* @file
* QDM2 decoder
* @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
+ *
* The decoder is not perfect yet, there are still some distortions
* especially on files encoded with 16 or 8 subbands.
*/
#include <stddef.h>
#include <stdio.h>
-#define ALT_BITSTREAM_READER_LE
+#define BITSTREAM_READER_LE
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
-#include "fft.h"
+#include "rdft.h"
+#include "mpegaudiodsp.h"
#include "mpegaudio.h"
#include "qdm2data.h"
#define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
-#define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
-
#define SAMPLES_NEEDED \
av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
#define SAMPLES_NEEDED_2(why) \
av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
+#define QDM2_MAX_FRAME_SIZE 512
typedef int8_t sb_int8_array[2][30][64];
* QDM2 decoder context
*/
typedef struct {
+ AVFrame frame;
+
/// Parameters from codec header, do not change during playback
int nb_channels; ///< number of channels
int channels; ///< number of channels
/// Parameters built from header parameters, do not change during playback
int group_order; ///< order of frame group
int fft_order; ///< order of FFT (actually fftorder+1)
- int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im)
int frame_size; ///< size of data frame
int frequency_range;
int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
/// I/O data
const uint8_t *compressed_data;
int compressed_size;
- float output_buffer[1024];
+ float output_buffer[QDM2_MAX_FRAME_SIZE * 2];
/// Synthesis filter
- DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2];
+ MPADSPContext mpadsp;
+ DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
int synth_buf_offset[MPA_MAX_CHANNELS];
- DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
+ DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
+ DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
/// Mixed temporary data used in decoding
float tone_level[MPA_MAX_CHANNELS][30][64];
} QDM2Context;
-static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
-
static VLC vlc_tab_level;
static VLC vlc_tab_diff;
static VLC vlc_tab_run;
}
}
-
-/* for floating point to fixed point conversion */
-static const float f2i_scale = (float) (1 << (FRAC_BITS - 15));
-
-
static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
{
int value;
for (ch = 0; ch < q->nb_channels; ch++)
for (j = 0; j < 64; j++) {
- q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
- q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
+ q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
+ q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
}
}
int j,k;
int ch;
int run, case_val;
- int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};
+ static const int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};
for (ch = 0; ch < channels; ch++) {
for (j = 0; j < 64; ) {
else if (sb >= 24)
joined_stereo = 1;
else
- joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
+ joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1 (gb) : 0;
if (joined_stereo) {
- if (BITS_LEFT(length,gb) >= 16)
+ if (get_bits_left(gb) >= 16)
for (j = 0; j < 16; j++)
sign_bits[j] = get_bits1 (gb);
}
for (ch = 0; ch < channels; ch++) {
- zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
+ zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
type34_predictor = 0.0;
type34_first = 1;
for (j = 0; j < 128; ) {
switch (q->coding_method[ch][sb][j / 2]) {
case 8:
- if (BITS_LEFT(length,gb) >= 10) {
+ if (get_bits_left(gb) >= 10) {
if (zero_encoding) {
for (k = 0; k < 5; k++) {
if ((j + 2 * k) >= 128)
break;
case 10:
- if (BITS_LEFT(length,gb) >= 1) {
+ if (get_bits_left(gb) >= 1) {
float f = 0.81;
if (get_bits1(gb))
break;
case 16:
- if (BITS_LEFT(length,gb) >= 10) {
+ if (get_bits_left(gb) >= 10) {
if (zero_encoding) {
for (k = 0; k < 5; k++) {
if ((j + k) >= 128)
break;
case 24:
- if (BITS_LEFT(length,gb) >= 7) {
+ if (get_bits_left(gb) >= 7) {
n = get_bits(gb, 7);
for (k = 0; k < 3; k++)
samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
break;
case 30:
- if (BITS_LEFT(length,gb) >= 4)
+ if (get_bits_left(gb) >= 4)
samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
else
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
break;
case 34:
- if (BITS_LEFT(length,gb) >= 7) {
+ if (get_bits_left(gb) >= 7) {
if (type34_first) {
type34_div = (float)(1 << get_bits(gb, 2));
samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
for (chs = 0; chs < q->nb_channels; chs++)
for (k = 0; k < run; k++)
if ((j + k) < 128)
- q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
+ q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
} else {
for (k = 0; k < run; k++)
