July 2008 - FFmpeg-soc - lists.ffmpeg.org

[soc]: r2920 - mxf/mxfenc.c
by spyfeng 30 Jul '08

30 Jul '08

Author: spyfeng Date: Wed Jul 30 16:11:23 2008 New Revision: 2920 Log: fix mistake of former patch Modified: mxf/mxfenc.c Modified: mxf/mxfenc.c ============================================================================== --- mxf/mxfenc.c (original) +++ mxf/mxfenc.c Wed Jul 30 16:11:23 2008 @@ -235,7 +235,7 @@ static void mxf_generate_uuid(AVFormatCo } // the 7th byte is version according to ISO 11578 uuid[6] &= 0x0f; - uuid[6] &= 0x40; + uuid[6] |= 0x40; // the 8th byte is variant for current use according to ISO 11578 uuid[8] &= 0x3f;

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[soc]: r2919 - aac/aac.c
by superdump 30 Jul '08

30 Jul '08

Author: superdump Date: Wed Jul 30 14:41:34 2008 New Revision: 2919 Log: Add doxygen comment for apply_coupling_method function pointer argument of apply_channel_coupling() Modified: aac/aac.c Modified: aac/aac.c ============================================================================== --- aac/aac.c (original) +++ aac/aac.c Wed Jul 30 14:41:34 2008 @@ -2085,6 +2085,7 @@ static void independent_coupling(AACCont * channel coupling transformation interface * * @param index which gain to use for coupling + * @param apply_coupling_method pointer to (in)dependent coupling function */ static void apply_channel_coupling(AACContext * ac, ChannelElement * cc, void (*apply_coupling_method)(AACContext * ac, ChannelElement * cc, SingleChannelElement * sce, int index))

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[soc]: r2917 - aac/aac.c
by superdump 30 Jul '08

