Merge remote branch 'qatar/master'
* qatar/master: vorbisdec: Rename silly "class_" variable to plain "class". simple_idct_alpha: Drop some useless casts. Simplify av_log_missing_feature(). ac3enc: remove check for mismatching channels and channel_layout If AVCodecContext.channels is 0 and AVCodecContext.channel_layout is non-zero, set channels based on channel_layout. If AVCodecContext.channel_layout and AVCodecContext.channels are both non-zero, check to make sure they do not contradict eachother. cosmetics: indentation Check AVCodec.supported_samplerates and AVCodec.channel_layouts in avcodec_open(). aacdec: remove sf_scale and sf_offset. aacdec: use a scale of 2 in the LTP MDCT rather than doubling the coefficient table values from the spec. Define POW_SF2_ZERO in aac.h and use for ff_aac_pow2sf_tabp[] offsets instead of hardcoding 200 everywhere. Large intensity stereo and PNS indices are legal. Clip them instead of erroring out. A magnitude of 100 corresponds to 2^25 so the will most likely result in clipped output anyway. qpeg: use reget_buffer() in decode_frame() ultimotion: use reget_buffer() in ulti_decode_frame() smacker: remove unnecessary call to avctx->release_buffer in decode_frame() avparser: don't av_malloc(0). Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
commit
0665199e43
@ -130,6 +130,7 @@ typedef struct {
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#define SCALE_MAX_POS 255 ///< scalefactor index maximum value
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#define SCALE_MAX_POS 255 ///< scalefactor index maximum value
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#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard
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#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard
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#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference
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#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference
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#define POW_SF2_ZERO 200 ///< ff_aac_pow2sf_tab index corresponding to pow(2, 0);
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/**
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/**
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* Long Term Prediction
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* Long Term Prediction
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@ -292,8 +293,6 @@ typedef struct {
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* @{
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* @{
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*/
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*/
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float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
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float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
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float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
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int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
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/** @} */
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/** @} */
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DECLARE_ALIGNED(32, float, temp)[128];
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DECLARE_ALIGNED(32, float, temp)[128];
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@ -29,13 +29,14 @@
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#include "libavcodec/aac_tables.h"
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#include "libavcodec/aac_tables.h"
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#else
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#else
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#include "libavutil/mathematics.h"
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#include "libavutil/mathematics.h"
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#include "libavcodec/aac.h"
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float ff_aac_pow2sf_tab[428];
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float ff_aac_pow2sf_tab[428];
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void ff_aac_tableinit(void)
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void ff_aac_tableinit(void)
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{
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{
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int i;
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int i;
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for (i = 0; i < 428; i++)
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for (i = 0; i < 428; i++)
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ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
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ff_aac_pow2sf_tab[i] = pow(2, (i - POW_SF2_ZERO) / 4.);
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}
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}
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#endif /* CONFIG_HARDCODED_TABLES */
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#endif /* CONFIG_HARDCODED_TABLES */
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@ -109,8 +109,8 @@ static av_always_inline float quantize_and_encode_band_cost_template(
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int *bits, int BT_ZERO, int BT_UNSIGNED,
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int *bits, int BT_ZERO, int BT_UNSIGNED,
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int BT_PAIR, int BT_ESC)
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int BT_PAIR, int BT_ESC)
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{
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{
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const float IQ = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
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const float IQ = ff_aac_pow2sf_tab[POW_SF2_ZERO + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
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const float Q = ff_aac_pow2sf_tab[200 - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
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const float Q = ff_aac_pow2sf_tab[POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
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const float CLIPPED_ESCAPE = 165140.0f*IQ;
