August 2009 - ffmpeg-cvslog - lists.ffmpeg.org

r19562 - trunk/libavcodec/aac.c
by subversion＠mplayerhq.hu 02 Aug '09

02 Aug '09

Author: diego Date: Sun Aug 2 12:50:59 2009 New Revision: 19562 Log: cosmetics: K&R coding style Modified: trunk/libavcodec/aac.c Modified: trunk/libavcodec/aac.c ============================================================================== --- trunk/libavcodec/aac.c Sun Aug 2 12:34:30 2009 (r19561) +++ trunk/libavcodec/aac.c Sun Aug 2 12:50:59 2009 (r19562) @@ -93,13 +93,17 @@ #include <math.h> #include <string.h> -union float754 { float f; uint32_t i; }; +union float754 { + float f; + uint32_t i; +}; static VLC vlc_scalefactors; static VLC vlc_spectral[11]; -static ChannelElement* get_che(AACContext *ac, int type, int elem_id) { +static ChannelElement *get_che(AACContext *ac, int type, int elem_id) +{ static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0 }; if (ac->tag_che_map[type][elem_id]) { return ac->tag_che_map[type][elem_id]; @@ -108,44 +112,44 @@ static ChannelElement* get_che(AACContex return NULL; } switch (ac->m4ac.chan_config) { - case 7: - if (ac->tags_mapped == 3 && type == TYPE_CPE) { - ac->tags_mapped++; - return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2]; - } - case 6: - /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1] - instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have - encountered such a stream, transfer the LFE[0] element to SCE[1] */ - if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) { - ac->tags_mapped++; - return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0]; - } - case 5: - if (ac->tags_mapped == 2 && type == TYPE_CPE) { - ac->tags_mapped++; - return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1]; - } - case 4: - if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) { - ac->tags_mapped++; - return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1]; - } - case 3: - case 2: - if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) { - ac->tags_mapped++; - return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0]; - } else if (ac->m4ac.chan_config == 2) { - return NULL; - } - case 1: - if (!ac->tags_mapped && type == TYPE_SCE) { - ac->tags_mapped++; - return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0]; - } - default: + case 7: + if (ac->tags_mapped == 3 && type == TYPE_CPE) { + ac->tags_mapped++; + return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2]; + } + case 6: + /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1] + instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have + encountered such a stream, transfer the LFE[0] element to SCE[1] */ + if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) { + ac->tags_mapped++; + return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0]; + } + case 5: + if (ac->tags_mapped == 2 && type == TYPE_CPE) { + ac->tags_mapped++; + return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1]; + } + case 4: + if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) { + ac->tags_mapped++; + return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1]; + } + case 3: + case 2: + if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) { + ac->tags_mapped++; + return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0]; + } else if (ac->m4ac.chan_config == 2) { return NULL; + } + case 1: + if (!ac->tags_mapped && type == TYPE_SCE) { + ac->tags_mapped++; + return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0]; + } + default: + return NULL; } } @@ -157,8 +161,11 @@ static ChannelElement* get_che(AACContex * * @return Returns error status. 0 - OK, !0 - error */ -static int output_configure(AACContext *ac, enum ChannelPosition che_pos[4][MAX_ELEM_ID], - enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], int channel_config) { +static int output_configure(AACContext *ac, + enum ChannelPosition che_pos[4][MAX_ELEM_ID], + enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], + int channel_config) +{ AVCodecContext *avctx = ac->avccontext; int i, type, channels = 0; @@ -173,14 +180,14 @@ static int output_configure(AACContext * * [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ] */ - for(i = 0; i < MAX_ELEM_ID; i++) { - for(type = 0; type < 4; type++) { - if(che_pos[type][i]) { - if(!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement)))) + for (i = 0; i < MAX_ELEM_ID; i++) { + for (type = 0; type < 4; type++) { + if (che_pos[type][i]) { + if (!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement)))) return AVERROR(ENOMEM); - if(type != TYPE_CCE) { + if (type != TYPE_CCE) { ac->output_data[channels++] = ac->che[type][i]->ch[0].ret; - if(type == TYPE_CPE) { + if (type == TYPE_CPE) { ac->output_data[channels++] = ac->che[type][i]->ch[1].ret; } } @@ -194,7 +201,7 @@ static int output_configure(AACContext * ac->tags_mapped = 0; } else { memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); - ac->tags_mapped = 4*MAX_ELEM_ID; + ac->tags_mapped = 4 * MAX_ELEM_ID; } avctx->channels = channels; @@ -212,8 +219,11 @@ static int output_configure(AACContext * * @param type speaker type/position for these channels */ static void decode_channel_map(enum ChannelPosition *cpe_map, - enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) { - while(n--) { + enum ChannelPosition *sce_map, + enum ChannelPosition type, + GetBitContext *gb, int n) +{ + while (n--) { enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map map[get_bits(gb, 4)] = type; } @@ -226,8 +236,9 @@ static void decode_channel_map(enum Chan * * @return Returns error status. 0 - OK, !0 - error */ -static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], - GetBitContext * gb) { +static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], + GetBitContext *gb) +{ int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index; skip_bits(gb, 2); // object_type @@ -275,10 +286,11 @@ static int decode_pce(AACContext * ac, e * * @return Returns error status. 0 - OK, !0 - error */ -static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], - int channel_config) +static int set_default_channel_config(AACContext *ac, + enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], + int channel_config) { - if(channel_config < 1 || channel_config > 7) { + if (channel_config < 1 || channel_config > 7) { av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", channel_config); return -1; @@ -295,18 +307,18 @@ static int set_default_channel_config(AA * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE */ - if(channel_config != 2) + if (channel_config != 2) new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) - if(channel_config > 1) + if (channel_config > 1) new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) - if(channel_config == 4) + if (channel_config == 4) new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center - if(channel_config > 4) + if (channel_config > 4) new_che_pos[TYPE_CPE][(channel_config == 7) + 1] - = AAC_CHANNEL_BACK; // back stereo - if(channel_config > 5) + = AAC_CHANNEL_BACK; // back stereo + if (channel_config > 5) new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE - if(channel_config == 7) + if (channel_config == 7) new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right return 0; @@ -317,11 +329,13 @@ static int set_default_channel_config(AA * * @return Returns error status. 