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

How is everyone progressing?
by Robert Swain 01 Aug '08

01 Aug '08

Hello GSoCers, I was wondering how everyone was progressing on their tasks. Here is the list of people and what I perceive as their status: LC AAC encoder - Kostya - Active - Working on psychoacoustic tuning stuff WMA Pro decoder - Sascha - ? - ? Multi-threaded decoding - Alexander - Active - Still working on h.264 but it is performing well. MLP encoder - Ramiro - ? - ? ALAC encoder - Jai - ? - The last I heard was that it is compressing better than FLAC. LGPL swscale - Keiji - ? - ? Nellymoser - Bart - Active - ? MXF - Zhentan - Active - ? Please respond to this mail commenting on your (or your student's) progress so we can all see what's going on. Regards, Rob

7 8

[soc]: r2963 - aacenc/aacpsy.c
by kostya 31 Jul '08

31 Jul '08

Author: kostya Date: Thu Jul 31 19:04:08 2008 New Revision: 2963 Log: Scalefactors limit should be calculated and applied over full frame, not an individual window Modified: aacenc/aacpsy.c Modified: aacenc/aacpsy.c ============================================================================== --- aacenc/aacpsy.c (original) +++ aacenc/aacpsy.c Thu Jul 31 19:04:08 2008 @@ -745,21 +745,25 @@ static void psy_3gpp_process(AACPsyConte cpe->ch[ch].sf_idx[w][g] = av_clip(cpe->ch[ch].sf_idx[w][g], prev_scale - SCALE_MAX_DIFF, prev_scale + SCALE_MAX_DIFF); prev_scale = cpe->ch[ch].sf_idx[w][g]; } + } //limit scalefactors + for(w = 0; w < cpe->ch[ch].ics.num_windows; w++) for(g = 0; g < cpe->ch[ch].ics.num_swb; g++){ if(cpe->ch[ch].zeroes[w][g]) continue; min_scale = FFMIN(min_scale, cpe->ch[ch].sf_idx[w][g]); } + for(w = 0; w < cpe->ch[ch].ics.num_windows; w++) for(g = 0; g < cpe->ch[ch].ics.num_swb; g++){ if(cpe->ch[ch].zeroes[w][g]) continue; cpe->ch[ch].sf_idx[w][g] = FFMIN(cpe->ch[ch].sf_idx[w][g], min_scale + SCALE_MAX_DIFF); } + for(w = 0; w < cpe->ch[ch].ics.num_windows; w++) for(g = 0; g < cpe->ch[ch].ics.num_swb; g++){ if(cpe->ch[ch].zeroes[w][g]) continue; cpe->ch[ch].sf_idx[w][g] = av_clip(SCALE_ONE_POS + cpe->ch[ch].sf_idx[w][g], 0, SCALE_MAX_POS); 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;

