplerate == 0) {
- av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
- return -1;
- }
- if (mh.group1_samplerate > MAX_SAMPLERATE) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Sampling rate %d is greater than the supported maximum (%d).\n",
- mh.group1_samplerate, MAX_SAMPLERATE);
- return -1;
- }
- if (mh.access_unit_size > MAX_BLOCKSIZE) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Block size %d is greater than the supported maximum (%d).\n",
- mh.access_unit_size, MAX_BLOCKSIZE);
- return -1;
- }
- if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Block size pow2 %d is greater than the supported maximum (%d).\n",
- mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
- return -1;
- }
-
- if (mh.num_substreams == 0)
- return -1;
- if (mh.num_substreams > MAX_SUBSTREAMS) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Number of substreams %d is larger than the maximum supported "
- "by the decoder. %s\n", mh.num_substreams, sample_message);
- return -1;
- }
-
- m->access_unit_size = mh.access_unit_size;
- m->access_unit_size_pow2 = mh.access_unit_size_pow2;
-
- m->num_substreams = mh.num_substreams;
- m->max_decoded_substream = m->num_substreams - 1;
-
- m->avctx->sample_rate = mh.group1_samplerate;
- m->avctx->frame_size = mh.access_unit_size;
-
-#ifdef CONFIG_AUDIO_NONSHORT
- m->avctx->bits_per_sample = mh.group1_bits;
- if (mh.group1_bits > 16) {
- m->avctx->sample_fmt = SAMPLE_FMT_S32;
- }
-#endif
-
- m->params_valid = 1;
- for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
- m->substream[substr].restart_seen = 0;
-
- return 0;
-}
-
-/** Read a restart header from a block in a substream. This contains parameters
- * required to decode the audio that do not change very often. Generally
- * (always) present only in blocks following a major sync. */
-
-static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
- const uint8_t *buf, unsigned int substr)
-{
- SubStream *s = &m->substream[substr];
- unsigned int ch;
- int sync_word, tmp;
- uint8_t checksum;
- uint8_t lossless_check;
- int start_count = get_bits_count(gbp);
-
- sync_word = get_bits(gbp, 13);
-
- if (sync_word != 0x31ea >> 1) {
- av_log(m->avctx, AV_LOG_ERROR,
- "restart header sync incorrect (got 0x%04x)\n", sync_word);
- return -1;
- }
- s->noise_type = get_bits1(gbp);
-
- skip_bits(gbp, 16); /* Output timestamp */
-
- s->min_channel = get_bits(gbp, 4);
- s->max_channel = get_bits(gbp, 4);
- s->max_matrix_channel = get_bits(gbp, 4);
-
- if (s->min_channel > s->max_channel) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Substream min channel cannot be greater than max channel.\n");
- return -1;
- }
-
- if (m->avctx->request_channels > 0
- && s->max_channel + 1 >= m->avctx->request_channels
- && substr < m->max_decoded_substream) {
- av_log(m->avctx, AV_LOG_INFO,
- "Extracting %d channel downmix from substream %d. "
- "Further substreams will be skipped.\n",
- s->max_channel + 1, substr);
- m->max_decoded_substream = substr;
- }
-
- s->noise_shift = get_bits(gbp, 4);
- s->noisegen_seed = get_bits(gbp, 23);
-
- skip_bits(gbp, 19);
-
- s->data_check_present = get_bits1(gbp);
- lossless_check = get_bits(gbp, 8);
- if (substr == m->max_decoded_substream
- && s->lossless_check_data != 0xffffffff) {
- tmp = s->lossless_check_data;
- tmp ^= tmp >> 16;
- tmp ^= tmp >> 8;
- tmp &= 0xff;
- if (tmp != lossless_check)
- av_log(m->avctx, AV_LOG_WARNING,
- "Lossless check failed - expected %02x, calculated %02x.\n",
- lossless_check, tmp);
- else
- dprintf(m->avctx, "Lossless check passed for substream %d (%x).\n",
- substr, tmp);
- }
-
- skip_bits(gbp, 16);
-
- for (ch = 0; ch <= s->max_matrix_channel; ch++) {
- int ch_assign = get_bits(gbp, 6);
- dprintf(m->avctx, "ch_assign[%d][%d] = %d\n", substr, ch,
- ch_assign);
- if (ch_assign != ch) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Non-1:1 channel assignments are used in this stream. %s\n",
- sample_message);
- return -1;
- }
- }
-
- checksum = mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
-
- if (checksum != get_bits(gbp, 8))
- av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
-
- /* Set default decoding parameters. */
- s->param_presence_flags = 0xff;
- s->num_primitive_matrices = 0;
- s->blocksize = 8;
- s->lossless_check_data = 0;
-
- memset(s->output_shift , 0, sizeof(s->output_shift ));
- memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
-
- for (ch = s->min_channel; ch <= s->max_channel; ch++) {
- ChannelParams *cp = &m->channel_params[ch];
- cp->filter_params[FIR].order = 0;
- cp->filter_params[IIR].order = 0;
- cp->filter_params[FIR].shift = 0;
- cp->filter_params[IIR].shift = 0;
-
- /* Default audio coding is 24-bit raw PCM. */
- cp->huff_offset = 0;
- cp->sign_huff_offset = (-1) << 23;
- cp->codebook = 0;
- cp->huff_lsbs = 24;
- }
-
- if (substr == m->max_decoded_substream) {
- m->avctx->channels = s->max_channel + 1;
- }
-
- return 0;
-}
-
-/** Read parameters for one of the prediction filters. */
-
-static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
- unsigned int channel, unsigned int filter)
-{
- FilterParams *fp = &m->channel_params[channel].filter_params[filter];
- const char fchar = filter ? 'I' : 'F';
- int i, order;
-
- // Filter is 0 for FIR, 1 for IIR.
- assert(filter < 2);
-
- order = get_bits(gbp, 4);
- if (order > MAX_FILTER_ORDER) {
- av_log(m->avctx, AV_LOG_ERROR,
- "%cIR filter order %d is greater than maximum %d.\n",
- fchar, order, MAX_FILTER_ORDER);
- return -1;
- }
- fp->order = order;
-
- if (order > 0) {
- int coeff_bits, coeff_shift;
-
- fp->shift = get_bits(gbp, 4);
-
- coeff_bits = get_bits(gbp, 5);
- coeff_shift = get_bits(gbp, 3);
- if (coeff_bits < 1 || coeff_bits > 16) {
- av_log(m->avctx, AV_LOG_ERROR,
- "%cIR filter coeff_bits must be between 1 and 16.\n",
- fchar);
- return -1;
- }
- if (coeff_bits + coeff_shift > 16) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
- fchar);
- return -1;
- }
-
- for (i = 0; i < order; i++)
- fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
-
- if (get_bits1(gbp)) {
- int state_bits, state_shift;
-
- if (filter == FIR) {
- av_log(m->avctx, AV_LOG_ERROR,
- "FIR filter has state data specified.\n");
- return -1;
- }
-
- state_bits = get_bits(gbp, 4);
- state_shift = get_bits(gbp, 4);
-
- /* TODO: Check validity of state data. */
-
- for (i = 0; i < order; i++)
- fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
- }
- }
-
- return 0;
-}
-
-/** Read decoding parameters that change more often than those in the restart
- * header. */
-
-static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
- unsigned int substr)
-{
- SubStream *s = &m->substream[substr];
- unsigned int mat, ch;
-
- if (get_bits1(gbp))
- s->param_presence_flags = get_bits(gbp, 8);
-
- if (s->param_presence_flags & PARAM_BLOCKSIZE)
- if (get_bits1(gbp)) {
- s->blocksize = get_bits(gbp, 9);
- if (s->blocksize > MAX_BLOCKSIZE) {
- av_log(m->avctx, AV_LOG_ERROR, "block size too large\n");
- s->blocksize = 0;
- return -1;
- }
- }
-
- if (s->param_presence_flags & PARAM_MATRIX)
- if (get_bits1(gbp)) {
- s->num_primitive_matrices = get_bits(gbp, 4);
-
- for (mat = 0; mat < s->num_primitive_matrices; mat++) {
- int frac_bits, max_chan;
- s->matrix_out_ch[mat] = get_bits(gbp, 4);
- frac_bits = get_bits(gbp, 4);
- s->lsb_bypass [mat] = get_bits1(gbp);
-
- if (s->matrix_out_ch[mat] > s->max_channel) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Invalid channel %d specified as output from matrix.\n",
- s->matrix_out_ch[mat]);
- return -1;
- }
- if (frac_bits > 14) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Too many fractional bits specified.\n");
- return -1;
- }
-
- max_chan = s->max_matrix_channel;
- if (!s->noise_type)
- max_chan+=2;
-
- for (ch = 0; ch <= max_chan; ch++) {
- int coeff_val = 0;
- if (get_bits1(gbp))
- coeff_val = get_sbits(gbp, frac_bits + 2);
-
- s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
- }
-
- if (s->noise_type)
- s->matrix_noise_shift[mat] = get_bits(gbp, 4);
- else
- s->matrix_noise_shift[mat] = 0;
- }
- }
-
- if (s->param_presence_flags & PARAM_OUTSHIFT)
- if (get_bits1(gbp))
- for (ch = 0; ch <= s->max_matrix_channel; ch++) {
- s->output_shift[ch] = get_bits(gbp, 4);
- dprintf(m->avctx, "output shift[%d] = %d\n",
- ch, s->output_shift[ch]);
- /* TODO: validate */
- }
-
- if (s->param_presence_flags & PARAM_QUANTSTEP)
- if (get_bits1(gbp))
- for (ch = 0; ch <= s->max_channel; ch++) {
- ChannelParams *cp = &m->channel_params[ch];
-
- s->quant_step_size[ch] = get_bits(gbp, 4);
- /* TODO: validate */
-
- cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
- }
-
- for (ch = s->min_channel; ch <= s->max_channel; ch++)
- if (get_bits1(gbp)) {