30 Jul '08

Author: superdump Date: Wed Jul 30 14:25:06 2008 New Revision: 2917 Log: Use window_sequence[2] instead of window_sequence and window_sequence_prev Modified: aac/aac.c Modified: aac/aac.c ============================================================================== --- aac/aac.c (original) +++ aac/aac.c Wed Jul 30 14:25:06 2008 @@ -236,8 +236,7 @@ typedef struct { typedef struct { int intensity_present; uint8_t max_sfb; ///< number of scalefactor bands per group - enum WindowSequence window_sequence; - enum WindowSequence window_sequence_prev; + enum WindowSequence window_sequence[2]; uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window, otherwise use a sinus window. int num_window_groups; uint8_t group_len[8]; @@ -972,13 +971,13 @@ static int decode_ics_info(AACContext * av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n"); return -1; } - ics->window_sequence_prev = ics->window_sequence; - ics->window_sequence = get_bits(gb, 2); + ics->window_sequence[1] = ics->window_sequence[0]; + ics->window_sequence[0] = get_bits(gb, 2); ics->use_kb_window[1] = ics->use_kb_window[0]; ics->use_kb_window[0] = get_bits1(gb); ics->num_window_groups = 1; ics->group_len[0] = 1; - if (ics->window_sequence == EIGHT_SHORT_SEQUENCE) { + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { int i; ics->max_sfb = get_bits(gb, 4); grouping = get_bits(gb, 7); @@ -1005,7 +1004,7 @@ static int decode_ics_info(AACContext * return -1; } - if (ics->window_sequence == EIGHT_SHORT_SEQUENCE) { + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { ics->num_windows = 8; ics->tns_max_bands = tns_max_bands_128[ac->m4ac.sampling_index]; } else { @@ -1062,7 +1061,7 @@ static inline float ivquant(AACContext * */ static int decode_band_types(AACContext * ac, GetBitContext * gb, IndividualChannelStream * ics, enum BandType band_type[][64], int band_type_run_end[][64]) { int g; - const int bits = (ics->window_sequence == EIGHT_SHORT_SEQUENCE) ? 3 : 5; + const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; for (g = 0; g < ics->num_window_groups; g++) { int k = 0; while (k < ics->max_sfb) { @@ -1103,7 +1102,7 @@ static int decode_band_types(AACContext */ static int decode_scalefactors(AACContext * ac, GetBitContext * gb, float mix_gain, unsigned int global_gain, IndividualChannelStream * ics, const enum BandType band_type[][64], const int band_type_run_end[][64], float sf[][64]) { - const int sf_offset = ac->sf_offset + (ics->window_sequence == EIGHT_SHORT_SEQUENCE ? 12 : 0); + const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); int g, i; int offset[3] = { global_gain, global_gain - 90, 100 }; int noise_flag = 1; @@ -1176,7 +1175,7 @@ static void decode_pulses(AACContext * a */ static void decode_tns(AACContext * ac, GetBitContext * gb, const IndividualChannelStream * ics, TemporalNoiseShaping * tns) { int w, filt, i, coef_len, coef_res = 0, coef_compress; - const int is8 = ics->window_sequence == EIGHT_SHORT_SEQUENCE; + const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; for (w = 0; w < ics->num_windows; w++) { tns->n_filt[w] = get_bits(gb, 2 - is8); @@ -1208,7 +1207,7 @@ static int decode_gain_control(AACContex {8, 0, 2}, //EIGHT_SHORT_SEQUENCE, {2, 1, 5}, //LONG_STOP_SEQUENCE }; - const int mode = sce->ics.window_sequence; + const int mode = sce->ics.window_sequence[0]; int bd, wd, ad; ScalableSamplingRate * ssr = sce->ssr; if (!ssr && !