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const float CLIPPED_ESCAPE = 165140.0f*IQ;
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int i, j;
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int i, j;
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float cost = 0;
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float cost = 0;
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@ -281,7 +281,7 @@ static float find_max_val(int group_len, int swb_size, const float *scaled) {
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}
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}
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static int find_min_book(float maxval, int sf) {
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static int find_min_book(float maxval, int sf) {
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float Q = ff_aac_pow2sf_tab[200 - sf + SCALE_ONE_POS - SCALE_DIV_512];
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float Q = ff_aac_pow2sf_tab[POW_SF2_ZERO - sf + SCALE_ONE_POS - SCALE_DIV_512];
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float Q34 = sqrtf(Q * sqrtf(Q));
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float Q34 = sqrtf(Q * sqrtf(Q));
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int qmaxval, cb;
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int qmaxval, cb;
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qmaxval = maxval * Q34 + 0.4054f;
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qmaxval = maxval * Q34 + 0.4054f;
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@ -956,7 +956,7 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
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dist -= b;
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dist -= b;
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}
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}
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dist *= 1.0f / 512.0f / lambda;
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dist *= 1.0f / 512.0f / lambda;
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quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[200 - scf + SCALE_ONE_POS - SCALE_DIV_512]);
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quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[POW_SF2_ZERO - scf + SCALE_ONE_POS - SCALE_DIV_512]);
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if (quant_max >= 8191) { // too much, return to the previous quantizer
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if (quant_max >= 8191) { // too much, return to the previous quantizer
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sce->sf_idx[w*16+g] = prev_scf;
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sce->sf_idx[w*16+g] = prev_scf;
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break;
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break;
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@ -579,12 +579,6 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
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ac->random_state = 0x1f2e3d4c;
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ac->random_state = 0x1f2e3d4c;
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// -1024 - Compensate wrong IMDCT method.
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// 60 - Required to scale values to the correct range [-32768,32767]
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// for float to int16 conversion. (1 << (60 / 4)) == 32768
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ac->sf_scale = 1. / -1024.;
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ac->sf_offset = 60;
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ff_aac_tableinit();
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ff_aac_tableinit();
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INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
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INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
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@ -592,9 +586,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
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ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
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ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
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352);
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352);
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ff_mdct_init(&ac->mdct, 11, 1, 1.0);
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ff_mdct_init(&ac->mdct, 11, 1, 1.0/1024.0);
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ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);
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ff_mdct_init(&ac->mdct_small, 8, 1, 1.0/128.0);
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ff_mdct_init(&ac->mdct_ltp, 11, 0, 1.0);
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ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0);
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// window initialization
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// window initialization
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ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
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ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
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ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
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ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
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@ -652,7 +646,7 @@ static void decode_ltp(AACContext *ac, LongTermPrediction *ltp,
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int sfb;
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int sfb;
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ltp->lag = get_bits(gb, 11);
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ltp->lag = get_bits(gb, 11);
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ltp->coef = ltp_coef[get_bits(gb, 3)] * ac->sf_scale;
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ltp->coef = ltp_coef[get_bits(gb, 3)];
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for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
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for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