- ChannelParams *cp = &m->channel_params[ch];
- FilterParams *fir = &cp->filter_params[FIR];
- FilterParams *iir = &cp->filter_params[IIR];
-
- if (s->param_presence_flags & PARAM_FIR)
- if (get_bits1(gbp))
- if (read_filter_params(m, gbp, ch, FIR) < 0)
- return -1;
-
- if (s->param_presence_flags & PARAM_IIR)
- if (get_bits1(gbp))
- if (read_filter_params(m, gbp, ch, IIR) < 0)
- return -1;
-
- if (fir->order && iir->order &&
- fir->shift != iir->shift) {
- av_log(m->avctx, AV_LOG_ERROR,
- "FIR and IIR filters must use the same precision.\n");
- return -1;
- }
- /* The FIR and IIR filters must have the same precision.
- * To simplify the filtering code, only the precision of the
- * FIR filter is considered. If only the IIR filter is employed,
- * the FIR filter precision is set to that of the IIR filter, so
- * that the filtering code can use it. */
- if (!fir->order && iir->order)
- fir->shift = iir->shift;
-
- if (s->param_presence_flags & PARAM_HUFFOFFSET)
- if (get_bits1(gbp))
- cp->huff_offset = get_sbits(gbp, 15);
-
- cp->codebook = get_bits(gbp, 2);
- cp->huff_lsbs = get_bits(gbp, 5);
-
- cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
-
- /* TODO: validate */
- }
-
- return 0;
-}
-
-#define MSB_MASK(bits) (-1u << bits)
-
-/** Generate PCM samples using the prediction filters and residual values
- * read from the data stream, and update the filter state. */
-
-static void filter_channel(MLPDecodeContext *m, unsigned int substr,
- unsigned int channel)
-{
- SubStream *s = &m->substream[substr];
- int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER];
- FilterParams *fp[NUM_FILTERS] = { &m->channel_params[channel].filter_params[FIR],
- &m->channel_params[channel].filter_params[IIR], };
- unsigned int filter_shift = fp[FIR]->shift;
- int32_t mask = MSB_MASK(s->quant_step_size[channel]);
- int index = MAX_BLOCKSIZE;
- int j, i;
-
- for (j = 0; j < NUM_FILTERS; j++) {
- memcpy(&filter_state_buffer[j][MAX_BLOCKSIZE], &fp[j]->state[0],
- MAX_FILTER_ORDER * sizeof(int32_t));
- }
-
- for (i = 0; i < s->blocksize; i++) {
- int32_t residual = m->sample_buffer[i + s->blockpos][channel];
- unsigned int order;
- int64_t accum = 0;
- int32_t result;
-
- /* TODO: Move this code to DSPContext? */
-
- for (j = 0; j < NUM_FILTERS; j++)
- for (order = 0; order < fp[j]->order; order++)
- accum += (int64_t)filter_state_buffer[j][index + order] *
- fp[j]->coeff[order];
-
- accum = accum >> filter_shift;
- result = (accum + residual) & mask;
-
- --index;
-
- filter_state_buffer[FIR][index] = result;
- filter_state_buffer[IIR][index] = result - accum;
-
- m->sample_buffer[i + s->blockpos][channel] = result;
- }
-
- for (j = 0; j < NUM_FILTERS; j++) {
- memcpy(&fp[j]->state[0], &filter_state_buffer[j][index],
- MAX_FILTER_ORDER * sizeof(int32_t));
- }
-}
-
-/** Read a block of PCM residual data (or actual if no filtering active). */
-
-static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
- unsigned int substr)
-{
- SubStream *s = &m->substream[substr];
- unsigned int i, ch, expected_stream_pos = 0;
-
- if (s->data_check_present) {
- expected_stream_pos = get_bits_count(gbp);
- expected_stream_pos += get_bits(gbp, 16);
- av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
- "we have not tested yet. %s\n", sample_message);
- }
-
- if (s->blockpos + s->blocksize > m->access_unit_size) {
- av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
- return -1;
- }
-
- memset(&m->bypassed_lsbs[s->blockpos][0], 0,
- s->blocksize * sizeof(m->bypassed_lsbs[0]));
-
- for (i = 0; i < s->blocksize; i++) {
- if (read_huff_channels(m, gbp, substr, i) < 0)
- return -1;
- }
-
- for (ch = s->min_channel; ch <= s->max_channel; ch++) {
- filter_channel(m, substr, ch);
- }
-
- s->blockpos += s->blocksize;
-
- if (s->data_check_present) {
- if (get_bits_count(gbp) != expected_stream_pos)
- av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
- skip_bits(gbp, 8);
- }
-
- return 0;
-}
-
-/** Data table used for TrueHD noise generation function. */
-
-static const int8_t noise_table[256] = {
- 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
- 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
- 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
- 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
- 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