(ssr = sce->ssr = av_mallocz(sizeof(ScalableSamplingRate)))) @@ -1392,7 +1391,7 @@ static int decode_ics(AACContext * ac, G pulse_present = 0; if (!scale_flag) { if ((pulse_present = get_bits1(gb))) { - if (ics->window_sequence == EIGHT_SHORT_SEQUENCE) { + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n"); return -1; } @@ -1781,15 +1780,15 @@ static void windowing_and_mdct_ltp(AACCo const float * lwindow_prev = ics->use_kb_window[1] ? kbd_long_1024 : sine_long_1024; const float * swindow_prev = ics->use_kb_window[1] ? kbd_short_128 : sine_short_128; float * buf = ac->buf_mdct; - assert(ics->window_sequence != EIGHT_SHORT_SEQUENCE); - if (ics->window_sequence != LONG_STOP_SEQUENCE) { + assert(ics->window_sequence[0] != EIGHT_SHORT_SEQUENCE); + if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) { vector_fmul_dst(ac, buf, in, lwindow_prev, 1024); } else { memset(buf, 0, 448 * sizeof(float)); vector_fmul_dst(ac, buf + 448, in + 448, swindow_prev, 128); memcpy(buf + 576, in + 576, 448 * sizeof(float)); } - if (ics->window_sequence != LONG_START_SEQUENCE) { + if (ics->window_sequence[0] != LONG_START_SEQUENCE) { ac->dsp.vector_fmul_reverse(buf + 1024, in + 1024, lwindow, 1024); } else { memcpy(buf + 1024, in + 1024, 448 * sizeof(float)); @@ -1807,7 +1806,7 @@ static int apply_ltp(AACContext * ac, Si return 0; if (!sce->ltp_state && !(sce->ltp_state = av_mallocz(4 * 1024 * sizeof(int16_t)))) return AVERROR(ENOMEM); - if (sce->ics.window_sequence != EIGHT_SHORT_SEQUENCE && ac->is_saved) { + if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE && ac->is_saved) { float x_est[2 * 1024], X_est[2 * 1024]; for (i = 0; i < 2 * 1024; i++) x_est[i] = sce->ltp_state[i + 2 * 1024 - ltp->lag] * ltp->coef; @@ -1872,16 +1871,16 @@ static void imdct_and_windowing(AACConte float * buf = ac->buf_mdct; int i; - if (ics->window_sequence != EIGHT_SHORT_SEQUENCE) { + if (ics->window_sequence[0] != EIGHT_SHORT_SEQUENCE) { ff_imdct_calc(&ac->mdct, buf, in, out); // Out can be abused, for now, as a temp buffer. - if (ics->window_sequence != LONG_STOP_SEQUENCE) { + if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) { ac->dsp.vector_fmul_add_add(out, buf, lwindow_prev, saved, ac->add_bias, 1024, 1); } else { for (i = 0; i < 448; i++) out[i] = saved[i] + ac->add_bias; ac->dsp.vector_fmul_add_add(out + 448, buf + 448, swindow_prev, saved + 448, ac->add_bias, 128, 1); for (i = 576; i < 1024; i++) out[i] = buf[i] + saved[i] + ac->add_bias; } - if (ics->window_sequence != LONG_START_SEQUENCE) { + if (ics->window_sequence[0] != LONG_START_SEQUENCE) { ac->dsp.vector_fmul_reverse(saved, buf + 1024, lwindow, 1024); } else { memcpy(saved, buf + 1024, 448 * sizeof(float)); @@ -1889,7 +1888,7 @@ static void imdct_and_windowing(AACConte memset(saved + 576, 0, 448 * sizeof(float)); } } else { - if (ics->window_sequence_prev == ONLY_LONG_SEQUENCE || ics->window_sequence_prev == LONG_STOP_SEQUENCE) + if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) av_log(ac->avccontext, AV_LOG_WARNING, "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. " "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n"); @@ -1929,16 +1928,16 @@ static void windowing_and_imdct_ssr(AACC const float * lwindow_prev = ics->use_kb_window[1] ? kbd_long_1024 : sine_long_1024; const float * swindow_prev = ics->use_kb_window[1] ? kbd_short_128 : sine_short_128; float * buf = ac->buf_mdct; - if (ics->window_sequence != EIGHT_SHORT_SEQUENCE) { + if (ics->window_sequence[0] != EIGHT_SHORT_SEQUENCE) { ff_imdct_calc(&ac->mdct, buf, in, out); - if (ics->window_sequence != LONG_STOP_SEQUENCE) { + if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) { vector_fmul_dst(ac, out, buf, lwindow_prev, 256); } else { memset(out, 0, 112 * sizeof(float)); vector_fmul_dst(ac, out + 112, buf + 112, swindow_prev, 32); memcpy(out + 144, buf + 144, 112 * sizeof(float)); } - if (ics->window_sequence != LONG_START_SEQUENCE) { + if (ics->window_sequence[0] != LONG_START_SEQUENCE) { ac->dsp.vector_fmul_reverse(out + 256, buf + 256, lwindow, 256); } else { memcpy(out + 256, buf + 256, 112 * sizeof(float)); @@ -1963,7 +1962,7 @@ static void vector_add_dst(AACContext * static void apply_ssr_gains(AACContext * ac, SingleChannelElement * sce, int band, float * in, float * preret, float * saved) { // TODO: 'in' buffer gain normalization - if (sce->ics.window_sequence != EIGHT_SHORT_SEQUENCE) { + if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { vector_add_dst(ac, preret, in, saved, 256); memcpy(saved, in + 256, 256 * sizeof(float)); } else {