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ltp->used[sfb] = get_bits1(gb);
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ltp->used[sfb] = get_bits1(gb);
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}
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}
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@ -790,9 +784,9 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
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enum BandType band_type[120],
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enum BandType band_type[120],
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int band_type_run_end[120])
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int band_type_run_end[120])
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{
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{
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const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
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int g, i, idx = 0;
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int g, i, idx = 0;
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int offset[3] = { global_gain, global_gain - 90, 100 };
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int offset[3] = { global_gain, global_gain - 90, 0 };
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int clipped_offset;
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int noise_flag = 1;
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int noise_flag = 1;
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static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
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static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
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for (g = 0; g < ics->num_window_groups; g++) {
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for (g = 0; g < ics->num_window_groups; g++) {
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@ -804,12 +798,14 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
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} else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
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} else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
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for (; i < run_end; i++, idx++) {
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for (; i < run_end; i++, idx++) {
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offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
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offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
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if (offset[2] > 255U) {
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clipped_offset = av_clip(offset[2], -155, 100);
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av_log(ac->avctx, AV_LOG_ERROR,
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if (offset[2] != clipped_offset) {
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"%s (%d) out of range.\n", sf_str[2], offset[2]);
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av_log_ask_for_sample(ac->avctx, "Intensity stereo "
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return -1;
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"position clipped (%d -> %d).\nIf you heard an "
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"audible artifact, there may be a bug in the "
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"decoder. ", offset[2], clipped_offset);
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}
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}
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sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
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sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
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}
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}
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} else if (band_type[idx] == NOISE_BT) {
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} else if (band_type[idx] == NOISE_BT) {
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for (; i < run_end; i++, idx++) {
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for (; i < run_end; i++, idx++) {
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@ -817,12 +813,14 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
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offset[1] += get_bits(gb, 9) - 256;
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offset[1] += get_bits(gb, 9) - 256;
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else
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else
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offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
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offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
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if (offset[1] > 255U) {
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clipped_offset = av_clip(offset[1], -100, 155);
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av_log(ac->avctx, AV_LOG_ERROR,
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if (offset[2] != clipped_offset) {
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"%s (%d) out of range.\n", sf_str[1], offset[1]);
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av_log_ask_for_sample(ac->avctx, "Noise gain clipped "
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return -1;
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"(%d -> %d).\nIf you heard an audible "
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"artifact, there may be a bug in the decoder. ",
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offset[1], clipped_offset);
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}
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}
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sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100];
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sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
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}
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}
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} else {
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} else {