1 0

[soc]: r2962 - aacenc/aacenc.c
by kostya 31 Jul '08

31 Jul '08

Author: kostya Date: Thu Jul 31 19:02:56 2008 New Revision: 2962 Log: Make encoder encode multichannel audio corresponding to default configuration Modified: aacenc/aacenc.c Modified: aacenc/aacenc.c ============================================================================== --- aacenc/aacenc.c (original) +++ aacenc/aacenc.c Thu Jul 31 19:02:56 2008 @@ -143,6 +143,16 @@ static const struct { { -1, NULL , NULL , 0 }, // intensity in-phase }; +/** default channel configurations */ +static const uint8_t aac_chan_configs[6][5] = { + {1, ID_SCE}, // 1 channel - single channel element + {1, ID_CPE}, // 2 channels - channel pair + {2, ID_SCE, ID_CPE}, // 3 channels - center + stereo + {3, ID_SCE, ID_CPE, ID_SCE}, // 4 channels - front center + stereo + back center + {3, ID_SCE, ID_CPE, ID_CPE}, // 5 channels - front center + stereo + back stereo + {4, ID_SCE, ID_CPE, ID_CPE, ID_LFE}, // 6 channels - front center + stereo + back stereo + LFE +}; + typedef struct { PutBitContext pb; MDCTContext mdct1024; @@ -197,6 +207,10 @@ static av_cold int aac_encode_init(AVCod av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate); return -1; } + if(avctx->channels > 6){ + av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels); + return -1; + } s->samplerate_index = i; s->swb_sizes1024 = swb_size_1024[i]; s->swb_num1024 = num_swb_1024[i]; @@ -212,9 +226,9 @@ static av_cold int aac_encode_init(AVCod ff_sine_window_init(sine_long_1024, 1024); ff_sine_window_init(sine_short_128, 128); - s->cpe = av_mallocz(sizeof(ChannelElement) * ((avctx->channels + 1) >> 1)); + s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]); //TODO: psy model selection with some option - ff_aac_psy_init(&s->psy, avctx, AAC_PSY_3GPP, (avctx->channels + 1) >> 1, 0, s->swb_sizes1024, s->swb_num1024, s->swb_sizes128, s->swb_num128); + ff_aac_psy_init(&s->psy, avctx, AAC_PSY_3GPP, aac_chan_configs[avctx->channels-1][0], 0, s->swb_sizes1024, s->swb_num1024, s->swb_sizes128, s->swb_num128); avctx->extradata = av_malloc(2); avctx->extradata_size = 2; put_audio_specific_config(avctx); @@ -667,9 +681,11 @@ static int aac_encode_frame(AVCodecConte uint8_t *frame, int buf_size, void *data) { AACEncContext *s = avctx->priv_data; - int16_t *samples = s->samples, *samples2; + int16_t *samples = s->samples, *samples2, *la; ChannelElement *cpe; - int i, j, chans, tag; + int i, j, chans, tag, start_ch; + const uint8_t *chan_map = aac_chan_configs[avctx->channels-1]; + int chan_el_counter[4]; if(!samples){ s->samples = av_malloc(1024 * avctx->channels * sizeof(s->samples[0])); @@ -687,18 +703,21 @@ static int aac_encode_frame(AVCodecConte put_bits(&s->pb, 4, 0); put_program_config_element(avctx, s); }*/ - for(i = 0; i < avctx->channels; i += 2){ - chans = FFMIN(avctx->channels - i, 2); - tag = chans > 1 ? ID_CPE : ID_SCE; - cpe = &s->cpe[(i + 1)/2]; - samples2 = samples + i; - ff_aac_psy_suggest_window(&s->psy, samples2, data, (i + 1) >> 1, tag, cpe); + start_ch = 0; + memset(chan_el_counter, 0, sizeof(chan_el_counter)); + for(i = 0; i < chan_map[0]; i++){ + tag = chan_map[i+1]; + chans = tag == ID_CPE ? 2 : 1; + cpe = &s->cpe[i]; + samples2 = samples + start_ch; + la = (uint16_t*)data + start_ch; + ff_aac_psy_suggest_window(&s->psy, samples2, la, i, tag, cpe); for(j = 0; j < chans; j++){ analyze(avctx, s, cpe, samples2, j); } - ff_aac_psy_analyze(&s->psy, (i + 1) >> 1, tag, cpe); + ff_aac_psy_analyze(&s->psy, i, tag, cpe); put_bits(&s->pb, 3, tag); - put_bits(&s->pb, 4, i >> 1); + put_bits(&s->pb, 4, chan_el_counter[tag]++); if(chans == 2){ put_bits(&s->pb, 1, cpe->common_window); if(cpe->common_window){ @@ -709,6 +728,7 @@ static int aac_encode_frame(AVCodecConte for(j = 0; j < chans; j++){ encode_individual_channel(avctx, cpe, j); } + start_ch += chans; } put_bits(&s->pb, 3, ID_END);

1 0

[soc]: r2961 - aacenc/aacpsy.c
by kostya 31 Jul '08

31 Jul '08

Author: kostya Date: Thu Jul 31 19:01:46 2008 New Revision: 2961 Log: Make 3GPP-inspired model correctly work with multichannel audio Modified: aacenc/aacpsy.c Modified: aacenc/aacpsy.c ============================================================================== --- aacenc/aacpsy.c (original) +++ aacenc/aacpsy.c Thu Jul 31 19:01:46 2008 @@ -348,7 +348,7 @@ typedef struct Psy3gppContext{ int avg_bits; float ath_l[64]; float ath_s[16]; - Psy3gppChannel ch; + Psy3gppChannel *ch; }Psy3gppContext; /** @@ -425,6 +425,7 @@ static av_cold int psy_3gpp_init(AACPsyC } pctx->avg_bits = apc->avctx->bit_rate * 1024 / apc->avctx->sample_rate; + pctx->ch = av_mallocz(sizeof(Psy3gppChannel) * elements); return 0; } @@ -460,7 +461,7 @@ static void psy_3gpp_window(AACPsyContex 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; + Psy3gppChannel *pch = &pctx->ch[tag]; uint8_t grouping[2]; enum WindowSequence win[2]; @@ -580,7 +581,7 @@ static void psy_3gpp_process(AACPsyConte float stereo_att, pe_target; int bits_avail; int chans = type == ID_CPE ? 2 : 1; - Psy3gppChannel *pch = &pctx->ch; + Psy3gppChannel *pch = &pctx->ch[tag]; //calculate and apply stereo attenuation factor - 5.2 if(chans > 1 && cpe->common_window){ @@ -785,6 +786,8 @@ static void psy_3gpp_process(AACPsyConte static av_cold void psy_3gpp_end(AACPsyContext *apc) { + Psy3gppContext *pctx = (Psy3gppContext*) apc->model_priv_data; + av_freep(&pctx->ch); av_freep(&apc->model_priv_data); }