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[soc]: r2916 - aac/aac.c
by superdump 30 Jul '08

30 Jul '08

Author: superdump Date: Wed Jul 30 13:56:58 2008 New Revision: 2916 Log: Add doxygen comments to DynamicRangeControl struct Modified: aac/aac.c Modified: aac/aac.c ============================================================================== --- aac/aac.c (original) +++ aac/aac.c Wed Jul 30 13:56:58 2008 @@ -276,15 +276,18 @@ typedef struct { * Dynamic Range Control - decoded from the bitstream but not processed further. */ typedef struct { - int pce_instance_tag; - int dyn_rng_sgn[17]; - int dyn_rng_ctl[17]; - int exclude_mask[MAX_CHANNELS]; - int additional_excluded_chns[MAX_CHANNELS]; - int band_incr; - int interpolation_scheme; - int band_top[17]; - int prog_ref_level; + int pce_instance_tag; ///< Indicates with which program the DRC info is associated. + int dyn_rng_sgn[17]; ///< DRC sign information; 0 - positive, 1 - negative + int dyn_rng_ctl[17]; ///< DRC magnitude information + int exclude_mask[MAX_CHANNELS]; ///< Channels to be excluded from DRC processing. + int additional_excluded_chns[MAX_CHANNELS]; /**< The exclude_mask bits are + coded in groups of 7 with 1 bit preceeding each group (except the first) + indicating that 7 more mask bits are coded. */ + int band_incr; ///< Number of DRC bands greater than 1 having DRC info. + int interpolation_scheme; ///< Indicates the interpolation scheme used in the SBR QMF domain. + int band_top[17]; ///< Indicates the top of the i-th DRC band in units of 4 spectral lines. + int prog_ref_level; /**< A reference level for the long-term program audio level for all + channels combined. */ } DynamicRangeControl; /**

