*/
/**
- * @file mlpdec.c
+ * @file
* MLP decoder
*/
+#include <stdint.h>
+
#include "avcodec.h"
+#include "dsputil.h"
#include "libavutil/intreadwrite.h"
-#include "bitstream.h"
+#include "get_bits.h"
#include "libavutil/crc.h"
#include "parser.h"
#include "mlp_parser.h"
+#include "mlp.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 again I haven't seen any examples with more than two. */
-#define MAX_SUBSTREAMS 2
-
-/** Maximum sample frequency seen in files. */
-#define MAX_SAMPLERATE 192000
-
-/** The maximum number of audio samples within one access unit. */
-#define MAX_BLOCKSIZE (40 * (MAX_SAMPLERATE / 48000))
-/** The 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. */
+/** 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 "
+ "http://ffmpeg.org/bugreports.html and include "
"a sample of this file.";
typedef struct SubStream {
- //! Set if a valid restart header has been read. Otherwise the substream can not be decoded.
+ //! Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
uint8_t restart_seen;
//@{
uint8_t max_channel;
//! The number of channels input into the rematrix stage.
uint8_t max_matrix_channel;
+ //! For each channel output by the matrix, the output channel to map it to
+ uint8_t ch_assign[MAX_CHANNELS];
+
+ //! Channel coding parameters for channels in the substream
+ ChannelParams channel_params[MAX_CHANNELS];
//! The left shift applied to random noise in 0x31ea substreams.
uint8_t noise_shift;
#define PARAM_FIR (1 << 3)
#define PARAM_IIR (1 << 2)
#define PARAM_HUFFOFFSET (1 << 1)
+#define PARAM_PRESENCE (1 << 0)
//@}
//@{
//! 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];
+ int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS];
//! 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.
+ //! Left shift to apply to Huffman-decoded residuals.
uint8_t quant_step_size[MAX_CHANNELS];
- //! Number of PCM samples in current audio block.
+ //! number of PCM samples in current audio block
uint16_t blocksize;
//! Number of PCM samples decoded so far in this frame.
uint16_t blockpos;
typedef struct MLPDecodeContext {
AVCodecContext *avctx;
+ //! Current access unit being read has a major sync.
+ int is_major_sync_unit;
+
//! Set if a valid major sync block has been read. Otherwise no decoding is possible.
uint8_t params_valid;
//! Index of the last substream to decode - further substreams are skipped.
uint8_t max_decoded_substream;
- //! Number of PCM samples contained in each frame.
+ //! number of PCM samples contained in each frame
int access_unit_size;
- //! Next power of two above the number of samples in each frame.
+ //! next power of two above the number of samples in each frame
int access_unit_size_pow2;
SubStream substream[MAX_SUBSTREAMS];
- //@{
- /** filter data */
-#define FIR 0
-#define IIR 1
- //! Number of taps in filter.
- uint8_t filter_order[MAX_CHANNELS][NUM_FILTERS];
- //! Right shift to apply to output of filter.
- uint8_t filter_shift[MAX_CHANNELS][NUM_FILTERS];
-
- int32_t filter_coeff[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER];
- int32_t filter_state[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER];
- //@}
-
- //@{
- /** sample data coding infomation */
- //! Offset to apply to residual values.
- int16_t huff_offset[MAX_CHANNELS];
- //! Sign/rounding corrected version of huff_offset.
- int32_t sign_huff_offset[MAX_CHANNELS];
- //! Which VLC codebook to use to read residuals.
- uint8_t codebook[MAX_CHANNELS];
- //! Size of residual suffix not encoded using VLC.