- 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
- 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
- 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
- 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
- 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
- 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
- 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
- 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
- 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
- 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
- -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
-};
-
-/** Noise generation functions.
- * I'm not sure what these are for - they seem to be some kind of pseudorandom
- * sequence generators, used to generate noise data which is used when the
- * channels are rematrixed. I'm not sure if they provide a practical benefit
- * to compression, or just obfuscate the decoder. Are they for some kind of
- * dithering? */
-
-/** Generate two channels of noise, used in the matrix when
- * restart sync word == 0x31ea. */
-
-static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
-{
- SubStream *s = &m->substream[substr];
- unsigned int i;
- uint32_t seed = s->noisegen_seed;
- unsigned int maxchan = s->max_matrix_channel;
-
- for (i = 0; i < s->blockpos; i++) {
- uint16_t seed_shr7 = seed >> 7;
- m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
- m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
-
- seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
- }
-
- s->noisegen_seed = seed;
-}
-
-/** Generate a block of noise, used when restart sync word == 0x31eb. */
-
-static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
-{
- SubStream *s = &m->substream[substr];
- unsigned int i;
- uint32_t seed = s->noisegen_seed;
-
- for (i = 0; i < m->access_unit_size_pow2; i++) {
- uint8_t seed_shr15 = seed >> 15;
- m->noise_buffer[i] = noise_table[seed_shr15];
- seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
- }
-
- s->noisegen_seed = seed;
-}
-
-
-/** Apply the channel matrices in turn to reconstruct the original audio
- * samples. */
-
-static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
-{
- SubStream *s = &m->substream[substr];
- unsigned int mat, src_ch, i;
- unsigned int maxchan;
-
- maxchan = s->max_matrix_channel;
- if (!s->noise_type) {
- generate_2_noise_channels(m, substr);
- maxchan += 2;
- } else {
- fill_noise_buffer(m, substr);
- }
-
- for (mat = 0; mat < s->num_primitive_matrices; mat++) {
- int matrix_noise_shift = s->matrix_noise_shift[mat];
- unsigned int dest_ch = s->matrix_out_ch[mat];
- int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
-
- /* TODO: DSPContext? */
-
- for (i = 0; i < s->blockpos; i++) {
- int64_t accum = 0;
- for (src_ch = 0; src_ch <= maxchan; src_ch++) {
- accum += (int64_t)m->sample_buffer[i][src_ch]
- * s->matrix_coeff[mat][src_ch];
- }
- if (matrix_noise_shift) {
- uint32_t index = s->num_primitive_matrices - mat;
- index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
- accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
- }
- m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
- + m->bypassed_lsbs[i][mat];
- }
- }
-}
-
-/** Write the audio data into the output buffer. */
-
-static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
- uint8_t *data, unsigned int *data_size, int is32)
-{
- SubStream *s = &m->substream[substr];
- unsigned int i, ch = 0;
- int32_t *data_32 = (int32_t*) data;
- int16_t *data_16 = (int16_t*) data;
-
- if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
- return -1;
-
- for (i = 0; i < s->blockpos; i++) {
- for (ch = 0; ch <= s->max_channel; ch++) {
- int32_t sample = m->sample_buffer[i][ch] << s->output_shift[ch];
- s->lossless_check_data ^= (sample & 0xffffff) << ch;
- if (is32) *data_32++ = sample << 8;
- else *data_16++ = sample >> 8;
- }
- }
-
- *data_size = i * ch * (is32 ? 4 : 2);
-
- return 0;
-}
-
-static int output_data(MLPDecodeContext *m, unsigned int substr,
- uint8_t *data, unsigned int *data_size)
-{
- if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
- return output_data_internal(m, substr, data, data_size, 1);
- else
- return output_data_internal(m, substr, data, data_size, 0);
-}
-
+uint8_t ff_mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size);
/** XOR together all the bytes of a buffer.
