#include <stddef.h>
#include <stdio.h>
-#define BITSTREAM_READER_LE
#include "libavutil/channel_layout.h"
+
+#define BITSTREAM_READER_LE
#include "avcodec.h"
#include "get_bits.h"
#include "internal.h"
-#include "rdft.h"
-#include "mpegaudiodsp.h"
#include "mpegaudio.h"
+#include "mpegaudiodsp.h"
+#include "rdft.h"
#include "qdm2data.h"
#include "qdm2_tablegen.h"
-#undef NDEBUG
-#include <assert.h>
-
#define QDM2_LIST_ADD(list, size, packet) \
do { \
/**
* Subpacket
*/
-typedef struct {
+typedef struct QDM2SubPacket {
int type; ///< subpacket type
unsigned int size; ///< subpacket size
const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
} QDM2SubPNode;
-typedef struct {
+typedef struct QDM2Complex {
float re;
float im;
} QDM2Complex;
-typedef struct {
+typedef struct FFTTone {
float level;
QDM2Complex *complex;
const float *table;
short cutoff;
} FFTTone;
-typedef struct {
+typedef struct FFTCoefficient {
int16_t sub_packet;
uint8_t channel;
int16_t offset;
uint8_t phase;
} FFTCoefficient;
-typedef struct {
+typedef struct QDM2FFT {
DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
} QDM2FFT;
/**
* QDM2 decoder context
*/
-typedef struct {
+typedef struct QDM2Context {
/// Parameters from codec header, do not change during playback
int nb_channels; ///< number of channels
int channels; ///< number of channels
/**
* QDM2 checksum
*
- * @param data pointer to data to be checksum'ed
+ * @param data pointer to data to be checksummed
* @param length data length
* @param value checksum value
*
}
tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
}
- acc = 0;
- for (ch = 0; ch < nb_channels; ch++)
- for (sb = 0; sb < 30; sb++)
- for (j = 0; j < 64; j++)
- acc += tone_level_idx_temp[ch][sb][j];
-
- multres = 0x66666667LL * (acc * 10);
- esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
- for (ch = 0; ch < nb_channels; ch++)
- for (sb = 0; sb < 30; sb++)
- for (j = 0; j < 64; j++) {
- comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
- if (comp < 0)
- comp += 0xff;
- comp /= 256; // signed shift
- switch(sb) {
- case 0:
- if (comp < 30)
- comp = 30;
- comp += 15;
- break;
- case 1:
- if (comp < 24)
- comp = 24;
- comp += 10;
- break;
- case 2:
- case 3:
- case 4:
- if (comp < 16)
- comp = 16;
- }
- if (comp <= 5)
- tmp = 0;
- else if (comp <= 10)
- tmp = 10;
- else if (comp <= 16)
- tmp = 16;
- else if (comp <= 24)
- tmp = -1;
- else
- tmp = 0;
- coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
+
+ acc = 0;
+ for (ch = 0; ch < nb_channels; ch++)
+ for (sb = 0; sb < 30; sb++)
+ for (j = 0; j < 64; j++)
+ acc += tone_level_idx_temp[ch][sb][j];
+
+ multres = 0x66666667LL * (acc * 10);
+ esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
+ for (ch = 0; ch < nb_channels; ch++)
+ for (sb = 0; sb < 30; sb++)
+ for (j = 0; j < 64; j++) {
+ comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
+ if (comp < 0)
+ comp += 0xff;
+ comp /= 256; // signed shift
+ switch(sb) {
+ case 0:
+ if (comp < 30)
+ comp = 30;
+ comp += 15;
+ break;
+ case 1:
+ if (comp < 24)
+ comp = 24;
+ comp += 10;
+ break;
+ case 2:
+ case 3:
+ case 4:
+ if (comp < 16)
+ comp = 16;
}
+ if (comp <= 5)
+ tmp = 0;
+ else if (comp <= 10)
+ tmp = 10;
+ else if (comp <= 16)
+ tmp = 16;
+ else if (comp <= 24)
+ tmp = -1;
+ else
+ tmp = 0;
+ coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
+ }
+ for (sb = 0; sb < 30; sb++)
+ fix_coding_method_array(sb, nb_channels, coding_method);
+ for (ch = 0; ch < nb_channels; ch++)
for (sb = 0; sb < 30; sb++)
- fix_coding_method_array(sb, nb_channels, coding_method);
- for (ch = 0; ch < nb_channels; ch++)
- for (sb = 0; sb < 30; sb++)
- for (j = 0; j < 64; j++)
- if (sb >= 10) {
- if (coding_method[ch][sb][j] < 10)
- coding_method[ch][sb][j] = 10;
+ for (j = 0; j < 64; j++)
+ if (sb >= 10) {
+ if (coding_method[ch][sb][j] < 10)
+ coding_method[ch][sb][j] = 10;
+ } else {
+ if (sb >= 2) {
+ if (coding_method[ch][sb][j] < 16)
+ coding_method[ch][sb][j] = 16;
} else {
- if (sb >= 2) {
- if (coding_method[ch][sb][j] < 16)
- coding_method[ch][sb][j] = 16;
- } else {
- if (coding_method[ch][sb][j] < 30)
- coding_method[ch][sb][j] = 30;
- }
+ if (coding_method[ch][sb][j] < 30)
+ coding_method[ch][sb][j] = 30;
}
+ }
} else { // superblocktype_2_3 != 0
for (ch = 0; ch < nb_channels; ch++)
for (sb = 0; sb < 30; sb++)
}
/**
- *
* Called by process_subpacket_11 to process more data from subpacket 11
* with sb 0-8.
* Called by process_subpacket_12 to process data from subpacket 12 with
if (!avctx->extradata || (avctx->extradata_size < 48)) {
av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
extradata = avctx->extradata;
if (extradata_size < 12) {
av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
extradata_size);
- return -1;
+ return AVERROR_INVALIDDATA;
}
if (memcmp(extradata, "frmaQDM", 7)) {
av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
if (extradata[7] == 'C') {
// s->is_qdmc = 1;
- av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
- return -1;
+ avpriv_report_missing_feature(avctx, "QDMC version 1");
+ return AVERROR_PATCHWELCOME;
}
extradata += 8;
if(size > extradata_size){
av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
extradata_size, size);
- return -1;
+ return AVERROR_INVALIDDATA;
}
extradata += 4;
av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
extradata += 8;
// Fail on unknown fft order
if ((s->fft_order < 7) || (s->fft_order > 9)) {
- av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
- return -1;
+ avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
+ return AVERROR_PATCHWELCOME;
}
if (s->fft_size != (1 << (s->fft_order - 1))) {
av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
q->sub_packet = (q->sub_packet + 1) % 16;
- /* clip and convert output float[] to 16bit signed samples */
+ /* clip and convert output float[] to 16-bit signed samples */
for (i = 0; i < frame_size; i++) {
int value = (int)q->output_buffer[i];
out = (int16_t *)frame->data[0];
for (i = 0; i < 16; i++) {
- if (qdm2_decode(s, buf, out) < 0)
- return -1;
+ if ((ret = qdm2_decode(s, buf, out)) < 0)
+ return ret;
out += s->channels * s->frame_size;
}
.init_static_data = qdm2_init_static_data,
.close = qdm2_decode_close,
.decode = qdm2_decode_frame,
- .capabilities = CODEC_CAP_DR1,
+ .capabilities = AV_CODEC_CAP_DR1,
};
projects / ffmpeg / blobdiff