0 - OK, !0 - error */ -static int decode_ga_specific_config(AACContext * ac, GetBitContext * gb, int channel_config) { +static int decode_ga_specific_config(AACContext *ac, GetBitContext *gb, + int channel_config) +{ enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; int extension_flag, ret; - if(get_bits1(gb)) { // frameLengthFlag + if (get_bits1(gb)) { // frameLengthFlag av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1); return -1; } @@ -330,37 +344,37 @@ static int decode_ga_specific_config(AAC skip_bits(gb, 14); // coreCoderDelay extension_flag = get_bits1(gb); - if(ac->m4ac.object_type == AOT_AAC_SCALABLE || - ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) + if (ac->m4ac.object_type == AOT_AAC_SCALABLE || + ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) skip_bits(gb, 3); // layerNr memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); if (channel_config == 0) { skip_bits(gb, 4); // element_instance_tag - if((ret = decode_pce(ac, new_che_pos, gb))) + if ((ret = decode_pce(ac, new_che_pos, gb))) return ret; } else { - if((ret = set_default_channel_config(ac, new_che_pos, channel_config))) + if ((ret = set_default_channel_config(ac, new_che_pos, channel_config))) return ret; } - if((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config))) + if ((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config))) return ret; if (extension_flag) { switch (ac->m4ac.object_type) { - case AOT_ER_BSAC: - skip_bits(gb, 5); // numOfSubFrame - skip_bits(gb, 11); // layer_length - break; - case AOT_ER_AAC_LC: - case AOT_ER_AAC_LTP: - case AOT_ER_AAC_SCALABLE: - case AOT_ER_AAC_LD: - skip_bits(gb, 3); /* aacSectionDataResilienceFlag + case AOT_ER_BSAC: + skip_bits(gb, 5); // numOfSubFrame + skip_bits(gb, 11); // layer_length + break; + case AOT_ER_AAC_LC: + case AOT_ER_AAC_LTP: + case AOT_ER_AAC_SCALABLE: + case AOT_ER_AAC_LD: + skip_bits(gb, 3); /* aacSectionDataResilienceFlag * aacScalefactorDataResilienceFlag * aacSpectralDataResilienceFlag */ - break; + break; } skip_bits1(gb); // extensionFlag3 (TBD in version 3) } @@ -375,15 +389,17 @@ static int decode_ga_specific_config(AAC * * @return Returns error status. 0 - OK, !0 - error */ -static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) { +static int decode_audio_specific_config(AACContext *ac, void *data, + int data_size) +{ GetBitContext gb; int i; init_get_bits(&gb, data, data_size * 8); - if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) + if ((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) return -1; - if(ac->m4ac.sampling_index > 12) { + if (ac->m4ac.sampling_index > 12) { av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); return -1; } @@ -411,47 +427,52 @@ static int decode_audio_specific_config( * * @return Returns a 32-bit pseudorandom integer */ -static av_always_inline int lcg_random(int previous_val) { +static av_always_inline int lcg_random(int previous_val) +{ return previous_val * 1664525 + 1013904223; } -static void reset_predict_state(PredictorState * ps) { - ps->r0 = 0.0f; - ps->r1 = 0.0f; +static void reset_predict_state(PredictorState *ps) +{ + ps->r0 = 0.0f; + ps->r1 = 0.0f; ps->cor0 = 0.0f; ps->cor1 = 0.0f; ps->var0 = 1.0f; ps->var1 = 1.0f; } -static void reset_all_predictors(PredictorState * ps) { +static void reset_all_predictors(PredictorState *ps) +{ int i; for (i = 0; i < MAX_PREDICTORS; i++) reset_predict_state(&ps[i]); } -static void reset_predictor_group(PredictorState * ps, int group_num) { +static void reset_predictor_group(PredictorState *ps, int group_num) +{ int i; - for (i = group_num-1; i < MAX_PREDICTORS; i+=30) + for (i = group_num - 1; i < MAX_PREDICTORS; i += 30) reset_predict_state(&ps[i]); } -static av_cold int aac_decode_init(AVCodecContext * avccontext) { - AACContext * ac = avccontext->priv_data; +static av_cold int aac_decode_init(AVCodecContext *avccontext) +{ + AACContext *ac = avccontext->priv_data; int i; ac->avccontext = avccontext; if (avccontext->extradata_size > 0) { - if(decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size)) + if (decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size)) return -1; avccontext->sample_rate = ac->m4ac.sample_rate; } else if (avccontext->channels > 0) { ac->m4ac.sample_rate = avccontext->sample_rate; } - avccontext->sample_fmt = SAMPLE_FMT_S16; - avccontext->frame_size = 1024; + avccontext->sample_fmt = SAMPLE_FMT_S16; + avccontext->frame_size = 1024; AAC_INIT_VLC_STATIC( 0, 144); AAC_INIT_VLC_STATIC( 1, 114); @@ -473,25 +494,25 @@ static av_cold int aac_decode_init(AVCod // 32768 - Required to scale values to the correct range for the bias method // for float to int16 conversion. - if(ac->dsp.float_to_int16 == ff_float_to_int16_c) { - ac->add_bias = 385.0f; - ac->sf_scale = 1. / (-1024. * 32768.); + if (ac->dsp.float_to_int16 == ff_float_to_int16_c) { + ac->add_bias = 385.0f; + ac->sf_scale = 1. / (-1024. * 32768.); ac->sf_offset = 0; } else { - ac->add_bias = 0.0f; - ac->sf_scale = 1. / -1024.; + ac->add_bias = 0.0f; + ac->sf_scale = 1. / -1024.; ac->sf_offset = 60; } #if !CONFIG_HARDCODED_TABLES for (i = 0; i < 428; i++) - ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.); + ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.); #endif /* CONFIG_HARDCODED_TABLES */ INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code), - ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), - ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), - 352); + ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), + ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), + 352); ff_mdct_init(&ac->mdct, 11, 1, 1.0); ff_mdct_init(&ac->mdct_small, 8, 1, 1.0); @@ -507,7 +528,8 @@ static av_cold int aac_decode_init(AVCod /** * Skip data_stream_element; reference: table 4.10. */ -static void skip_data_stream_element(GetBitContext * gb) { +static void skip_data_stream_element(GetBitContext *gb) +{ int byte_align = get_bits1(gb); int count = get_bits(gb, 8); if (count == 255) @@ -517,7 +539,9 @@ static void skip_data_stream_element(Get skip_bits_long(gb, 8 * count); } -static int decode_prediction(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb) { +static int decode_prediction(AACContext *ac, IndividualChannelStream *ics, + GetBitContext *gb) +{ int sfb; if (get_bits1(gb)) { ics->predictor_reset_group = get_bits(gb, 5); @@ -537,7 +561,9 @@ static int decode_prediction(AACContext * * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. */ -static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) { +static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics, + GetBitContext *gb, int common_window) +{ if (get_bits1(gb)) { av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n"); memset(ics, 0, sizeof(IndividualChannelStream)); @@ -545,33 +571,33 @@ static int decode_ics_info(AACContext * } 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; + 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[0] == EIGHT_SHORT_SEQUENCE) { int i; ics->max_sfb = get_bits(gb, 4); for (i = 0; i < 7; i++) { if (get_bits1(gb)) { - ics->group_len[ics->num_window_groups-1]++; + ics->group_len[ics->num_window_groups - 1]++; } else { ics->num_window_groups++; - ics->group_len[ics->num_window_groups-1] = 1; + ics->group_len[ics->num_window_groups - 1] = 1; } } - ics->num_windows = 8; - ics->swb_offset = ff_swb_offset_128[ac->m4ac.sampling_index]; - ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index]; - ics->tns_max_bands = ff_tns_max_bands_128[ac->m4ac.sampling_index]; + ics->num_windows = 8; + ics->swb_offset = ff_swb_offset_128[ac->m4ac.sampling_index]; + ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index]; + ics->tns_max_bands = ff_tns_max_bands_128[ac->m4ac.sampling_index]; ics->predictor_present = 0; } else { - ics->max_sfb = get_bits(gb, 6); - ics->num_windows = 1; - ics->swb_offset = ff_swb_offset_1024[ac->m4ac.sampling_index]; - ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index]; - ics->tns_max_bands = ff_tns_max_bands_1024[ac->m4ac.sampling_index]; - ics->predictor_present = get_bits1(gb); + ics->max_sfb = get_bits(gb, 6); + ics->num_windows = 1; + ics->swb_offset = ff_swb_offset_1024[ac->m4ac.sampling_index]; + ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index]; + ics->tns_max_bands = ff_tns_max_bands_1024[ac->m4ac.sampling_index]; + ics->predictor_present = get_bits1(gb); ics->predictor_reset_group = 0; if (ics->predictor_present) { if (ac->m4ac.object_type == AOT_AAC_MAIN) { @@ -591,10 +617,10 @@ static int decode_ics_info(AACContext * } } - if(ics->max_sfb > ics->num_swb) { + if (ics->max_sfb > ics->num_swb) { av_log(ac->avccontext, AV_LOG_ERROR, - "Number of scalefactor bands in group (%d) exceeds limit (%d).