- uint8_t huff_lsbs[MAX_CHANNELS];
- //@}
+ int matrix_changed;
+ int filter_changed[MAX_CHANNELS][NUM_FILTERS];
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;
+ int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS];
-/** 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},
- }
-};
+ DSPContext dsp;
+} MLPDecodeContext;
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;
- }
-}
-
-
-/** MLP uses checksums that seem to be based on the standard CRC algorithm,
- * but 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)
+static av_cold void init_static(void)
{
- 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;
+ if (!huff_vlc[0].bits) {
+ INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
+ &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,
+ &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,
+ &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,
unsigned int substr, unsigned int ch)
{
SubStream *s = &m->substream[substr];
- int lsb_bits = m->huff_lsbs[ch] - s->quant_step_size[ch];
- int sign_shift = lsb_bits + (m->codebook[ch] ? 2 - m->codebook[ch] : -1);
- int32_t sign_huff_offset = m->huff_offset[ch];
+ ChannelParams *cp = &s->channel_params[ch];
+ 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 (m->codebook[ch] > 0)
+ if (cp->codebook > 0)
sign_huff_offset -= 7 << lsb_bits;
if (sign_shift >= 0)
m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
for (channel = s->min_channel; channel <= s->max_channel; channel++) {
- int codebook = m->codebook[channel];
+ ChannelParams *cp = &s->channel_params[channel];
+ int codebook = cp->codebook;
int quant_step_size = s->quant_step_size[channel];
- int lsb_bits = m->huff_lsbs[channel] - quant_step_size;
+ int lsb_bits = cp->huff_lsbs - quant_step_size;
int result = 0;
if (codebook > 0)
if (lsb_bits > 0)
result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
- result += m->sign_huff_offset[channel];
+ result += cp->sign_huff_offset;
result <<= quant_step_size;
m->sample_buffer[pos + s->blockpos][channel] = result;
m->avctx = avctx;
for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
m->substream[substr].lossless_check_data = 0xffffffff;
+ dsputil_init(&m->dsp, avctx);
+
return 0;
}
return -1;
if (mh.group1_bits == 0) {
- av_log(m->avctx, AV_LOG_ERROR, "Invalid/unknown bits per sample\n");
+ 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");
+ "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 not currently supported\n");
+ "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");
+ 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 maximum supported (%d)\n",
+ "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 maximum supported (%d)\n",
+ "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 maximum supported (%d)\n",
+ "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 (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) {
+ av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
+ return -1;
+ }
if (mh.num_substreams > MAX_SUBSTREAMS) {
av_log(m->avctx, AV_LOG_ERROR,
- "Number of substreams %d is more than maximum supported by "
- "decoder. %s\n", mh.num_substreams, sample_message);
+ "Number of substreams %d is larger than the maximum supported "
+ "by the decoder. %s\n", mh.num_substreams, sample_message);
return -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->avctx->bits_per_raw_sample = mh.group1_bits;
+ if (mh.group1_bits > 16)
+ m->avctx->sample_fmt = AV_SAMPLE_FMT_S32;
+ else
+ m->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
m->params_valid = 1;
for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
uint8_t checksum;
uint8_t lossless_check;
int start_count = get_bits_count(gbp);
+ const int max_matrix_channel = m->avctx->codec_id == CODEC_ID_MLP
+ ? MAX_MATRIX_CHANNEL_MLP
+ : MAX_MATRIX_CHANNEL_TRUEHD;
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);
+ "restart header sync incorrect (got 0x%04x)\n", sync_word);
return -1;
}
+
s->noise_type = get_bits1(gbp);
+ if (m->avctx->codec_id == CODEC_ID_MLP && s->noise_type) {
+ av_log(m->avctx, AV_LOG_ERROR, "MLP must have 0x31ea sync word.\n");
+ return -1;
+ }
+
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->max_matrix_channel > max_matrix_channel) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Max matrix channel cannot be greater than %d.\n",
+ max_matrix_channel);
+ return -1;
+ }
+
+ if (s->max_channel != s->max_matrix_channel) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Max channel must be equal max matrix channel.\n");
+ return -1;
+ }
+
+ /* This should happen for TrueHD streams with >6 channels and MLP's noise
+ * type. It is not yet known if this is allowed. */
+ if (s->max_channel > MAX_MATRIX_CHANNEL_MLP && !s->noise_type) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Number of channels %d is larger than the maximum supported "
+ "by the decoder. %s\n", s->max_channel+2, sample_message);
+ return -1;
+ }
+
if (s->min_channel > s->max_channel) {
av_log(m->avctx, AV_LOG_ERROR,
"Substream min channel cannot be greater than max channel.\n");
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,
+ av_log(m->avctx, AV_LOG_DEBUG,
"Extracting %d channel downmix from substream %d. "
"Further substreams will be skipped.\n",
s->max_channel + 1, substr);
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;
+ tmp = xor_32_to_8(s->lossless_check_data);
if (tmp != lossless_check)
av_log(m->avctx, AV_LOG_WARNING,
- "Lossless check failed - expected %02x, calculated %02x\n",
+ "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);
+ memset(s->ch_assign, 0, sizeof(s->ch_assign));
+
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) {
+ if (ch_assign > s->max_matrix_channel) {
av_log(m->avctx, AV_LOG_ERROR,
- "Non 1:1 channel assignments are used in this stream. %s\n",
- sample_message);
+ "Assignment of matrix channel %d to invalid output channel %d. %s\n",
+ ch, ch_assign, sample_message);
return -1;
}
+ s->ch_assign[ch_assign] = ch;
}
- 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");
+ av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
- /* Set default decoding parameters */
+ /* Set default decoding parameters. */
s->param_presence_flags = 0xff;
s->num_primitive_matrices = 0;
s->blocksize = 8;
memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
for (ch = s->min_channel; ch <= s->max_channel; ch++) {
- m->filter_order[ch][FIR] = 0;
- m->filter_order[ch][IIR] = 0;
- m->filter_shift[ch][FIR] = 0;
- m->filter_shift[ch][IIR] = 0;
-
- /* Default audio coding is 24-bit raw PCM */
- m->huff_offset [ch] = 0;
- m->sign_huff_offset[ch] = (-1) << 23;
- m->codebook [ch] = 0;
- m->huff_lsbs [ch] = 24;
+ ChannelParams *cp = &s->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;
- }
+ if (substr == m->max_decoded_substream)
+ m->avctx->channels = s->max_matrix_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)
+ unsigned int substr, unsigned int channel,
+ unsigned int filter)
{
+ SubStream *s = &m->substream[substr];
+ FilterParams *fp = &s->channel_params[channel].filter_params[filter];
+ const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
const char fchar = filter ? 'I' : 'F';
int i, order;
- // filter is 0 for FIR, 1 for IIR
+ // Filter is 0 for FIR, 1 for IIR.