* Does this belong in dspcontext? */
-static uint8_t calculate_parity(const uint8_t *buf, unsigned int buf_size)
-{
- uint32_t scratch = 0;
- const uint8_t *buf_end = buf + buf_size;
-
- for (; buf < buf_end - 3; buf += 4)
- scratch ^= *((const uint32_t*)buf);
-
- scratch ^= scratch >> 16;
- scratch ^= scratch >> 8;
-
- for (; buf < buf_end; buf++)
- scratch ^= *buf;
-
- return scratch;
-}
-
-/** Read an access unit from the stream.
- * Returns < 0 on error, 0 if not enough data is present in the input stream
- * otherwise returns the number of bytes consumed. */
-
-static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
- const uint8_t *buf, int buf_size)
-{
- MLPDecodeContext *m = avctx->priv_data;
- GetBitContext gb;
- unsigned int length, substr;
- unsigned int substream_start;
- unsigned int header_size = 4;
- unsigned int substr_header_size = 0;
- uint8_t substream_parity_present[MAX_SUBSTREAMS];
- uint16_t substream_data_len[MAX_SUBSTREAMS];
- uint8_t parity_bits;
-
- if (buf_size < 4)
- return 0;
-
- length = (AV_RB16(buf) & 0xfff) * 2;
-
- if (length > buf_size)
- return -1;
-
- init_get_bits(&gb, (buf + 4), (length - 4) * 8);
-
- if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
- dprintf(m->avctx, "Found major sync.\n");
- if (read_major_sync(m, &gb) < 0)
- goto error;
- header_size += 28;
- }
-
- if (!m->params_valid) {
- av_log(m->avctx, AV_LOG_WARNING,
- "Stream parameters not seen; skipping frame.\n");
- *data_size = 0;
- return length;
- }
-
- substream_start = 0;
-
- for (substr = 0; substr < m->num_substreams; substr++) {
- int extraword_present, checkdata_present, end;
-
- extraword_present = get_bits1(&gb);
- skip_bits1(&gb);
- checkdata_present = get_bits1(&gb);
- skip_bits1(&gb);
-
- end = get_bits(&gb, 12) * 2;
-
- substr_header_size += 2;
-
- if (extraword_present) {
- skip_bits(&gb, 16);
- substr_header_size += 2;
- }
-
- if (end + header_size + substr_header_size > length) {
- av_log(m->avctx, AV_LOG_ERROR,
- "Indicated length of substream %d data goes off end of "
- "packet.\n", substr);
- end = length - header_size - substr_header_size;
- }
-
- if (end < substream_start) {
- av_log(avctx, AV_LOG_ERROR,
- "Indicated end offset of substream %d data "
- "is smaller than calculated start offset.\n",
- substr);
- goto error;
- }
-
- if (substr > m->max_decoded_substream)
- continue;
-
- substream_parity_present[substr] = checkdata_present;
- substream_data_len[substr] = end - substream_start;
- substream_start = end;
- }
-
- parity_bits = calculate_parity(buf, 4);
- parity_bits ^= calculate_parity(buf + header_size, substr_header_size);
-
- if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
- av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
- goto error;
- }
-
- buf += header_size + substr_header_size;
-
- for (substr = 0; substr <= m->max_decoded_substream; substr++) {
- SubStream *s = &m->substream[substr];
- init_get_bits(&gb, buf, substream_data_len[substr] * 8);
-
- s->blockpos = 0;
- do {
- if (get_bits1(&gb)) {
- if (get_bits1(&gb)) {
- /* A restart header should be present. */
- if (read_restart_header(m, &gb, buf, substr) < 0)
- goto next_substr;
- s->restart_seen = 1;
- }
-
- if (!s->restart_seen) {
- av_log(m->avctx, AV_LOG_ERROR,
- "No restart header present in substream %d.\n",
- substr);
- goto next_substr;
- }
-
- if (read_decoding_params(m, &gb, substr) < 0)
- goto next_substr;
- }
-
- if (!s->restart_seen) {