\n", - ics->max_sfb, ics->num_swb); + "Number of scalefactor bands in group (%d) exceeds limit (%d).\n", + ics->max_sfb, ics->num_swb); memset(ics, 0, sizeof(IndividualChannelStream)); return -1; } @@ -610,8 +636,10 @@ static int decode_ics_info(AACContext * * * @return Returns error status. 0 - OK, !0 - error */ -static int decode_band_types(AACContext * ac, enum BandType band_type[120], - int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) { +static int decode_band_types(AACContext *ac, enum BandType band_type[120], + int band_type_run_end[120], GetBitContext *gb, + IndividualChannelStream *ics) +{ int g, idx = 0; const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; for (g = 0; g < ics->num_window_groups; g++) { @@ -624,13 +652,13 @@ static int decode_band_types(AACContext av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n"); return -1; } - while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1) + while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1) sect_len += sect_len_incr; sect_len += sect_len_incr; if (sect_len > ics->max_sfb) { av_log(ac->avccontext, AV_LOG_ERROR, - "Number of bands (%d) exceeds limit (%d).\n", - sect_len, ics->max_sfb); + "Number of bands (%d) exceeds limit (%d).\n", + sect_len, ics->max_sfb); return -1; } for (; k < sect_len; k++) { @@ -652,9 +680,12 @@ static int decode_band_types(AACContext * * @return Returns error status. 0 - OK, !0 - error */ -static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb, - unsigned int global_gain, IndividualChannelStream * ics, - enum BandType band_type[120], int band_type_run_end[120]) { +static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb, + unsigned int global_gain, + IndividualChannelStream *ics, + enum BandType band_type[120], + int band_type_run_end[120]) +{ const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); int g, i, idx = 0; int offset[3] = { global_gain, global_gain - 90, 100 }; @@ -664,37 +695,37 @@ static int decode_scalefactors(AACContex for (i = 0; i < ics->max_sfb;) { int run_end = band_type_run_end[idx]; if (band_type[idx] == ZERO_BT) { - for(; i < run_end; i++, idx++) + for (; i < run_end; i++, idx++) sf[idx] = 0.; - }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { - for(; i < run_end; i++, idx++) { + } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { + for (; i < run_end; i++, idx++) { offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; - if(offset[2] > 255U) { + if (offset[2] > 255U) { av_log(ac->avccontext, AV_LOG_ERROR, - "%s (%d) out of range.\n", sf_str[2], offset[2]); + "%s (%d) out of range.\n", sf_str[2], offset[2]); return -1; } - sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; + sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; } - }else if(band_type[idx] == NOISE_BT) { - for(; i < run_end; i++, idx++) { - if(noise_flag-- > 0) + } else if (band_type[idx] == NOISE_BT) { + for (; i < run_end; i++, idx++) { + if (noise_flag-- > 0) offset[1] += get_bits(gb, 9) - 256; else offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; - if(offset[1] > 255U) { + if (offset[1] > 255U) { av_log(ac->avccontext, AV_LOG_ERROR, - "%s (%d) out of range.\n", sf_str[1], offset[1]); + "%s (%d) out of range.\n", sf_str[1], offset[1]); return -1; } - sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset + 100]; + sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100]; } - }else { - for(; i < run_end; i++, idx++) { + } else { + for (; i < run_end; i++, idx++) { offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; - if(offset[0] > 255U) { + if (offset[0] > 255U) { av_log(ac->avccontext, AV_LOG_ERROR, - "%s (%d) out of range.\n", sf_str[0], offset[0]); + "%s (%d) out of range.\n", sf_str[0], offset[0]); return -1; } sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset]; @@ -708,7 +739,9 @@ static int decode_scalefactors(AACContex /** * Decode pulse data; reference: table 4.7. */ -static int decode_pulses(Pulse * pulse, GetBitContext * gb, const uint16_t * swb_offset, int num_swb) { +static int decode_pulses(Pulse *pulse, GetBitContext *gb, + const uint16_t *swb_offset, int num_swb) +{ int i, pulse_swb; pulse->num_pulse = get_bits(gb, 2) + 1; pulse_swb = get_bits(gb, 6); @@ -720,7 +753,7 @@ static int decode_pulses(Pulse * pulse, return -1; pulse->amp[0] = get_bits(gb, 4); for (i = 1; i < pulse->num_pulse; i++) { - pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i-1]; + pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1]; if (pulse->pos[i] > 1023) return -1; pulse->amp[i] = get_bits(gb, 4); @@ -733,8 +766,9 @@ static int decode_pulses(Pulse * pulse, * * @return Returns error status. 0 - OK, !0 - error */ -static int decode_tns(AACContext * ac, TemporalNoiseShaping * tns, - GetBitContext * gb, const IndividualChannelStream * ics) { +static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns, + GetBitContext *gb, const IndividualChannelStream *ics) +{ int w, filt, i, coef_len, coef_res, coef_compress; const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12; @@ -744,9 +778,9 @@ static int decode_tns(AACContext * ac, T for (filt = 0; filt < tns->n_filt[w]; filt++) { int tmp2_idx; - tns->length[w][filt] = get_bits(gb, 6 - 2*is8); + tns->length[w][filt] = get_bits(gb, 6 - 2 * is8); - if ((tns->order[w][filt] = get_bits(gb, 5 - 2*is8)) > tns_max_order) { + if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) { av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.", tns->order[w][filt], tns_max_order); tns->order[w][filt] = 0; @@ -756,7 +790,7 @@ static int decode_tns(AACContext * ac, T tns->direction[w][filt] = get_bits1(gb); coef_compress = get_bits1(gb); coef_len = coef_res + 3 - coef_compress; - tmp2_idx = 2*coef_compress + coef_res; + tmp2_idx = 2 * coef_compress + coef_res; for (i = 0; i < tns->order[w][filt]; i++) tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]; @@ -774,8 +808,9 @@ static int decode_tns(AACContext * ac, T * [1] mask is decoded from bitstream; [2] mask is all 1s; * [3] reserved for scalable AAC */ -static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb, - int ms_present) { +static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb, + int ms_present) +{ int idx; if (ms_present == 1) { for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++) @@ -797,16 +832,20 @@ static void decode_mid_side_stereo(Chann * * @return Returns error status. 