assert(filter < 2);
+ if (m->filter_changed[channel][filter]++ > 1) {
+ av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
+ return -1;
+ }
+
order = get_bits(gbp, 4);
- if (order > MAX_FILTER_ORDER) {
+ if (order > max_order) {
av_log(m->avctx, AV_LOG_ERROR,
- "%cIR filter order %d is greater than maximum %d\n",
- fchar, order, MAX_FILTER_ORDER);
+ "%cIR filter order %d is greater than maximum %d.\n",
+ fchar, order, max_order);
return -1;
}
- m->filter_order[channel][filter] = order;
+ fp->order = order;
if (order > 0) {
+ int32_t *fcoeff = s->channel_params[channel].coeff[filter];
int coeff_bits, coeff_shift;
- m->filter_shift[channel][filter] = get_bits(gbp, 4);
+ 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",
+ "%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",
+ "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++)
- m->filter_coeff[channel][filter][i] =
- get_sbits(gbp, coeff_bits) << coeff_shift;
+ fcoeff[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");
+ "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 */
+ /* TODO: Check validity of state data. */
for (i = 0; i < order; i++)
- m->filter_state[channel][filter][i] =
- get_sbits(gbp, state_bits) << state_shift;
+ fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
}
}
return 0;
}
+/** Read parameters for primitive matrices. */
+
+static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
+{
+ SubStream *s = &m->substream[substr];
+ unsigned int mat, ch;
+ const int max_primitive_matrices = m->avctx->codec_id == CODEC_ID_MLP
+ ? MAX_MATRICES_MLP
+ : MAX_MATRICES_TRUEHD;
+
+ if (m->matrix_changed++ > 1) {
+ av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
+ return -1;
+ }
+
+ s->num_primitive_matrices = get_bits(gbp, 4);
+
+ if (s->num_primitive_matrices > max_primitive_matrices) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Number of primitive matrices cannot be greater than %d.\n",
+ max_primitive_matrices);
+ return -1;
+ }
+
+ 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_matrix_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;
+ }
+
+ return 0;
+}
+
+/** Read channel parameters. */
+
+static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
+ GetBitContext *gbp, unsigned int ch)
+{
+ SubStream *s = &m->substream[substr];
+ ChannelParams *cp = &s->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, substr, ch, FIR) < 0)
+ return -1;
+
+ if (s->param_presence_flags & PARAM_IIR)
+ if (get_bits1(gbp))
+ if (read_filter_params(m, gbp, substr, ch, IIR) < 0)
+ return -1;
+
+ if (fir->order + iir->order > 8) {
+ av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
+ 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);
+
+ if (cp->huff_lsbs > 24) {
+ av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
+ return -1;
+ }
+
+ cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
+
+ return 0;
+}
+
/** Read decoding parameters that change more often than those in the restart
* header. */
unsigned int substr)
{
SubStream *s = &m->substream[substr];
- unsigned int mat, ch;
+ unsigned int ch;
- if (get_bits1(gbp))
- s->param_presence_flags = get_bits(gbp, 8);
+ if (s->param_presence_flags & PARAM_PRESENCE)
+ 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");
+ if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
+ av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
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 (get_bits1(gbp))
+ if (read_matrix_params(m, substr, gbp) < 0)
+ return -1;
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 */
- }
+ for (ch = 0; ch <= s->max_matrix_channel; ch++)
+ s->output_shift[ch] = get_sbits(gbp, 4);
if (s->param_presence_flags & PARAM_QUANTSTEP)
if (get_bits1(gbp))
for (ch = 0; ch <= s->max_channel; ch++) {
+ ChannelParams *cp = &s->channel_params[ch];
+
s->quant_step_size[ch] = get_bits(gbp, 4);
- /* TODO: validate */
- m->sign_huff_offset[ch] = calculate_sign_huff(m, substr, ch);
+ cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
}
for (ch = s->min_channel; ch <= s->max_channel; ch++)
- if (get_bits1(gbp)) {
- 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 (m->filter_order[ch][FIR] && m->filter_order[ch][IIR] &&
- m->filter_shift[ch][FIR] != m->filter_shift[ch][IIR]) {
- av_log(m->avctx, AV_LOG_ERROR,
- "FIR and IIR filters must use same precision\n");