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for (; i < run_end; i++, idx++) {
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for (; i < run_end; i++, idx++) {
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@ -832,7 +830,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
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"%s (%d) out of range.\n", sf_str[0], offset[0]);
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"%s (%d) out of range.\n", sf_str[0], offset[0]);
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return -1;
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return -1;
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}
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}
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sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
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sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
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}
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}
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}
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}
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}
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}
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@ -1243,7 +1241,6 @@ static av_always_inline float flt16_trunc(float pf)
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}
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}
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|
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static av_always_inline void predict(PredictorState *ps, float *coef,
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static av_always_inline void predict(PredictorState *ps, float *coef,
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float sf_scale, float inv_sf_scale,
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int output_enable)
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int output_enable)
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{
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{
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const float a = 0.953125; // 61.0 / 64
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const float a = 0.953125; // 61.0 / 64
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@ -1260,9 +1257,9 @@ static av_always_inline void predict(PredictorState *ps, float *coef,
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|
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pv = flt16_round(k1 * r0 + k2 * r1);
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pv = flt16_round(k1 * r0 + k2 * r1);
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if (output_enable)
|
if (output_enable)
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*coef += pv * sf_scale;
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*coef += pv;
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|
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e0 = *coef * inv_sf_scale;
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e0 = *coef;
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e1 = e0 - k1 * r0;
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e1 = e0 - k1 * r0;
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|
|
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ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
|
ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
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@ -1280,7 +1277,6 @@ static av_always_inline void predict(PredictorState *ps, float *coef,
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static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
|
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
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{
|
{
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int sfb, k;
|
int sfb, k;
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float sf_scale = ac->sf_scale, inv_sf_scale = 1 / ac->sf_scale;
|
|
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|
|
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if (!sce->ics.predictor_initialized) {
|
if (!sce->ics.predictor_initialized) {
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reset_all_predictors(sce->predictor_state);
|
reset_all_predictors(sce->predictor_state);
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@ -1291,7 +1287,6 @@ static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
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for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
|
for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
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for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
|
for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
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predict(&sce->predictor_state[k], &sce->coeffs[k],
|
predict(&sce->predictor_state[k], &sce->coeffs[k],
|
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sf_scale, inv_sf_scale,
|
|
||||||
sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
|
sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
|
@ -36,11 +36,11 @@
|
|||||||
#include <stdint.h>
|
#include <stdint.h>
|
||||||
|
|
||||||
/* @name ltp_coef
|
/* @name ltp_coef
|
||||||
* Table of the LTP coefficient (multiplied by 2)
|
* Table of the LTP coefficients
|
||||||
*/
|
*/
|
||||||
static const float ltp_coef[8] = {
|
static const float ltp_coef[8] = {
|
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1.141658, 1.393232, 1.626008, 1.822608,
|
0.570829, 0.696616, 0.813004, 0.911304,
|
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1.969800, 2.135788, 2.2389202, 2.739066,
|
0.984900, 1.067894, 1.194601, 1.369533,
|
||||||
};
|
};
|
||||||
|
|
||||||
/* @name tns_tmp2_map
|
/* @name tns_tmp2_map
|
||||||
|
@ -1962,8 +1962,6 @@ static av_cold int set_channel_info(AC3EncodeContext *s, int channels,
|
|||||||
ch_layout = *channel_layout;
|
ch_layout = *channel_layout;
|
||||||
if (!ch_layout)
|
if (!ch_layout)
|
||||||
ch_layout = avcodec_guess_channel_layout(channels, CODEC_ID_AC3, NULL);
|
ch_layout = avcodec_guess_channel_layout(channels, CODEC_ID_AC3, NULL);
|
||||||
if (av_get_channel_layout_nb_channels(ch_layout) != channels)
|
|
||||||
return AVERROR(EINVAL);
|
|
||||||
|
|
||||||
s->lfe_on = !!(ch_layout & AV_CH_LOW_FREQUENCY);
|
s->lfe_on = !!(ch_layout & AV_CH_LOW_FREQUENCY);
|
||||||
s->channels = channels;
|
s->channels = channels;
|
||||||
|
@ -33,13 +33,13 @@