if ((j + k) < 128)
- q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
+ q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
}
j += run;
*
* @param quantized_coeffs pointer to quantized_coeffs[ch][0]
* @param gb bitreader context
- * @param length packet length in bits
*/
-static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
+static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb)
{
int i, k, run, level, diff;
- if (BITS_LEFT(length,gb) < 16)
+ if (get_bits_left(gb) < 16)
return;
level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
quantized_coeffs[0] = level;
for (i = 0; i < 7; ) {
- if (BITS_LEFT(length,gb) < 16)
+ if (get_bits_left(gb) < 16)
break;
run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
- if (BITS_LEFT(length,gb) < 16)
+ if (get_bits_left(gb) < 16)
break;
diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
*
* @param q context
* @param gb bitreader context
- * @param length packet length in bits
*/
-static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
+static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb)
{
int sb, j, k, n, ch;
for (ch = 0; ch < q->nb_channels; ch++) {
- init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
+ init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
- if (BITS_LEFT(length,gb) < 16) {
+ if (get_bits_left(gb) < 16) {
memset(q->quantized_coeffs[ch][0], 0, 8);
break;
}
for (sb = 0; sb < n; sb++)
for (ch = 0; ch < q->nb_channels; ch++)
for (j = 0; j < 8; j++) {
- if (BITS_LEFT(length,gb) < 1)
+ if (get_bits_left(gb) < 1)
break;
if (get_bits1(gb)) {
for (k=0; k < 8; k++) {
- if (BITS_LEFT(length,gb) < 16)
+ if (get_bits_left(gb) < 16)
break;
q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
}
for (sb = 0; sb < n; sb++)
for (ch = 0; ch < q->nb_channels; ch++) {
- if (BITS_LEFT(length,gb) < 16)
+ if (get_bits_left(gb) < 16)
break;
q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
if (sb > 19)
for (sb = 0; sb < n; sb++)
for (ch = 0; ch < q->nb_channels; ch++)
for (j = 0; j < 8; j++) {
- if (BITS_LEFT(length,gb) < 16)
+ if (get_bits_left(gb) < 16)
break;
q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
}
* @param node pointer to node with packet
* @param length packet length in bits
*/
-static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
+static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node)
{
GetBitContext gb;
- init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
-
- if (length != 0) {
- init_tone_level_dequantization(q, &gb, length);
+ if (node) {
+ init_get_bits(&gb, node->packet->data, node->packet->size * 8);
+ init_tone_level_dequantization(q, &gb);
fill_tone_level_array(q, 1);
} else {
fill_tone_level_array(q, 0);
*
* @param q context
* @param node pointer to node with packet
- * @param length packet length in bit
*/
-static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
+static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node)
{
GetBitContext gb;
+ int length = 0;
+
+ if (node) {
+ length = node->packet->size * 8;
+ init_get_bits(&gb, node->packet->data, length);
+ }
- init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
if (length >= 32) {
int c = get_bits (&gb, 13);
* @param node pointer to node with packet
* @param length packet length in bits
*/
-static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
+static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node)
{
GetBitContext gb;
+ int length = 0;
+
+ if (node) {
+ length = node->packet->size * 8;
+ init_get_bits(&gb, node->packet->data, length);
+ }
- init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
}
nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
if (nodes[1] != NULL)
- process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
+ process_subpacket_10(q, nodes[1]);
else
- process_subpacket_10(q, NULL, 0);
+ process_subpacket_10(q, NULL);
nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
- process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
+ process_subpacket_11(q, nodes[2]);
else
- process_subpacket_11(q, NULL, 0);
+ process_subpacket_11(q, NULL);
nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
- process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
+ process_subpacket_12(q, nodes[3]);
else
- process_subpacket_12(q, NULL, 0);
+ process_subpacket_12(q, NULL);
}
process_synthesis_subpackets(q, q->sub_packet_list_D);
q->do_synth_filter = 1;
} else if (q->do_synth_filter) {
- process_subpacket_10(q, NULL, 0);
- process_subpacket_11(q, NULL, 0);
- process_subpacket_12(q, NULL, 0);
+ process_subpacket_10(q, NULL);
+ process_subpacket_11(q, NULL);
+ process_subpacket_12(q, NULL);
}
/* **************************************************************** */