0 - OK, !0 - error */ -static int decode_spectrum_and_dequant(AACContext * ac, float coef[1024], GetBitContext * gb, float sf[120], - int pulse_present, const Pulse * pulse, const IndividualChannelStream * ics, enum BandType band_type[120]) { +static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024], + GetBitContext *gb, float sf[120], + int pulse_present, const Pulse *pulse, + const IndividualChannelStream *ics, + enum BandType band_type[120]) +{ int i, k, g, idx = 0; - const int c = 1024/ics->num_windows; - const uint16_t * offsets = ics->swb_offset; + const int c = 1024 / ics->num_windows; + const uint16_t *offsets = ics->swb_offset; float *coef_base = coef; static const float sign_lookup[] = { 1.0f, -1.0f }; for (g = 0; g < ics->num_windows; g++) - memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float)*(c - offsets[ics->max_sfb])); + memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb])); for (g = 0; g < ics->num_window_groups; g++) { for (i = 0; i < ics->max_sfb; i++, idx++) { @@ -816,42 +855,46 @@ static int decode_spectrum_and_dequant(A int group; if (cur_band_type == ZERO_BT || cur_band_type == INTENSITY_BT2 || cur_band_type == INTENSITY_BT) { for (group = 0; group < ics->group_len[g]; group++) { - memset(coef + group * 128 + offsets[i], 0, (offsets[i+1] - offsets[i])*sizeof(float)); + memset(coef + group * 128 + offsets[i], 0, (offsets[i + 1] - offsets[i]) * sizeof(float)); } - }else if (cur_band_type == NOISE_BT) { + } else if (cur_band_type == NOISE_BT) { for (group = 0; group < ics->group_len[g]; group++) { float scale; float band_energy = 0; - for (k = offsets[i]; k < offsets[i+1]; k++) { + for (k = offsets[i]; k < offsets[i + 1]; k++) { ac->random_state = lcg_random(ac->random_state); - coef[group*128+k] = ac->random_state; - band_energy += coef[group*128+k]*coef[group*128+k]; + coef[group * 128 + k] = ac->random_state; + band_energy += coef[group * 128 + k] * coef[group * 128 + k]; } scale = sf[idx] / sqrtf(band_energy); - for (k = offsets[i]; k < offsets[i+1]; k++) { - coef[group*128+k] *= scale; + for (k = offsets[i]; k < offsets[i + 1]; k++) { + coef[group * 128 + k] *= scale; } } - }else { + } else { for (group = 0; group < ics->group_len[g]; group++) { - for (k = offsets[i]; k < offsets[i+1]; k += dim) { + for (k = offsets[i]; k < offsets[i + 1]; k += dim) { const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3); const int coef_tmp_idx = (group << 7) + k; const float *vq_ptr; int j; - if(index >= ff_aac_spectral_sizes[cur_band_type - 1]) { + if (index >= ff_aac_spectral_sizes[cur_band_type - 1]) { av_log(ac->avccontext, AV_LOG_ERROR, - "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n", - cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]); + "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n", + cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]); return -1; } vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim]; if (is_cb_unsigned) { - if (vq_ptr[0]) coef[coef_tmp_idx ] = sign_lookup[get_bits1(gb)]; - if (vq_ptr[1]) coef[coef_tmp_idx + 1] = sign_lookup[get_bits1(gb)]; + if (vq_ptr[0]) + coef[coef_tmp_idx ] = sign_lookup[get_bits1(gb)]; + if (vq_ptr[1]) + coef[coef_tmp_idx + 1] = sign_lookup[get_bits1(gb)]; if (dim == 4) { - if (vq_ptr[2]) coef[coef_tmp_idx + 2] = sign_lookup[get_bits1(gb)]; - if (vq_ptr[3]) coef[coef_tmp_idx + 3] = sign_lookup[get_bits1(gb)]; + if (vq_ptr[2]) + coef[coef_tmp_idx + 2] = sign_lookup[get_bits1(gb)]; + if (vq_ptr[3]) + coef[coef_tmp_idx + 3] = sign_lookup[get_bits1(gb)]; } if (cur_band_type == ESC_BT) { for (j = 0; j < 2; j++) { @@ -860,17 +903,16 @@ static int decode_spectrum_and_dequant(A /* The total length of escape_sequence must be < 22 bits according to the specification (i.e. max is 11111111110xxxxxxxxxx). */ while (get_bits1(gb) && n < 15) n++; - if(n == 15) { + if (n == 15) { av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); return -1; } - n = (1<<n) + get_bits(gb, n); + n = (1 << n) + get_bits(gb, n); coef[coef_tmp_idx + j] *= cbrtf(n) * n; - }else + } else coef[coef_tmp_idx + j] *= vq_ptr[j]; } - }else - { + } else { coef[coef_tmp_idx ] *= vq_ptr[0]; coef[coef_tmp_idx + 1] *= vq_ptr[1]; if (dim == 4) { @@ -878,7 +920,7 @@ static int decode_spectrum_and_dequant(A coef[coef_tmp_idx + 3] *= vq_ptr[3]; } } - }else { + } else { coef[coef_tmp_idx ] = vq_ptr[0]; coef[coef_tmp_idx + 1] = vq_ptr[1]; if (dim == 4) { @@ -896,14 +938,14 @@ static int decode_spectrum_and_dequant(A } } } - coef += ics->group_len[g]<<7; + coef += ics->group_len[g] << 7; } if (pulse_present) { idx = 0; - for(i = 0; i < pulse->num_pulse; i++){ - float co = coef_base[ pulse->pos[i] ]; - while(offsets[idx + 1] <= pulse->pos[i]) + for (i = 0; i < pulse->num_pulse; i++) { + float co = coef_base[ pulse->pos[i] ]; + while (offsets[idx + 1] <= pulse->pos[i]) idx++; if (band_type[idx] != NOISE_BT && sf[idx]) { float ico = -pulse->amp[i]; @@ -918,30 +960,35 @@ static int decode_spectrum_and_dequant(A return 0; } -static av_always_inline float flt16_round(float pf) { +static av_always_inline float flt16_round(float pf) +{ union float754 tmp; tmp.f = pf; tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U; return tmp.f; } -static av_always_inline float flt16_even(float pf) { +static av_always_inline float flt16_even(float pf) +{ union float754 tmp; tmp.f = pf; - tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U>>16)) & 0xFFFF0000U; + tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U; return tmp.f; } -static av_always_inline float flt16_trunc(float pf) { +static av_always_inline float flt16_trunc(float pf) +{ union float754 pun; pun.f = pf; pun.i &= 0xFFFF0000U; return pun.f; } -static void predict(AACContext * ac, PredictorState * ps, float* coef, int output_enable) { - const float a = 0.953125; // 61.0/64 - const float alpha = 0.90625; // 29.0/32 +static void predict(AACContext *ac, PredictorState *ps, float *coef, + int output_enable) +{ + const float a = 0.953125; // 61.0 / 64 + const float alpha = 0.90625; // 29.0 / 32 float e0, e1; float pv; float k1, k2; @@ -968,7 +1015,8 @@ static void predict(AACContext * ac, Pre /** * Apply AAC-Main style frequency domain prediction. */ -static void apply_prediction(AACContext * ac, SingleChannelElement * sce) { +static void apply_prediction(AACContext *ac, SingleChannelElement *sce) +{ int sfb, k; if (!sce->ics.predictor_initialized) { @@ -980,7 +1028,7 @@ static void apply_prediction(AACContext for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) { for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) { predict(ac, &sce->predictor_state[k], &sce->coeffs[k], - sce->ics.predictor_present && sce->ics.prediction_used[sfb]); + sce->ics.predictor_present && sce->ics.prediction_used[sfb]); } } if (sce->ics.predictor_reset_group) @@ -997,11 +1045,13 @@ static void apply_prediction(AACContext * * @return Returns error status. 0 - OK, !0 - error */ -static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) { +static int decode_ics(AACContext *ac, SingleChannelElement *sce, + GetBitContext *gb, int common_window, int scale_flag) +{ Pulse pulse; - TemporalNoiseShaping * tns = &sce->tns; - IndividualChannelStream * ics = &sce->ics; - float * out = sce->coeffs; + TemporalNoiseShaping *tns = &sce->tns; + IndividualChannelStream *ics = &sce->ics; + float *out = sce->coeffs; int global_gain, pulse_present = 0; /* This assignment is to silence a GCC warning about the variable being used @@ -1044,7 +1094,7 @@ static int decode_ics(AACContext * ac, S if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0) return -1; - if(ac->m4ac.object_type == AOT_AAC_MAIN && !common_window) + if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window) apply_prediction(ac, sce); return 0; @@ -1053,27 +1103,28 @@ static int decode_ics(AACContext * ac, S /** * Mid/Side stereo decoding; reference: 4.6.8.1.3. */ -static void apply_mid_side_stereo(ChannelElement * cpe) { - const IndividualChannelStream * ics = &cpe->ch[0].ics; +static void apply_mid_side_stereo(ChannelElement *cpe) +{ + const IndividualChannelStream *ics = &cpe->ch[0].ics; float *ch0 = cpe->ch[0].coeffs; float *ch1 = cpe->ch[1].coeffs; int g, i, k, group, idx = 0; - const uint16_t * offsets = ics->swb_offset; + const uint16_t *offsets = ics->swb_offset; for (g = 0; g < ics->num_window_groups; g++) { for (i = 0; i < ics->max_sfb; i++, idx++) { if (cpe->ms_mask[idx] && - cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { + cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { for (group = 0; group < ics->group_len[g]; group++) { - for (k = offsets[i]; k < offsets[i+1]; k++) { - float tmp = ch0[group*128 + k] - ch1[group*128 + k]; - ch0[group*128 + k] += ch1[group*128 + k]; - ch1[group*128 + k] = tmp; + for (k = offsets[i]; k < offsets[i + 1]; k++) { + float tmp = ch0[group * 128 + k] - ch1[group * 128 + k]; + ch0[group * 128 + k] += ch1[group * 128 + k]; + ch1[group * 128 + k] = tmp; } } } } - ch0 += ics->group_len[g]*128; - ch1 += ics->group_len[g]*128; + ch0 += ics->group_len[g] * 128; + ch1 += ics->group_len[g] * 128; } } @@ -1084,11 +1135,12 @@ static void apply_mid_side_stereo(Channe * [1] mask is decoded from bitstream; [2] mask is all 1s; * [3] reserved for scalable AAC */ -static void apply_intensity_stereo(ChannelElement * cpe, int ms_present) { - const IndividualChannelStream * ics = &cpe->ch[1].ics; - SingleChannelElement * sce1 = &cpe->ch[1]; +static void apply_intensity_stereo(ChannelElement *cpe, int ms_present) +{ + const IndividualChannelStream *ics = &cpe->ch[1].ics; + SingleChannelElement *sce1 = &cpe->ch[1]; float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs; - const uint16_t * offsets = ics->swb_offset; + const uint16_t *offsets = ics->swb_offset; int g, group, i, k, idx = 0; int c; float scale; @@ -1102,8 +1154,8 @@ static void apply_intensity_stereo(Chann c *= 1 - 2 * cpe->ms_mask[idx]; scale = c * sce1->sf[idx]; for (group = 0; group < ics->group_len[g]; group++) - for (k = offsets[i]; k < offsets[i+1]; k++) - coef1[group*128 + k] = scale * coef0[group*128 + k]; + for (k = offsets[i]; k < offsets[i + 1]; k++) + coef1[group * 128 + k] = scale * coef0[group * 128 + k]; } } else { int bt_run_end = sce1->band_type_run_end[idx]; @@ -1111,8 +1163,8 @@ static void apply_intensity_stereo(Chann i = bt_run_end; } } - coef0 += ics->group_len[g]*128; - coef1 += ics->group_len[g]*128; + coef0 += ics->group_len[g] * 128; + coef1 += ics->group_len[g] * 128; } } @@ -1123,7 +1175,8 @@ static void apply_intensity_stereo(Chann * * @return Returns error status. 0 - OK, !0 - error */ -static int decode_cpe(AACContext * ac, GetBitContext * gb, ChannelElement * cpe) { +static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe) +{ int i, ret, common_window, ms_present = 0; common_window = get_bits1(gb); @@ -1134,10 +1187,10 @@ static int decode_cpe(AACContext * ac, G cpe->ch[1].ics = cpe->ch[0].ics; cpe->ch[1].ics.use_kb_window[1] = i; ms_present = get_bits(gb, 2); - if(ms_present == 3) { + if (ms_present == 3) { av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); return -1; - } else if(ms_present) + } else if (ms_present) decode_mid_side_stereo(cpe, gb, ms_present); } if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) @@ -1165,15 +1218,16 @@ static int decode_cpe(AACContext * ac, G * * @return Returns error status. 0 - OK, !0 - error */ -static int decode_cce(AACContext * ac, GetBitContext * gb, ChannelElement * che) { +static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che) +{ int num_gain = 0; int c, g, sfb, ret; int sign; float scale; - SingleChannelElement * sce = &che->ch[0]; - ChannelCoupling * coup = &che->coup; + SingleChannelElement *sce = &che->ch[0]; + ChannelCoupling *coup = &che->coup; - coup->coupling_point = 2*get_bits1(gb); + coup->coupling_point = 2 * get_bits1(gb); coup->num_coupled = get_bits(gb, 3); for (c = 0; c <= coup->num_coupled; c++) { num_gain++; @@ -1186,17 +1240,17 @@ static int decode_cce(AACContext * ac, G } else coup->ch_select[c] = 2; } - coup->coupling_point += get_bits1(gb) || (coup->coupling_point>>1); + coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1); - sign = get_bits(gb, 1); + sign = get_bits(gb, 1); scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3)); if ((ret = decode_ics(ac, sce, gb, 0, 0))) return ret; for (c = 0; c < num_gain; c++) { - int idx = 0; - int cge = 1; + int idx = 0; + int cge = 1; int gain = 0; float gain_cache = 1.; if (c) { @@ -1212,7 +1266,7 @@ static int decode_cce(AACContext * ac, G if (sce->band_type[idx] != ZERO_BT) { if (!cge) { int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; - if (t) { + if (t) { int s = 1; t = gain += t; if (sign) { @@ -1239,10 +1293,12 @@ static int decode_cce(AACContext * ac, G * * @return Returns number of bytes consumed from the TYPE_FIL element. */ -static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) { +static int decode_sbr_extension(AACContext *ac, GetBitContext *gb, + int crc, int cnt) +{ // TODO : sbr_extension implementation av_log_missing_feature(ac->avccontext, "SBR", 0); - skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type + skip_bits_long(gb, 8 * cnt - 4); // -4 due to reading extension type return cnt; } @@ -1251,7 +1307,9 @@ static int decode_sbr_extension(AACConte * * @return Returns number of bytes consumed. */ -static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext * gb) { +static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, + GetBitContext *gb) +{ int i; int num_excl_chan = 0; @@ -1270,20 +1328,22 @@ static int decode_drc_channel_exclusions * * @return Returns number of bytes consumed. */ -static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) { - int n = 1; +static int decode_dynamic_range(DynamicRangeControl *che_drc, + GetBitContext *gb, int cnt) +{ + int n = 1; int drc_num_bands = 1; int i; /* pce_tag_present? */ - if(get_bits1(gb)) { + if (get_bits1(gb)) { che_drc->pce_instance_tag = get_bits(gb, 4); skip_bits(gb, 4); // tag_reserved_bits n++; } /* excluded_chns_present? */ - if(get_bits1(gb)) { + if (get_bits1(gb)) { n += decode_drc_channel_exclusions(che_drc, gb); } @@ -1322,24 +1382,25 @@ static int decode_dynamic_range(DynamicR * * @return Returns number of bytes consumed */ -static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) { +static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt) +{ int crc_flag = 0; int res = cnt; switch (get_bits(gb, 4)) { // extension type - case EXT_SBR_DATA_CRC: - crc_flag++; - case EXT_SBR_DATA: - res = decode_sbr_extension(ac, gb, crc_flag, cnt); - break; - case EXT_DYNAMIC_RANGE: - res = decode_dynamic_range(&ac->che_drc, gb, cnt); - break; - case EXT_FILL: - case EXT_FILL_DATA: - case EXT_DATA_ELEMENT: - default: - skip_bits_long(gb, 8*cnt - 4); - break; + case EXT_SBR_DATA_CRC: + crc_flag++; + case EXT_SBR_DATA: + res = decode_sbr_extension(ac, gb, crc_flag, cnt); + break; + case EXT_DYNAMIC_RANGE: + res = decode_dynamic_range(&ac->che_drc, gb, cnt); + break; + case EXT_FILL: + case EXT_FILL_DATA: + case EXT_DATA_ELEMENT: + default: + skip_bits_long(gb, 8 * cnt - 4); + break; }; return res; } @@ -1350,8 +1411,10 @@ static int decode_extension_payload(AACC * @param decode 1 if tool is used normally, 0 if tool is used in LTP. * @param coef spectral coefficients */ -static void apply_tns(float coef[1024], TemporalNoiseShaping * tns, IndividualChannelStream * ics, int decode) { - const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb); +static void apply_tns(float coef[1024], TemporalNoiseShaping *tns, + IndividualChannelStream *ics, int decode) +{ + const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb); int w, filt, m, i; int bottom, top, order, start, end, size, inc; float lpc[TNS_MAX_ORDER]; @@ -1373,7 +1436,8 @@ static void apply_tns(float coef[1024], if ((size = end - start) <= 0) continue; if (tns->direction[w][filt]) { - inc = -1; start = end - 1; + inc = -1; + start = end - 1; } else { inc = 1; } @@ -1382,7 +1446,7 @@ static void apply_tns(float coef[1024], // ar filter for (m = 0; m < size; m++, start += inc) for (i = 1; i <= FFMIN(m, order); i++) - coef[start] -= coef[start - i*inc] * lpc[i-1]; + coef[start] -= coef[start - i * inc] * lpc[i - 1]; } } } @@ -1390,16 +1454,17 @@ static void apply_tns(float coef[1024], /** * Conduct IMDCT and windowing. */ -static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) { - IndividualChannelStream * ics = &sce->ics; - float * in = sce->coeffs; - float * out = sce->ret; - float * saved = sce->saved; - const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; - const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; - const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; - float * buf = ac->buf_mdct; - float * temp = ac->temp; +static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce) +{ + IndividualChannelStream *ics = &sce->ics; + float *in = sce->coeffs; + float *out = sce->ret; + float *saved = sce->saved; + const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; + const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; + const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; + float *buf = ac->buf_mdct; + float *temp = ac->temp; int i; // imdct @@ -1420,7 +1485,7 @@ static void imdct_and_windowing(AACConte * with a little special sauce for EIGHT_SHORT_SEQUENCE. */ if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) && - (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) { + (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) { ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, ac->add_bias, 512); } else { for (i = 0; i < 448; i++) @@ -1461,13 +1526,16 @@ static void imdct_and_windowing(AACConte * * @param index index into coupling gain array */ -static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) { - IndividualChannelStream * ics = &cce->ch[0].ics; - const uint16_t * offsets = ics->swb_offset; - float * dest = target->coeffs; - const float * src = cce->ch[0].coeffs; +static void apply_dependent_coupling(AACContext *ac, + SingleChannelElement *target, + ChannelElement *cce, int index) +{ + IndividualChannelStream *ics = &cce->ch[0].ics; + const uint16_t *offsets = ics->swb_offset; + float *dest = target->coeffs; + const float *src = cce->ch[0].coeffs; int g, i, group, k, idx = 0; - if(ac->m4ac.object_type == AOT_AAC_LTP) { + if (ac->m4ac.object_type == AOT_AAC_LTP) { av_log(ac->avccontext, AV_LOG_ERROR, "Dependent coupling is not supported together with LTP\n"); return; @@ -1477,15 +1545,15 @@ static void apply_dependent_coupling(AAC if (cce->ch[0].band_type[idx] != ZERO_BT) { const float gain = cce->coup.gain[index][idx]; for (group = 0; group < ics->group_len[g]; group++) { - for (k = offsets[i]; k < offsets[i+1]; k++) { + for (k = offsets[i]; k < offsets[i + 1]; k++) { // XXX dsputil-ize - dest[group*128+k] += gain * src[group*128+k]; + dest[group * 128 + k] += gain * src[group * 128 + k]; } } } } - dest += ics->group_len[g]*128; - src += ics->group_len[g]*128; + dest += ics->group_len[g] * 128; + src += ics->group_len[g] * 128; } } @@ -1494,12 +1562,15 @@ static void apply_dependent_coupling(AAC * * @param index index into coupling gain array */ -static void apply_independent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) { +static void apply_independent_coupling(AACContext *ac, + SingleChannelElement *target, + ChannelElement *cce, int index) +{ int i; const float gain = cce->coup.gain[index][0]; const float bias = ac->add_bias; - const float* src = cce->ch[0].ret; - float* dest = target->ret; + const float *src = cce->ch[0].ret; + float *dest = target->ret; for (i = 0; i < 1024; i++) dest[i] += gain * (src[i] - bias); @@ -1511,9 +1582,10 @@ static void apply_independent_coupling(A * @param index index into coupling gain array * @param apply_coupling_method pointer to (in)dependent coupling function */ -static void apply_channel_coupling(AACContext * ac, ChannelElement * cc, - enum RawDataBlockType type, int elem_id, enum CouplingPoint coupling_point, - void (*apply_coupling_method)(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index)) +static void apply_channel_coupling(AACContext *ac, ChannelElement *cc, + enum RawDataBlockType type, int elem_id, + enum CouplingPoint coupling_point, + void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)) { int i, c; @@ -1522,7 +1594,7 @@ static void apply_channel_coupling(AACCo int index = 0; if (cce && cce->coup.coupling_point == coupling_point) { - ChannelCoupling * coup = &cce->coup; + ChannelCoupling *coup = &cce->coup; for (c = 0; c <= coup->num_coupled; c++) { if (coup->type[c] == type && coup->id_select[c] == elem_id) { @@ -1543,33 +1615,34 @@ static void apply_channel_coupling(AACCo /** * Convert spectral data to float samples, applying all supported tools as appropriate. */ -static void spectral_to_sample(AACContext * ac) { +static void spectral_to_sample(AACContext *ac) +{ int i, type; - for(type = 3; type >= 0; type--) { + for (type = 3; type >= 0; type--) { for (i = 0; i < MAX_ELEM_ID; i++) { ChannelElement *che = ac->che[type][i]; - if(che) { - if(type <= TYPE_CPE) + if (che) { + if (type <= TYPE_CPE) apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling); - if(che->ch[0].tns.present) + if (che->ch[0].tns.present) apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1); - if(che->ch[1].tns.present) + if (che->ch[1].tns.present) apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1); - if(type <= TYPE_CPE) + if (type <= TYPE_CPE) apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling); - if(type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) + if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) imdct_and_windowing(ac, &che->ch[0]); - if(type == TYPE_CPE) + if (type == TYPE_CPE) imdct_and_windowing(ac, &che->ch[1]); - if(type <= TYPE_CCE) + if (type <= TYPE_CCE) apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling); } } } } -static int parse_adts_frame_header(AACContext * ac, GetBitContext * gb) { - +static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb) +{ int size; AACADTSHeaderInfo hdr_info; @@ -1598,16 +1671,18 @@ static int parse_adts_frame_header(AACCo return size; } -static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, AVPacket *avpkt) { +static int aac_decode_frame(AVCodecContext *avccontext, void *data, + int *data_size, AVPacket *avpkt) +{ const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; - AACContext * ac = avccontext->priv_data; - ChannelElement * che = NULL; + AACContext *ac = avccontext->priv_data; + ChannelElement *che = NULL; GetBitContext gb; enum RawDataBlockType elem_type; int err, elem_id, data_size_tmp; - init_get_bits(&gb, buf, buf_size*8); + init_get_bits(&gb, buf, buf_size * 8); if (show_bits(&gb, 12) == 0xfff) { if (parse_adts_frame_header(ac, &gb) < 0) { @@ -1624,7 +1699,7 @@ static int aac_decode_frame(AVCodecConte while ((elem_type = get_bits(&gb, 3)) != TYPE_END) { elem_id = get_bits(&gb, 4); - if(elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) { + if (elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) { av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id); return -1; } @@ -1652,11 +1727,10 @@ static int aac_decode_frame(AVCodecConte err = 0; break; - case TYPE_PCE: - { + case TYPE_PCE: { enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); - if((err = decode_pce(ac, new_che_pos, &gb))) + if ((err = decode_pce(ac, new_che_pos, &gb))) break; if (ac->output_configured) av_log(avccontext, AV_LOG_ERROR, @@ -1679,7 +1753,7 @@ static int aac_decode_frame(AVCodecConte break; } - if(err) + if (err) return err; } @@ -1692,7 +1766,7 @@ static int aac_decode_frame(AVCodecConte } data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t); - if(*data_size < data_size_tmp) { + if (*data_size < data_size_tmp) { 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, data_size_tmp); @@ -1705,18 +1779,19 @@ static int aac_decode_frame(AVCodecConte return buf_size; } -static av_cold int aac_decode_close(AVCodecContext * avccontext) { - AACContext * ac = avccontext->priv_data; +static av_cold int aac_decode_close(AVCodecContext *avccontext) +{ + AACContext *ac = avccontext->priv_data; int i, type; for (i = 0; i < MAX_ELEM_ID; i++) { - for(type = 0; type < 4; type++) + for (type = 0; type < 4; type++) av_freep(&ac->che[type][i]); } ff_mdct_end(&ac->mdct); ff_mdct_end(&ac->mdct_small); - return 0 ; + return 0; } AVCodec aac_decoder = { @@ -1729,5 +1804,7 @@ AVCodec aac_decoder = { aac_decode_close, aac_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), - .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, + .sample_fmts = (enum SampleFormat[]) { + SAMPLE_FMT_S16,SAMPLE_FMT_NONE + }, };