|
|||||||
// cos(i * M_PI / 16) * sqrt(2) * (1 << 14)
|
// cos(i * M_PI / 16) * sqrt(2) * (1 << 14)
|
||||||
// W4 is actually exactly 16384, but using 16383 works around
|
// W4 is actually exactly 16384, but using 16383 works around
|
||||||
// accumulating rounding errors for some encoders
|
// accumulating rounding errors for some encoders
|
||||||
#define W1 ((int_fast32_t) 22725)
|
#define W1 22725
|
||||||
#define W2 ((int_fast32_t) 21407)
|
#define W2 21407
|
||||||
#define W3 ((int_fast32_t) 19266)
|
#define W3 19266
|
||||||
#define W4 ((int_fast32_t) 16383)
|
#define W4 16383
|
||||||
#define W5 ((int_fast32_t) 12873)
|
#define W5 12873
|
||||||
#define W6 ((int_fast32_t) 8867)
|
#define W6 8867
|
||||||
#define W7 ((int_fast32_t) 4520)
|
#define W7 4520
|
||||||
#define ROW_SHIFT 11
|
#define ROW_SHIFT 11
|
||||||
#define COL_SHIFT 20
|
#define COL_SHIFT 20
|
||||||
|
|
||||||
|
@ -259,12 +259,9 @@ static int decode_frame(AVCodecContext *avctx,
|
|||||||
int delta;
|
int delta;
|
||||||
const uint8_t *pal = av_packet_get_side_data(avpkt, AV_PKT_DATA_PALETTE, NULL);
|
const uint8_t *pal = av_packet_get_side_data(avpkt, AV_PKT_DATA_PALETTE, NULL);
|
||||||
|
|
||||||
if(p->data[0])
|
p->reference = 3;
|
||||||
avctx->release_buffer(avctx, p);
|
if (avctx->reget_buffer(avctx, p) < 0) {
|
||||||
|
av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed\n");
|
||||||
p->reference= 0;
|
|
||||||
if(avctx->get_buffer(avctx, p) < 0){
|
|
||||||
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
|
|
||||||
return -1;
|
return -1;
|
||||||
}
|
}
|
||||||
outdata = a->pic.data[0];
|
outdata = a->pic.data[0];
|
||||||
|
@ -360,8 +360,6 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPac
|
|||||||
|
|
||||||
if(buf_size <= 769)
|
if(buf_size <= 769)
|
||||||
return 0;
|
return 0;
|
||||||
if(smk->pic.data[0])
|
|
||||||
avctx->release_buffer(avctx, &smk->pic);
|
|
||||||
|
|
||||||
smk->pic.reference = 1;
|
smk->pic.reference = 1;
|
||||||
smk->pic.buffer_hints = FF_BUFFER_HINTS_VALID | FF_BUFFER_HINTS_PRESERVE | FF_BUFFER_HINTS_REUSABLE;
|
smk->pic.buffer_hints = FF_BUFFER_HINTS_VALID | FF_BUFFER_HINTS_PRESERVE | FF_BUFFER_HINTS_REUSABLE;
|
||||||
|
@ -224,13 +224,10 @@ static int ulti_decode_frame(AVCodecContext *avctx,
|
|||||||
int skip;
|
int skip;
|
||||||
int tmp;
|
int tmp;
|
||||||
|
|
||||||
if(s->frame.data[0])
|
|
||||||
avctx->release_buffer(avctx, &s->frame);
|
|
||||||
|
|
||||||
s->frame.reference = 1;
|
s->frame.reference = 1;
|
||||||
s->frame.buffer_hints = FF_BUFFER_HINTS_VALID | FF_BUFFER_HINTS_PRESERVE | FF_BUFFER_HINTS_REUSABLE;
|
s->frame.buffer_hints = FF_BUFFER_HINTS_VALID | FF_BUFFER_HINTS_PRESERVE | FF_BUFFER_HINTS_REUSABLE;
|
||||||
if(avctx->get_buffer(avctx, &s->frame) < 0) {
|
if (avctx->reget_buffer(avctx, &s->frame) < 0) {
|
||||||
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
|
av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed\n");
|
||||||
return -1;
|
return -1;
|
||||||
}
|
}
|
||||||
|
|
||||||
|
@ -555,15 +555,50 @@ int attribute_align_arg avcodec_open(AVCodecContext *avctx, AVCodec *codec)
|
|||||||
ret = AVERROR(EINVAL);
|
ret = AVERROR(EINVAL);
|
||||||
goto free_and_end;
|
goto free_and_end;
|
||||||
}
|
}
|
||||||
if (avctx->codec->sample_fmts && avctx->codec->encode) {
|
if (avctx->codec->encode) {
|
||||||
int i;
|
int i;
|
||||||
for (i = 0; avctx->codec->sample_fmts[i] != AV_SAMPLE_FMT_NONE; i++)
|
if (avctx->codec->sample_fmts) {
|
||||||
if (avctx->sample_fmt == avctx->codec->sample_fmts[i])
|
for (i = 0; avctx->codec->sample_fmts[i] != AV_SAMPLE_FMT_NONE; i++)
|
||||||
break;
|
if (avctx->sample_fmt == avctx->codec->sample_fmts[i])
|
||||||
if (avctx->codec->sample_fmts[i] == AV_SAMPLE_FMT_NONE) {
|
break;
|
||||||
av_log(avctx, AV_LOG_ERROR, "Specified sample_fmt is not supported.\n");
|
if (avctx->codec->sample_fmts[i] == AV_SAMPLE_FMT_NONE) {
|
||||||
ret = AVERROR(EINVAL);
|
av_log(avctx, AV_LOG_ERROR, "Specified sample_fmt is not supported.\n");
|
||||||
goto free_and_end;
|
ret = AVERROR(EINVAL);
|
||||||
|
goto free_and_end;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
if (avctx->codec->supported_samplerates) {
|
||||||
|
for (i = 0; avctx->codec->supported_samplerates[i] != 0; i++)
|
||||||
|
if (avctx->sample_rate == avctx->codec->supported_samplerates[i])
|
||||||
|
break;
|
||||||
|
if (avctx->codec->supported_samplerates[i] == 0) {
|
||||||
|
av_log(avctx, AV_LOG_ERROR, "Specified sample_rate is not supported\n");
|
||||||
|
ret = AVERROR(EINVAL);
|
||||||
|
goto free_and_end;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
if (avctx->codec->channel_layouts) {
|
||||||
|
if (!avctx->channel_layout) {
|
||||||
|
av_log(avctx, AV_LOG_WARNING, "channel_layout not specified\n");
|
||||||
|
} else {
|
||||||
|
for (i = 0; avctx->codec->channel_layouts[i] != 0; i++)
|
||||||
|
if (avctx->channel_layout == avctx->codec->channel_layouts[i])
|
||||||
|
break;
|
||||||
|
if (avctx->codec->channel_layouts[i] == 0) {
|
||||||
|
av_log(avctx, AV_LOG_ERROR, "Specified channel_layout is not supported\n");
|
||||||
|
ret = AVERROR(EINVAL);
|
||||||
|
goto free_and_end;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
}
|
||||||
|
if (avctx->channel_layout && avctx->channels) {
|
||||||
|
if (av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels) {
|
||||||
|
av_log(avctx, AV_LOG_ERROR, "channel layout does not match number of channels\n");
|
||||||
|
ret = AVERROR(EINVAL);
|
||||||
|
goto free_and_end;
|
||||||
|
}
|
||||||
|
} else if (avctx->channel_layout) {
|
||||||
|
avctx->channels = av_get_channel_layout_nb_channels(avctx->channel_layout);
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
|
|
||||||
@ -1194,11 +1229,9 @@ void av_log_missing_feature(void *avc, const char *feature, int want_sample)
|
|||||||
av_log(avc, AV_LOG_WARNING, "%s not implemented. Update your FFmpeg "
|
av_log(avc, AV_LOG_WARNING, "%s not implemented. Update your FFmpeg "
|
||||||
"version to the newest one from Git. If the problem still "
|
"version to the newest one from Git. If the problem still "
|
||||||
"occurs, it means that your file has a feature which has not "
|
"occurs, it means that your file has a feature which has not "
|
||||||
"been implemented.", feature);
|
"been implemented.\n", feature);
|
||||||
if(want_sample)
|
if(want_sample)
|
||||||
av_log_ask_for_sample(avc, NULL);
|
av_log_ask_for_sample(avc, NULL);
|
||||||
else
|
|
||||||
av_log(avc, AV_LOG_WARNING, "\n");
|
|
||||||
}
|
}
|
||||||
|
|
||||||
void av_log_ask_for_sample(void *avc, const char *msg, ...)