- av_log(m->avctx, AV_LOG_ERROR,
- "No restart header present in substream %d.\n",
- substr);
- goto next_substr;
- }
-
- if (read_block_data(m, &gb, substr) < 0)
- return -1;
-
- } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)
- && get_bits1(&gb) == 0);
-
- skip_bits(&gb, (-get_bits_count(&gb)) & 15);
- if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32 &&
- (show_bits_long(&gb, 32) == 0xd234d234 ||
- show_bits_long(&gb, 20) == 0xd234e)) {
- skip_bits(&gb, 18);
- if (substr == m->max_decoded_substream)
- av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
-
- if (get_bits1(&gb)) {
- int shorten_by = get_bits(&gb, 13);
- shorten_by = FFMIN(shorten_by, s->blockpos);
- s->blockpos -= shorten_by;
- } else
- skip_bits(&gb, 13);
- }
- if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 16 &&
- substream_parity_present[substr]) {
- uint8_t parity, checksum;
-
- parity = calculate_parity(buf, substream_data_len[substr] - 2);
- if ((parity ^ get_bits(&gb, 8)) != 0xa9)
- av_log(m->avctx, AV_LOG_ERROR,
- "Substream %d parity check failed.\n", substr);
-
- checksum = mlp_checksum8(buf, substream_data_len[substr] - 2);
- if (checksum != get_bits(&gb, 8))
- av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n",
- substr);
- }
- if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
- av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n",
- substr);
- return -1;
- }
-
-next_substr:
- buf += substream_data_len[substr];
- }
-
- rematrix_channels(m, m->max_decoded_substream);
-
- if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
- return -1;
-
- return length;
+uint8_t ff_mlp_calculate_parity(const uint8_t *buf, unsigned int buf_size);
-error:
- m->params_valid = 0;
- return -1;
-}
+int ff_mlp_init_crc2D(AVCodecParserContext *s);
-AVCodec mlp_decoder = {
- "mlp",
- CODEC_TYPE_AUDIO,
- CODEC_ID_MLP,
- sizeof(MLPDecodeContext),
- mlp_decode_init,
- NULL,
- NULL,
- read_access_unit,
- .long_name = NULL_IF_CONFIG_SMALL("Meridian Lossless Packing"),
-};
+void ff_mlp_init_crc();
+#endif /* FFMPEG_MLP_H */
Modified: trunk/libavcodec/mlp_parser.c
==============================================================================
--- trunk/libavcodec/mlp_parser.c (original)
+++ trunk/libavcodec/mlp_parser.c Wed Aug 13 20:47:03 2008
@@ -30,6 +30,7 @@
#include "bitstream.h"
#include "parser.h"
#include "mlp_parser.h"
+#include "mlp.h"
static const uint8_t mlp_quants[16] = {
16, 20, 24, 0, 0, 0, 0, 0,
@@ -64,34 +65,6 @@ static int truehd_channels(int chanmap)
return channels;
}
-static int crc_init = 0;
-static AVCRC crc_2D[1024];
-
-/** MLP uses checksums that seem to be based on the standard CRC algorithm, but
- * are not (in implementation terms, the table lookup and XOR are reversed).
- * We can implement this behavior using a standard av_crc on all but the
- * last element, then XOR that with the last element.
- */
-
-static uint16_t mlp_checksum16(const uint8_t *buf, unsigned int buf_size)
-{
- uint16_t crc;
-
- crc = av_crc(crc_2D, 0, buf, buf_size - 2);
- crc ^= AV_RL16(buf + buf_size - 2);
- return crc;
-}
-
-static int av_cold mlp_parse_init(AVCodecParserContext *s)
-{
- if (!crc_init) {
- av_crc_init(crc_2D, 0, 16, 0x002D, sizeof(crc_2D));
- crc_init = 1;
- }
-
- return 0;
-}
-
/** Read a major sync info header - contains high level information about
* the stream - sample rate, channel arrangement etc. Most of this
* information is not actually necessary for decoding, only for playback.