1 0

r19561 - in trunk/libavcodec: celp_filters.c celp_filters.h
by subversion＠mplayerhq.hu 02 Aug '09

02 Aug '09

Author: diego Date: Sun Aug 2 12:34:30 2009 New Revision: 19561 Log: cosmetics: K&R coding style Modified: trunk/libavcodec/celp_filters.c trunk/libavcodec/celp_filters.h Modified: trunk/libavcodec/celp_filters.c ============================================================================== --- trunk/libavcodec/celp_filters.c Sun Aug 2 10:47:02 2009 (r19560) +++ trunk/libavcodec/celp_filters.c Sun Aug 2 12:34:30 2009 (r19561) @@ -25,11 +25,10 @@ #include "avcodec.h" #include "celp_filters.h" -void ff_celp_convolve_circ( - int16_t* fc_out, - const int16_t* fc_in, - const int16_t* filter, - int len) +void ff_celp_convolve_circ(int16_t* fc_out, + const int16_t* fc_in, + const int16_t* filter, + int len) { int i, k; @@ -37,44 +36,39 @@ void ff_celp_convolve_circ( /* Since there are few pulses over an entire subframe (i.e. almost all fc_in[i] are zero) it is faster to loop over fc_in first. */ - for(i=0; i<len; i++) - { - if(fc_in[i]) - { - for(k=0; k<i; k++) + for (i = 0; i < len; i++) { + if (fc_in[i]) { + for (k = 0; k < i; k++) fc_out[k] += (fc_in[i] * filter[len + k - i]) >> 15; - for(k=i; k<len; k++) + for (k = i; k < len; k++) fc_out[k] += (fc_in[i] * filter[ k - i]) >> 15; } } } -int ff_celp_lp_synthesis_filter( - int16_t *out, - const int16_t* filter_coeffs, - const int16_t* in, - int buffer_length, - int filter_length, - int stop_on_overflow, - int rounder) +int ff_celp_lp_synthesis_filter(int16_t *out, + const int16_t* filter_coeffs, + const int16_t* in, + int buffer_length, + int filter_length, + int stop_on_overflow, + int rounder) { int i,n; // Avoids a +1 in the inner loop. filter_length++; - for(n=0; n<buffer_length; n++) - { + for (n = 0; n < buffer_length; n++) { int sum = rounder; - for(i=1; i<filter_length; i++) + for (i = 1; i < filter_length; i++) sum -= filter_coeffs[i-1] * out[n-i]; sum = (sum >> 12) + in[n]; - if(sum + 0x8000 > 0xFFFFU) - { - if(stop_on_overflow) + if (sum + 0x8000 > 0xFFFFU) { + if (stop_on_overflow) return 1; sum = (sum >> 31) ^ 32767; } @@ -84,42 +78,38 @@ int ff_celp_lp_synthesis_filter( return 0; } -void ff_celp_lp_synthesis_filterf( - float *out, - const float* filter_coeffs, - const float* in, - int buffer_length, - int filter_length) +void ff_celp_lp_synthesis_filterf(float *out, + const float* filter_coeffs, + const float* in, + int buffer_length, + int filter_length) { int i,n; // Avoids a +1 in the inner loop. filter_length++; - for(n=0; n<buffer_length; n++) - { + for (n = 0; n < buffer_length; n++) { out[n] = in[n]; - for(i=1; i<filter_length; i++) + for (i = 1; i < filter_length; i++) out[n] -= filter_coeffs[i-1] * out[n-i]; } } -void ff_celp_lp_zero_synthesis_filterf( - float *out, - const float* filter_coeffs, - const float* in, - int buffer_length, - int filter_length) +void ff_celp_lp_zero_synthesis_filterf(float *out, + const float* filter_coeffs, + const float* in, + int buffer_length, + int filter_length) { int i,n; // Avoids a +1 in the inner loop. filter_length++; - for(n=0; n<buffer_length; n++) - { + for (n = 0; n < buffer_length; n++) { out[n] = in[n]; - for(i=1; i<filter_length; i++) + for (i = 1; i < filter_length; i++) out[n] -= filter_coeffs[i-1] * in[n-i]; } } Modified: trunk/libavcodec/celp_filters.h ============================================================================== --- trunk/libavcodec/celp_filters.h Sun Aug 2 10:47:02 2009 (r19560) +++ trunk/libavcodec/celp_filters.h Sun Aug 2 12:34:30 2009 (r19561) @@ -36,11 +36,10 @@ * * \note fc_in and fc_out should not overlap! */ -void ff_celp_convolve_circ( - int16_t* fc_out, - const int16_t* fc_in, - const int16_t* filter, - int len); +void ff_celp_convolve_circ(int16_t* fc_out, + const int16_t* fc_in, + const int16_t* filter, + int len); /** * LP synthesis filter. @@ -60,14 +59,13 @@ void ff_celp_convolve_circ( * * Routine applies 1/A(z) filter to given speech data. */ -int ff_celp_lp_synthesis_filter( - int16_t *out, - const int16_t* filter_coeffs, - const int16_t* in, - int buffer_length, - int filter_length, - int stop_on_overflow, - int rounder); +int ff_celp_lp_synthesis_filter(int16_t *out, + const int16_t* filter_coeffs, + const int16_t* in, + int buffer_length, + int filter_length, + int stop_on_overflow, + int rounder); /** * LP synthesis filter. @@ -84,12 +82,11 @@ int ff_celp_lp_synthesis_filter( * * Routine applies 1/A(z) filter to given speech data. */ -void ff_celp_lp_synthesis_filterf( - float *out, - const float* filter_coeffs, - const float* in, - int buffer_length, - int filter_length); +void ff_celp_lp_synthesis_filterf(float *out, + const float* filter_coeffs, + const float* in, + int buffer_length, + int filter_length); /** * LP zero synthesis filter. @@ -106,11 +103,10 @@ void ff_celp_lp_synthesis_filterf( * * Routine applies A(z) filter to given speech data. */ -void ff_celp_lp_zero_synthesis_filterf( - float *out, - const float* filter_coeffs, - const float* in, - int buffer_length, - int filter_length); +void ff_celp_lp_zero_synthesis_filterf(float *out, + const float* filter_coeffs, + const float* in, + int buffer_length, + int filter_length); #endif /* AVCODEC_CELP_FILTERS_H */