+ if (get_bits1(gbp))
+ if (read_channel_params(m, substr, gbp, ch) < 0)
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 (!m->filter_order[ch][FIR] && m->filter_order[ch][IIR])
- m->filter_shift[ch][FIR] = m->filter_shift[ch][IIR];
-
- if (s->param_presence_flags & PARAM_HUFFOFFSET)
- if (get_bits1(gbp))
- m->huff_offset[ch] = get_sbits(gbp, 15);
-
- m->codebook [ch] = get_bits(gbp, 2);
- m->huff_lsbs[ch] = get_bits(gbp, 5);
-
- m->sign_huff_offset[ch] = calculate_sign_huff(m, substr, ch);
-
- /* TODO: validate */
- }
return 0;
}
unsigned int channel)
{
SubStream *s = &m->substream[substr];
- int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER];
- unsigned int filter_shift = m->filter_shift[channel][FIR];
+ const int32_t *fircoeff = s->channel_params[channel].coeff[FIR];
+ int32_t state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FIR_ORDER];
+ int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE;
+ int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE;
+ FilterParams *fir = &s->channel_params[channel].filter_params[FIR];
+ FilterParams *iir = &s->channel_params[channel].filter_params[IIR];
+ unsigned int filter_shift = 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],
- &m->filter_state[channel][j][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 < m->filter_order[channel][j]; order++)
- accum += (int64_t)filter_state_buffer[j][index + order] *
- m->filter_coeff[channel][j][order];
+ memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t));
+ memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t));
- accum = accum >> filter_shift;
- result = (accum + residual) & mask;
+ m->dsp.mlp_filter_channel(firbuf, fircoeff,
+ fir->order, iir->order,
+ filter_shift, mask, s->blocksize,
+ &m->sample_buffer[s->blockpos][channel]);
- --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(&m->filter_state[channel][j][0],
- & filter_state_buffer [j][index],
- MAX_FILTER_ORDER * sizeof(int32_t));
- }
+ memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t));
+ memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t));
}
/** Read a block of PCM residual data (or actual if no filtering active). */
}
if (s->blockpos + s->blocksize > m->access_unit_size) {
- av_log(m->avctx, AV_LOG_ERROR, "Too many audio samples in frame\n");
+ 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++) {
+ 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++) {
+ 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");
+ 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 */
+/** 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,
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]);
+ int32_t *coeffs = s->matrix_coeff[mat];
+ int index = s->num_primitive_matrices - mat;
+ int index2 = 2 * index + 1;
/* TODO: DSPContext? */
for (i = 0; i < s->blockpos; i++) {
+ int32_t bypassed_lsb = m->bypassed_lsbs[i][mat];
+ int32_t *samples = m->sample_buffer[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];
- }
+
+ for (src_ch = 0; src_ch <= maxchan; src_ch++)
+ accum += (int64_t) samples[src_ch] * coeffs[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);
+ index &= m->access_unit_size_pow2 - 1;
accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
+ index += index2;
}
- m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
- + m->bypassed_lsbs[i][mat];
+
+ samples[dest_ch] = ((accum >> 14) & mask) + bypassed_lsb;
}
}
}
uint8_t *data, unsigned int *data_size, int is32)
{
SubStream *s = &m->substream[substr];
- unsigned int i, ch = 0;
+ unsigned int i, out_ch = 0;
int32_t *data_32 = (int32_t*) data;
int16_t *data_16 = (int16_t*) data;
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;
+ for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
+ int mat_ch = s->ch_assign[out_ch];
+ int32_t sample = m->sample_buffer[i][mat_ch]
+ << s->output_shift[mat_ch];
+ s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
if (is32) *data_32++ = sample << 8;
else *data_16++ = sample >> 8;
}
}
- *data_size = i * ch * (is32 ? 4 : 2);
+ *data_size = i * out_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)
+ if (m->avctx->sample_fmt == AV_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)
-{
- 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. */
+ * @return negative on error, 0 if not enough data is present in the input stream,