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[soc]: r2915 - aac/aac.c
by superdump 30 Jul '08

30 Jul '08

Author: superdump Date: Wed Jul 30 13:55:18 2008 New Revision: 2915 Log: Add appropriate error message for output buffer size versus desired size check Modified: aac/aac.c Modified: aac/aac.c ============================================================================== --- aac/aac.c (original) +++ aac/aac.c Wed Jul 30 13:55:18 2008 @@ -2189,8 +2189,12 @@ static int output_samples(AVCodecContext } i = 1024 * avccontext->channels * sizeof(int16_t); - if(*data_size < i) + if(*data_size < i) { + av_log(avccontext, AV_LOG_ERROR, + "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", + *data_size, i); return -1; + } *data_size = i; if(ac->mm[MIXDOWN_CENTER]) {

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[soc]: r2914 - aacenc/aacpsy.c
by kostya 30 Jul '08

30 Jul '08

Author: kostya Date: Wed Jul 30 13:50:16 2008 New Revision: 2914 Log: Blockswitching for 3gpp psy model Modified: aacenc/aacpsy.c Modified: aacenc/aacpsy.c ============================================================================== --- aacenc/aacpsy.c (original) +++ aacenc/aacpsy.c Wed Jul 30 13:50:16 2008 @@ -332,6 +332,11 @@ typedef struct Psy3gppChannel{ float thr[2]; Psy3gppBand band[2][128]; Psy3gppBand prev_band[2][128]; + + float win_nrg[2]; + float iir_state[2][2]; + uint8_t next_grouping[2]; + enum WindowSequence next_window_seq[2]; }Psy3gppChannel; /** @@ -430,6 +435,26 @@ static av_cold int psy_3gpp_init(AACPsyC } /** + * IIR filter used in block switching decision + */ +static float iir_filter(int in, float state[2]) +{ + float ret; + + ret = 0.7548f * (in - state[0]) + 0.5095f * state[1]; + state[0] = in; + state[1] = ret; + return ret; +} + +/** + * window grouping information stored as bits (0 - new group, 1 - group continues) + */ +static const uint8_t window_grouping[9] = { + 0xB6, 0x6C, 0xD8, 0xB2, 0x66, 0xC6, 0x96, 0x36, 0x36 +}; + +/** * Tell encoder which window types to use. * @see 3GPP TS26.403 5.4.1 */ @@ -437,17 +462,83 @@ static void psy_3gpp_window(AACPsyContex { int ch; int chans = FFMIN(apc->avctx->channels - channel, 2); + int i, j; + int br = apc->avctx->bit_rate / apc->avctx->channels; + int attack_ratio = (br <= 16000 + 8000*chans) ? 18 : 10; + Psy3gppContext *pctx = (Psy3gppContext*) apc->model_priv_data; + Psy3gppChannel *pch = &pctx->ch; + uint8_t grouping[2]; + enum WindowSequence win[2]; + + if(la){ + float s[8], v; + for(ch = 0; ch < chans; ch++){ + enum WindowSequence last_window_sequence = cpe->ch[ch].ics.window_sequence; + int switch_to_eight = 0; + float sum = 0.0, sum2 = 0.0; + int attack_n = 0; + for(i = 0; i < 8; i++){ + for(j = 0; j < 128; j++){ + v = iir_filter(audio[(i*128+j)*apc->avctx->channels+ch], pch->iir_state[ch]); + sum += v*v; + } + s[i] = sum; + sum2 += sum; + } + for(i = 0; i < 8; i++){ + if(s[i] > pch->win_nrg[ch] * attack_ratio){ + attack_n = i + 1; + switch_to_eight = 1; + break; + } + } + pch->win_nrg[ch] = pch->win_nrg[ch]*7/8 + sum2/64; + + switch(last_window_sequence){ + case ONLY_LONG_SEQUENCE: + win[ch] = switch_to_eight ? LONG_START_SEQUENCE : ONLY_LONG_SEQUENCE; + grouping[ch] = 0; + break; + case LONG_START_SEQUENCE: + win[ch] = EIGHT_SHORT_SEQUENCE; + grouping[ch] = pch->next_grouping[ch]; + break; + case LONG_STOP_SEQUENCE: + win[ch] = ONLY_LONG_SEQUENCE; + grouping[ch] = 0; + break; + case EIGHT_SHORT_SEQUENCE: + win[ch] = switch_to_eight ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE; + grouping[ch] = switch_to_eight ? pch->next_grouping[ch] : 0; + break; + } + pch->next_grouping[ch] = window_grouping[attack_n]; + } + }else{ + for(ch = 0; ch < chans; ch++){ + win[ch] = (cpe->ch[ch].ics.window_sequence == EIGHT_SHORT_SEQUENCE) ? EIGHT_SHORT_SEQUENCE : ONLY_LONG_SEQUENCE; + grouping[ch] = (cpe->ch[ch].ics.window_sequence == EIGHT_SHORT_SEQUENCE) ? window_grouping[0] : 0; + } + } -//XXX: stub, because encoder does not support long to short window transition yet :( for(ch = 0; ch < chans; ch++){ - cpe->ch[ch].ics.window_sequence = ONLY_LONG_SEQUENCE; + cpe->ch[ch].ics.window_sequence = win[ch]; cpe->ch[ch].ics.use_kb_window[0] = 1; - cpe->ch[ch].ics.num_windows = 1; - cpe->ch[ch].ics.swb_sizes = apc->bands1024; - cpe->ch[ch].ics.num_swb = apc->num_bands1024; - cpe->ch[ch].ics.group_len[0] = 0; + if(win[ch] != EIGHT_SHORT_SEQUENCE){ + cpe->ch[ch].ics.num_windows = 1; + cpe->ch[ch].ics.swb_sizes = apc->bands1024; + cpe->ch[ch].ics.num_swb = apc->num_bands1024; + }else{ + cpe->ch[ch].ics.num_windows = 8; + cpe->ch[ch].ics.swb_sizes = apc->bands128; + cpe->ch[ch].ics.num_swb = apc->num_bands128; + } + for(i = 0; i < 8; i++) + cpe->ch[ch].ics.group_len[i] = (grouping[ch] >> i) & 1; } - cpe->common_window = cpe->ch[0].ics.use_kb_window[0] == cpe->ch[1].ics.use_kb_window[0]; + cpe->common_window = chans > 1 && cpe->ch[0].ics.window_sequence == cpe->ch[1].ics.window_sequence && cpe->ch[0].ics.use_kb_window[0] == cpe->ch[1].ics.use_kb_window[0]; + if(cpe->common_window && cpe->ch[0].ics.window_sequence == EIGHT_SHORT_SEQUENCE && grouping[0] != grouping[1]) + cpe->common_window = 0; } /** @@ -489,7 +580,7 @@ static void calc_pe(Psy3gppBand *band, i static void psy_3gpp_process(AACPsyContext *apc, int channel, ChannelElement *cpe) { int start; - int ch, w, g, g2, i; + int ch, w, w2, g, g2, i; int prev_scale; Psy3gppContext *pctx = (Psy3gppContext*) apc->model_priv_data; float stereo_att, pe_target; @@ -674,6 +765,24 @@ static void psy_3gpp_process(AACPsyConte if(!cpe->ch[ch].gain) cpe->ch[ch].gain = cpe->ch[ch].sf_idx[w][g]; } } + //adjust scalefactors for window groups + for(w = 0; w < cpe->ch[ch].ics.num_windows - 1; w++){ + int min_scale = 256; + + if(cpe->ch[ch].ics.group_len[w]) continue; + w2 = w; + do{ + w2++; + }while(w2 < cpe->ch[ch].ics.num_windows && cpe->ch[ch].ics.group_len[w2]); + for(g = 0; g < cpe->ch[ch].ics.num_swb; g++){ + for(i = w; i < w2; i++){ + if(cpe->ch[ch].zeroes[i][g]) continue; + min_scale = FFMIN(min_scale, cpe->ch[ch].sf_idx[i][g]); + } + for(i = w; i < w2; i++) + cpe->ch[ch].sf_idx[i][g] = min_scale; + } + } } memcpy(pch->prev_band, pch->band, sizeof(pch->band));