1 0

r19560 - trunk/libavcodec/mjpeg_parser.c
by subversion＠mplayerhq.hu 02 Aug '09

02 Aug '09

Author: bcoudurier Date: Sun Aug 2 10:47:02 2009 New Revision: 19560 Log: cosmetics, reindent Modified: trunk/libavcodec/mjpeg_parser.c Modified: trunk/libavcodec/mjpeg_parser.c ============================================================================== --- trunk/libavcodec/mjpeg_parser.c Sun Aug 2 10:46:43 2009 (r19559) +++ trunk/libavcodec/mjpeg_parser.c Sun Aug 2 10:47:02 2009 (r19560) @@ -71,9 +71,9 @@ static int find_frame_end(ParseContext * } static int jpeg_parse(AVCodecParserContext *s, - AVCodecContext *avctx, - const uint8_t **poutbuf, int *poutbuf_size, - const uint8_t *buf, int buf_size) + AVCodecContext *avctx, + const uint8_t **poutbuf, int *poutbuf_size, + const uint8_t *buf, int buf_size) { ParseContext *pc = s->priv_data; int next;

1 0

r19559 - trunk/libavcodec/mjpeg_parser.c
by subversion＠mplayerhq.hu 02 Aug '09

02 Aug '09

Author: bcoudurier Date: Sun Aug 2 10:46:43 2009 New Revision: 19559 Log: cosmetics, reindent Modified: trunk/libavcodec/mjpeg_parser.c Modified: trunk/libavcodec/mjpeg_parser.c ============================================================================== --- trunk/libavcodec/mjpeg_parser.c Sun Aug 2 10:46:11 2009 (r19558) +++ trunk/libavcodec/mjpeg_parser.c Sun Aug 2 10:46:43 2009 (r19559) @@ -81,13 +81,13 @@ static int jpeg_parse(AVCodecParserConte if(s->flags & PARSER_FLAG_COMPLETE_FRAMES){ next= buf_size; }else{ - next= find_frame_end(pc, buf, buf_size); + next= find_frame_end(pc, buf, buf_size); - if (ff_combine_frame(pc, next, &buf, &buf_size) < 0) { - *poutbuf = NULL; - *poutbuf_size = 0; - return buf_size; - } + if (ff_combine_frame(pc, next, &buf, &buf_size) < 0) { + *poutbuf = NULL; + *poutbuf_size = 0; + return buf_size; + } } *poutbuf = buf;

1 0

r19558 - trunk/libavcodec/mjpeg_parser.c
by subversion＠mplayerhq.hu 02 Aug '09

02 Aug '09

Author: bcoudurier Date: Sun Aug 2 10:46:11 2009 New Revision: 19558 Log: fix jpeg parser when PARSER_FLAG_COMPLETE_FRAMES is set Modified: trunk/libavcodec/mjpeg_parser.c Modified: trunk/libavcodec/mjpeg_parser.c ============================================================================== --- trunk/libavcodec/mjpeg_parser.c Fri Jul 31 09:40:35 2009 (r19557) +++ trunk/libavcodec/mjpeg_parser.c Sun Aug 2 10:46:11 2009 (r19558) @@ -78,6 +78,9 @@ static int jpeg_parse(AVCodecParserConte ParseContext *pc = s->priv_data; int next; + if(s->flags & PARSER_FLAG_COMPLETE_FRAMES){ + next= buf_size; + }else{ next= find_frame_end(pc, buf, buf_size); if (ff_combine_frame(pc, next, &buf, &buf_size) < 0) { @@ -85,6 +88,7 @@ static int jpeg_parse(AVCodecParserConte *poutbuf_size = 0; return buf_size; } + } *poutbuf = buf; *poutbuf_size = buf_size;

1 0

r29464 - trunk/libswscale/swscale.c
by subversion＠mplayerhq.hu 01 Aug '09

01 Aug '09

Author: cehoyos Date: Sat Aug 1 23:30:36 2009 New Revision: 29464 Log: Fix regression when converting to nv12/nv21. Patch by Guennadi Liakhovetski, g D liakhovetski A gmx D de Modified: trunk/libswscale/swscale.c Modified: trunk/libswscale/swscale.c ============================================================================== --- trunk/libswscale/swscale.c Fri Jul 31 22:13:02 2009 (r29463) +++ trunk/libswscale/swscale.c Sat Aug 1 23:30:36 2009 (r29464) @@ -2759,7 +2759,9 @@ SwsContext *sws_getContext(int srcW, int || (isGray(dstFormat) && isGray(srcFormat)) || (isPlanarYUV(srcFormat) && isPlanarYUV(dstFormat) && c->chrDstHSubSample == c->chrSrcHSubSample - && c->chrDstVSubSample == c->chrSrcVSubSample)) + && c->chrDstVSubSample == c->chrSrcVSubSample + && dstFormat != PIX_FMT_NV12 && dstFormat != PIX_FMT_NV21 + && srcFormat != PIX_FMT_NV12 && srcFormat != PIX_FMT_NV21)) { if (isPacked(c->srcFormat)) c->swScale= packedCopy;

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