}
return;
local_int_14 = (offset >> local_int_8);
+ if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
+ return;
if (q->nb_channels > 1) {
channel = get_bits1(gb);
static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
{
const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
+ float *out = q->output_buffer + channel;
int i;
q->fft.complex[channel][0].re *= 2.0f;
q->fft.complex[channel][0].im = 0.0f;
q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
/* add samples to output buffer */
- for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
- q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
+ for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
+ out[0] += q->fft.complex[channel][i].re * gain;
+ out[q->channels] += q->fft.complex[channel][i].im * gain;
+ out += 2 * q->channels;
+ }
}
*/
static void qdm2_synthesis_filter (QDM2Context *q, int index)
{
- OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
int i, k, ch, sb_used, sub_sampling, dither_state = 0;
/* copy sb_samples */
q->sb_samples[ch][(8 * index) + i][k] = 0;
for (ch = 0; ch < q->nb_channels; ch++) {
- OUT_INT *samples_ptr = samples + ch;
+ float *samples_ptr = q->samples + ch;
for (i = 0; i < 8; i++) {
- ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
- ff_mpa_synth_window, &dither_state,
+ ff_mpa_synth_filter_float(&q->mpadsp,
+ q->synth_buf[ch], &(q->synth_buf_offset[ch]),
+ ff_mpa_synth_window_float, &dither_state,
samples_ptr, q->nb_channels,
q->sb_samples[ch][(8 * index) + i]);
samples_ptr += 32 * q->nb_channels;
for (ch = 0; ch < q->channels; ch++)
for (i = 0; i < q->frame_size; i++)
- q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
+ q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
}
initialized = 1;
qdm2_init_vlc();
- ff_mpa_synth_init(ff_mpa_synth_window);
+ ff_mpa_synth_init_float(ff_mpa_synth_window_float);
softclip_table_init();
rnd_table_init();
init_noise_samples();
PRINT("checksum_size",q->checksum_size);
PRINT("channels",q->channels);
PRINT("nb_channels",q->nb_channels);
- PRINT("fft_frame_size",q->fft_frame_size);
PRINT("fft_size",q->fft_size);
PRINT("sub_sampling",q->sub_sampling);
PRINT("fft_order",q->fft_order);
avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
extradata += 4;
+ if (s->channels > MPA_MAX_CHANNELS)
+ return AVERROR_INVALIDDATA;
avctx->sample_rate = AV_RB32(extradata);
extradata += 4;
extradata += 4;
s->checksum_size = AV_RB32(extradata);
+ if (s->checksum_size >= 1U << 28) {
+ av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
+ return AVERROR_INVALIDDATA;
+ }
s->fft_order = av_log2(s->fft_size) + 1;
- s->fft_frame_size = 2 * s->fft_size; // complex has two floats
// something like max decodable tones
s->group_order = av_log2(s->group_size) + 1;
s->frame_size = s->group_size / 16; // 16 iterations per super block
+ if (s->frame_size > QDM2_MAX_FRAME_SIZE)
+ return AVERROR_INVALIDDATA;
s->sub_sampling = s->fft_order - 7;
s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
}
ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
+ ff_mpadsp_init(&s->mpadsp);
qdm2_init(s);
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ avcodec_get_frame_defaults(&s->frame);
+ avctx->coded_frame = &s->frame;
+
// dump_context(s);
return 0;
}
}
-static int qdm2_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- AVPacket *avpkt)
+static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
QDM2Context *s = avctx->priv_data;
- int16_t *out = data;
- int i;
+ int16_t *out;
+ int i, ret;
if(!buf)
return 0;
if(buf_size < s->checksum_size)
return -1;
- av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
- buf_size, buf, s->checksum_size, data, *data_size);
+ /* get output buffer */
+ s->frame.nb_samples = 16 * s->frame_size;
+ if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+ out = (int16_t *)s->frame.data[0];
for (i = 0; i < 16; i++) {
if (qdm2_decode(s, buf, out) < 0)
out += s->channels * s->frame_size;
}
- *data_size = (uint8_t*)out - (uint8_t*)data;
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = s->frame;
return s->checksum_size;
}
AVCodec ff_qdm2_decoder =
{
- .name = "qdm2",
- .type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_QDM2,
+ .name = "qdm2",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_QDM2,
.priv_data_size = sizeof(QDM2Context),
- .init = qdm2_decode_init,
- .close = qdm2_decode_close,
- .decode = qdm2_decode_frame,
- .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
+ .init = qdm2_decode_init,
+ .close = qdm2_decode_close,
+ .decode = qdm2_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
+ .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
};
projects / ffmpeg / blobdiff