@@ -110,7 +83,7 @@ int ff_mlp_read_major_sync(void *log, ML
return -1;
}
- checksum = mlp_checksum16(gb->buffer, 26);
+ checksum = ff_mlp_checksum16(gb->buffer, 26);
if (checksum != AV_RL16(gb->buffer+26)) {
av_log(log, AV_LOG_ERROR, "major sync info header checksum error\n");
return -1;
@@ -310,7 +283,7 @@ lost_sync:
AVCodecParser mlp_parser = {
{ CODEC_ID_MLP },
sizeof(MLPParseContext),
- mlp_parse_init,
+ ff_mlp_init_crc2D,
mlp_parse,
NULL,
};
Modified: trunk/libavcodec/mlpdec.c
==============================================================================
--- trunk/libavcodec/mlpdec.c (original)
+++ trunk/libavcodec/mlpdec.c Wed Aug 13 20:47:03 2008
@@ -32,35 +32,7 @@
#include "libavutil/crc.h"
#include "parser.h"
#include "mlp_parser.h"
-
-/** Maximum number of channels that can be decoded. */
-#define MAX_CHANNELS 16
-
-/** Maximum number of matrices used in decoding; most streams have one matrix
- * per output channel, but some rematrix a channel (usually 0) more than once.
- */
-
-#define MAX_MATRICES 15
-
-/** Maximum number of substreams that can be decoded. This could also be set
- * higher, but I haven't seen any examples with more than two. */
-#define MAX_SUBSTREAMS 2
-
-/** maximum sample frequency seen in files */
-#define MAX_SAMPLERATE 192000
-
-/** maximum number of audio samples within one access unit */
-#define MAX_BLOCKSIZE (40 * (MAX_SAMPLERATE / 48000))
-/** next power of two greater than MAX_BLOCKSIZE */
-#define MAX_BLOCKSIZE_POW2 (64 * (MAX_SAMPLERATE / 48000))
-
-/** number of allowed filters */
-#define NUM_FILTERS 2
-
-/** The maximum number of taps in either the IIR or FIR filter;
- * I believe MLP actually specifies the maximum order for IIR filters as four,
- * and that the sum of the orders of both filters must be <= 8. */
-#define MAX_FILTER_ORDER 8
+#include "mlp.h"
/** number of bits used for VLC lookup - longest Huffman code is 9 */
#define VLC_BITS 9
@@ -139,28 +111,6 @@ typedef struct SubStream {
} SubStream;
-#define FIR 0
-#define IIR 1
-
-/** filter data */
-typedef struct {
- uint8_t order; ///< number of taps in filter
- uint8_t shift; ///< Right shift to apply to output of filter.
-
- int32_t coeff[MAX_FILTER_ORDER];
- int32_t state[MAX_FILTER_ORDER];
-} FilterParams;
-
-/** sample data coding information */
-typedef struct {
- FilterParams filter_params[NUM_FILTERS];
-
- int16_t huff_offset; ///< Offset to apply to residual values.
- int32_t sign_huff_offset; ///< sign/rounding-corrected version of huff_offset
- uint8_t codebook; ///< Which VLC codebook to use to read residuals.
- uint8_t huff_lsbs; ///< Size of residual suffix not encoded using VLC.
-} ChannelParams;
-
typedef struct MLPDecodeContext {
AVCodecContext *avctx;
@@ -187,88 +137,23 @@ typedef struct MLPDecodeContext {
int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2];
} MLPDecodeContext;
-/** Tables defining the Huffman codes.