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[soc]: r2913 - aac/aac.c
by superdump 30 Jul '08

30 Jul '08

Author: superdump Date: Wed Jul 30 13:47:52 2008 New Revision: 2913 Log: Alignment Modified: aac/aac.c Modified: aac/aac.c ============================================================================== --- aac/aac.c (original) +++ aac/aac.c Wed Jul 30 13:47:52 2008 @@ -2134,7 +2134,7 @@ static int spectral_to_sample(AACContext return ret; } #endif /* AAC_LTP */ - if( che->ch[0].tns.present) + if(che->ch[0].tns.present) apply_tns(ac, 1, &che->ch[0], che->ch[0].coeffs); if(che->ch[1].tns.present) apply_tns(ac, 1, &che->ch[1], che->ch[1].coeffs);

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[soc]: r2912 - aac/aac.c
by superdump 30 Jul '08

30 Jul '08

Author: superdump Date: Wed Jul 30 13:47:37 2008 New Revision: 2912 Log: che->ch[1].tns.present is only ever set when the che is a CPE anyway so this extra check is redundant Modified: aac/aac.c Modified: aac/aac.c ============================================================================== --- aac/aac.c (original) +++ aac/aac.c Wed Jul 30 13:47:37 2008 @@ -2136,7 +2136,7 @@ static int spectral_to_sample(AACContext #endif /* AAC_LTP */ if( che->ch[0].tns.present) apply_tns(ac, 1, &che->ch[0], che->ch[0].coeffs); - if(j == ID_CPE && che->ch[1].tns.present) + if(che->ch[1].tns.present) apply_tns(ac, 1, &che->ch[1], che->ch[1].coeffs); if(j == ID_CCE && !che->coup.is_indep_coup && (che->coup.domain == 1)) apply_channel_coupling(ac, che, dependent_coupling);

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[soc]: r2911 - aac/aac.c
by superdump 30 Jul '08

30 Jul '08

Author: superdump Date: Wed Jul 30 13:46:15 2008 New Revision: 2911 Log: Add more information about the mix-down channel types to the enum comment Modified: aac/aac.c Modified: aac/aac.c ============================================================================== --- aac/aac.c (original) +++ aac/aac.c Wed Jul 30 13:46:15 2008 @@ -196,6 +196,8 @@ enum ChannelType { /** * mix-down channel types + * MIXDOWN_CENTER is the index into the mix-down arrays for a Single Channel Element with AAC_CHANNEL_FRONT. + * MIXDOWN_(BACK|FRONT) are the indices for Channel Pair Elements with AAC_CHANNEL_(BACK|FRONT). */ enum { MIXDOWN_CENTER,

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[soc]: r2910 - aacenc/aacpsy.c
by kostya 30 Jul '08

30 Jul '08

Author: kostya Date: Wed Jul 30 13:43:20 2008 New Revision: 2910 Log: Do not try to merge coefficients from differently windowed MDCTs Modified: aacenc/aacpsy.c Modified: aacenc/aacpsy.c ============================================================================== --- aacenc/aacpsy.c (original) +++ aacenc/aacpsy.c Wed Jul 30 13:43:20 2008 @@ -498,7 +498,7 @@ static void psy_3gpp_process(AACPsyConte Psy3gppChannel *pch = &pctx->ch; //calculate and apply stereo attenuation factor - 5.2 - if(apc->avctx->channels - channel > 1){ + if(chans > 1 && cpe->common_window){ float l, r; stereo_att = 1.0 / 2.0; //XXX: find some way to determine it for(i = 0; i < 1024; i++){

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