+ * otherwise the number of bytes consumed. */
static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
- const uint8_t *buf, int buf_size)
+ AVPacket *avpkt)
{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
MLPDecodeContext *m = avctx->priv_data;
GetBitContext gb;
unsigned int length, substr;
length = (AV_RB16(buf) & 0xfff) * 2;
- if (length > buf_size)
+ if (length < 4 || length > buf_size)
return -1;
init_get_bits(&gb, (buf + 4), (length - 4) * 8);
+ m->is_major_sync_unit = 0;
if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
- dprintf(m->avctx, "Found major sync\n");
if (read_major_sync(m, &gb) < 0)
goto error;
+ m->is_major_sync_unit = 1;
header_size += 28;
}
if (!m->params_valid) {
av_log(m->avctx, AV_LOG_WARNING,
- "Stream parameters not seen; skipping frame\n");
+ "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;
+ int extraword_present, checkdata_present, end, nonrestart_substr;
extraword_present = get_bits1(&gb);
- skip_bits1(&gb);
+ nonrestart_substr = get_bits1(&gb);
checkdata_present = get_bits1(&gb);
skip_bits1(&gb);
substr_header_size += 2;
if (extraword_present) {
+ if (m->avctx->codec_id == CODEC_ID_MLP) {
+ av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
+ goto error;
+ }
skip_bits(&gb, 16);
substr_header_size += 2;
}
+ if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
+ av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
+ goto error;
+ }
+
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 "
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");
SubStream *s = &m->substream[substr];
init_get_bits(&gb, buf, substream_data_len[substr] * 8);
+ m->matrix_changed = 0;
+ memset(m->filter_changed, 0, sizeof(m->filter_changed));
+
s->blockpos = 0;
do {
if (get_bits1(&gb)) {
if (get_bits1(&gb)) {
- /* A restart header should be present */
+ /* 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);
+ if (!s->restart_seen)
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);
+ if (!s->restart_seen)
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);
+ if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
+ goto substream_length_mismatch;
+
+ } while (!get_bits1(&gb));
skip_bits(&gb, (-get_bits_count(&gb)) & 15);
- if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 48 &&
- (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) >= 32) {
+ int shorten_by;
+
+ if (get_bits(&gb, 16) != 0xD234)
+ return -1;
+
+ shorten_by = get_bits(&gb, 16);
+ if (m->avctx->codec_id == CODEC_ID_TRUEHD && shorten_by & 0x2000)
+ s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
+ else if (m->avctx->codec_id == CODEC_ID_MLP && shorten_by != 0xD234)
+ return -1;
+
+ if (substr == m->max_decoded_substream)
+ av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
}
+
if (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);
+ if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
+ goto substream_length_mismatch;
- 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;
+ parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
+ checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
+
+ if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
+ av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
+ if ( get_bits(&gb, 8) != checksum)
+ av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
}
+ if (substream_data_len[substr] * 8 != get_bits_count(&gb))
+ goto substream_length_mismatch;
+
next_substr:
+ if (!s->restart_seen)
+ av_log(m->avctx, AV_LOG_ERROR,
+ "No restart header present in substream %d.\n", substr);
+
buf += substream_data_len[substr];
}
return length;
+substream_length_mismatch:
+ av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
+ return -1;
+
error:
m->params_valid = 0;
return -1;
}
-AVCodec mlp_decoder = {
+AVCodec ff_mlp_decoder = {
"mlp",
- CODEC_TYPE_AUDIO,
+ AVMEDIA_TYPE_AUDIO,
CODEC_ID_MLP,
sizeof(MLPDecodeContext),
mlp_decode_init,
NULL,
NULL,
read_access_unit,
- .long_name = NULL_IF_CONFIG_SMALL("Meridian Lossless Packing"),
+ .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
};
+#if CONFIG_TRUEHD_DECODER
+AVCodec ff_truehd_decoder = {
+ "truehd",
+ AVMEDIA_TYPE_AUDIO,
+ CODEC_ID_TRUEHD,
+ sizeof(MLPDecodeContext),
+ mlp_decode_init,
+ NULL,
+ NULL,
+ read_access_unit,
+ .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
+};
+#endif /* CONFIG_TRUEHD_DECODER */