- * There are three entropy coding methods used in MLP (four if you count
- * "none" as a method). These use the same sequences for codes starting with
- * 00 or 01, but have different codes starting with 1. */
-
-static const uint8_t huffman_tables[3][18][2] = {
- { /* Huffman table 0, -7 - +10 */
- {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
- {0x04, 3}, {0x05, 3}, {0x06, 3}, {0x07, 3},
- {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
- }, { /* Huffman table 1, -7 - +8 */
- {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
- {0x02, 2}, {0x03, 2},
- {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
- }, { /* Huffman table 2, -7 - +7 */
- {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
- {0x01, 1},
- {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
- }
-};
-
static VLC huff_vlc[3];
-static int crc_init = 0;
-static AVCRC crc_63[1024];
-static AVCRC crc_1D[1024];
-
-
/** Initialize static data, constant between all invocations of the codec. */
static av_cold void init_static()
{
INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
- &huffman_tables[0][0][1], 2, 1,
- &huffman_tables[0][0][0], 2, 1, 512);
+ &ff_mlp_huffman_tables[0][0][1], 2, 1,
+ &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
- &huffman_tables[1][0][1], 2, 1,
- &huffman_tables[1][0][0], 2, 1, 512);
+ &ff_mlp_huffman_tables[1][0][1], 2, 1,
+ &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
- &huffman_tables[2][0][1], 2, 1,
- &huffman_tables[2][0][0], 2, 1, 512);
-
- if (!crc_init) {
- av_crc_init(crc_63, 0, 8, 0x63, sizeof(crc_63));
- av_crc_init(crc_1D, 0, 8, 0x1D, sizeof(crc_1D));
- crc_init = 1;
- }
-}
-
-
-/** MLP uses checksums that seem to be based on the standard CRC algorithm, but
- * are not (in implementation terms, the table lookup and XOR are reversed).
- * We can implement this behavior using a standard av_crc on all but the
- * last element, then XOR that with the last element. */
-
-static uint8_t mlp_checksum8(const uint8_t *buf, unsigned int buf_size)
-{
- uint8_t checksum = av_crc(crc_63, 0x3c, buf, buf_size - 1); // crc_63[0xa2] == 0x3c
- checksum ^= buf[buf_size-1];
- return checksum;
-}
-
-/** Calculate an 8-bit checksum over a restart header -- a non-multiple-of-8
- * number of bits, starting two bits into the first byte of buf. */
-
-static uint8_t mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size)
-{
- int i;
- int num_bytes = (bit_size + 2) / 8;
-
- int crc = crc_1D[buf[0] & 0x3f];
- crc = av_crc(crc_1D, crc, buf + 1, num_bytes - 2);
- crc ^= buf[num_bytes - 1];
-
- for (i = 0; i < ((bit_size + 2) & 7); i++) {
- crc <<= 1;
- if (crc & 0x100)
- crc ^= 0x11D;
- crc ^= (buf[num_bytes] >> (7 - i)) & 1;
- }
+ &ff_mlp_huffman_tables[2][0][1], 2, 1,
+ &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
- return crc;
+ ff_mlp_init_crc();
}
static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
@@ -506,7 +391,7 @@ static int read_restart_header(MLPDecode
}
}
- checksum = mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
+ checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
if (checksum != get_bits(gbp, 8))
av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
@@ -975,26 +860,6 @@ static int output_data(MLPDecodeContext
}
-/** XOR together all the bytes of a buffer.
- * Does this belong in dspcontext? */
-
-static uint8_t calculate_parity(const uint8_t *buf, unsigned int buf_size)
-{
- uint32_t scratch = 0;
- const uint8_t *buf_end = buf + buf_size;
-
- for (; buf < buf_end - 3; buf += 4)
- scratch ^= *((const uint32_t*)buf);
-
- scratch ^= scratch >> 16;
- scratch ^= scratch >> 8;
-
- for (; buf < buf_end; buf++)
- scratch ^= *buf;
-
- return scratch;
-}
-
/** Read an access unit from the stream.
* Returns < 0 on error, 0 if not enough data is present in the input stream
* otherwise returns the number of bytes consumed. */
@@ -1078,8 +943,8 @@ static int read_access_unit(AVCodecConte
substream_start = end;
}
- parity_bits = calculate_parity(buf, 4);
- parity_bits ^= calculate_parity(buf + header_size, substr_header_size);
+ parity_bits = ff_mlp_calculate_parity(buf, 4);
+ parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
@@ -1145,12 +1010,12 @@ static int read_access_unit(AVCodecConte
substream_parity_present[substr]) {
uint8_t parity, checksum;
- parity = calculate_parity(buf, substream_data_len[substr] - 2);
+ parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
if ((parity ^ get_bits(&gb, 8)) != 0xa9)
av_log(m->avctx, AV_LOG_ERROR,
"Substream %d parity check failed.\n", substr);
- checksum = mlp_checksum8(buf, substream_data_len[substr] - 2);
+ checksum = ff_mlp_checksum8(buf, substream_data_len[substr] - 2);
if (checksum != get_bits(&gb, 8))
av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n",
substr);
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