|
void av_log_ask_for_sample(void *avc, const char *msg, ...)
|
||||||
|
@ -1138,7 +1138,7 @@ static int vorbis_floor1_decode(vorbis_context *vc,
|
|||||||
uint_fast16_t floor1_Y[258];
|
uint_fast16_t floor1_Y[258];
|
||||||
uint_fast16_t floor1_Y_final[258];
|
uint_fast16_t floor1_Y_final[258];
|
||||||
int floor1_flag[258];
|
int floor1_flag[258];
|
||||||
uint_fast8_t class_;
|
uint_fast8_t class;
|
||||||
uint_fast8_t cdim;
|
uint_fast8_t cdim;
|
||||||
uint_fast8_t cbits;
|
uint_fast8_t cbits;
|
||||||
uint_fast8_t csub;
|
uint_fast8_t csub;
|
||||||
@ -1162,20 +1162,20 @@ static int vorbis_floor1_decode(vorbis_context *vc,
|
|||||||
|
|
||||||
offset = 2;
|
offset = 2;
|
||||||
for (i = 0; i < vf->partitions; ++i) {
|
for (i = 0; i < vf->partitions; ++i) {
|
||||||
class_ = vf->partition_class[i];
|
class = vf->partition_class[i];
|
||||||
cdim = vf->class_dimensions[class_];
|
cdim = vf->class_dimensions[class];
|
||||||
cbits = vf->class_subclasses[class_];
|
cbits = vf->class_subclasses[class];
|
||||||
csub = (1 << cbits) - 1;
|
csub = (1 << cbits) - 1;
|
||||||
cval = 0;
|
cval = 0;
|
||||||
|
|
||||||
AV_DEBUG("Cbits %d \n", cbits);
|
AV_DEBUG("Cbits %d \n", cbits);
|
||||||
|
|
||||||
if (cbits) // this reads all subclasses for this partition's class
|
if (cbits) // this reads all subclasses for this partition's class
|
||||||
cval = get_vlc2(gb, vc->codebooks[vf->class_masterbook[class_]].vlc.table,
|
cval = get_vlc2(gb, vc->codebooks[vf->class_masterbook[class]].vlc.table,
|
||||||
vc->codebooks[vf->class_masterbook[class_]].nb_bits, 3);
|
vc->codebooks[vf->class_masterbook[class]].nb_bits, 3);
|
||||||
|
|
||||||
for (j = 0; j < cdim; ++j) {
|
for (j = 0; j < cdim; ++j) {
|
||||||
book = vf->subclass_books[class_][cval & csub];
|
book = vf->subclass_books[class][cval & csub];
|
||||||
|
|
||||||
AV_DEBUG("book %d Cbits %d cval %d bits:%d \n", book, cbits, cval, get_bits_count(gb));
|
AV_DEBUG("book %d Cbits %d cval %d bits:%d \n", book, cbits, cval, get_bits_count(gb));
|
||||||
|
|
||||||
|
Loading…
Reference in New Issue
Block a user