1 0

[soc]: r2960 - aacenc/aacpsy.h
by kostya 31 Jul '08

31 Jul '08

Author: kostya Date: Thu Jul 31 19:00:50 2008 New Revision: 2960 Log: Document psychoacoustic model interface functions Modified: aacenc/aacpsy.h Modified: aacenc/aacpsy.h ============================================================================== --- aacenc/aacpsy.h (original) +++ aacenc/aacpsy.h Thu Jul 31 19:00:50 2008 @@ -61,12 +61,54 @@ typedef struct AACPsyModel { void (*end) (AACPsyContext *apc); }AACPsyModel; +/** + * Initialize psychoacoustic model. + * + * @param ctx model context + * @param avctx codec context + * @param model model implementation that will be used + * @param elements number of channel elements (single channel or channel pair) to handle by model + * @param flags model flags, may be ignored by model if unsupported + * @param bands1024 scalefactor band lengths for long (1024 samples) frame + * @param num_bands1024 number of scalefactor bands for long frame + * @param bands128 scalefactor band lengths for short (128 samples) frame + * @param num_bands128 number of scalefactor bands for short frame + * + * @return zero if successful, a negative value if not + */ int ff_aac_psy_init(AACPsyContext *ctx, AVCodecContext *avctx, enum AACPsyModelType model, int elements, int flags, const uint8_t *bands1024, int num_bands1024, const uint8_t *bands128, int num_bands128); + +/** + * Set window sequence and related parameters for channel element. + * + * @param ctx model context + * @param audio samples for the current frame + * @param la lookahead samples (NULL when unavailable) + * @param tag number of channel element to analyze + * @param type channel element type (e.g. ID_SCE or ID_CPE) + * @param cpe pointer to the current channel element + */ void ff_aac_psy_suggest_window(AACPsyContext *ctx, int16_t *audio, int16_t *la, int tag, int type, ChannelElement *cpe); + +/** + * Perform psychoacoustic analysis and output coefficients in integer form + * along with scalefactors, M/S flags, etc. + * + * @param ctx model context + * @param tag number of channel element to analyze + * @param type channel element type (e.g. ID_SCE or ID_CPE) + * @param cpe pointer to the current channel element + */ void ff_aac_psy_analyze(AACPsyContext *ctx, int tag, int type, ChannelElement *cpe); + +/** + * Cleanup model context at the end. + * + * @param ctx model context + */ void ff_aac_psy_end(AACPsyContext *ctx); #endif /* FFMPEG_AACPSY_H */

1 0

[soc]: r2959 - in aacenc: aacpsy.c aacpsy.h
by kostya 31 Jul '08

31 Jul '08

Author: kostya Date: Thu Jul 31 18:58:19 2008 New Revision: 2959 Log: Model type should belong to enum Modified: aacenc/aacpsy.c aacenc/aacpsy.h Modified: aacenc/aacpsy.c ============================================================================== --- aacenc/aacpsy.c (original) +++ aacenc/aacpsy.c Thu Jul 31 18:58:19 2008 @@ -814,7 +814,7 @@ static const AACPsyModel psy_models[AAC_ }; int av_cold ff_aac_psy_init(AACPsyContext *ctx, AVCodecContext *avctx, - int model, int elements, int flags, + enum AACPsyModelType model, int elements, int flags, const uint8_t *bands1024, int num_bands1024, const uint8_t *bands128, int num_bands128) { Modified: aacenc/aacpsy.h ============================================================================== --- aacenc/aacpsy.h (original) +++ aacenc/aacpsy.h Thu Jul 31 18:58:19 2008 @@ -62,7 +62,7 @@ typedef struct AACPsyModel { }AACPsyModel; int ff_aac_psy_init(AACPsyContext *ctx, AVCodecContext *avctx, - int model, int elements, int flags, + enum AACPsyModelType model, int elements, int flags, const uint8_t *bands1024, int num_bands1024, const uint8_t *bands128, int num_bands128); void ff_aac_psy_suggest_window(AACPsyContext *ctx, int16_t *audio, int16_t *la, int tag, int type, ChannelElement *cpe);

1 0

[soc]: r2958 - in aacenc: aacpsy.c aacpsy.h
by kostya 31 Jul '08

31 Jul '08

Author: kostya Date: Thu Jul 31 18:57:40 2008 New Revision: 2958 Log: Store model flags in context Modified: aacenc/aacpsy.c aacenc/aacpsy.h Modified: aacenc/aacpsy.c ============================================================================== --- aacenc/aacpsy.c (original) +++ aacenc/aacpsy.c Thu Jul 31 18:57:40 2008 @@ -829,6 +829,7 @@ int av_cold ff_aac_psy_init(AACPsyContex pow2sf_tab[i] = pow(2, (i - 200)/4.); ctx->avctx = avctx; + ctx->flags = flags; ctx->bands1024 = bands1024; ctx->num_bands1024 = num_bands1024; ctx->bands128 = bands128; Modified: aacenc/aacpsy.h ============================================================================== --- aacenc/aacpsy.h (original) +++ aacenc/aacpsy.h Thu Jul 31 18:57:40 2008 @@ -41,6 +41,7 @@ typedef struct AACPsyContext { AVCodecContext *avctx; DSPContext dsp; + int flags; int window_type[2]; int window_shape[2]; const uint8_t *bands1024;

1 0

[soc]: r2957 - in aacenc: aac.h aacenc.c aacpsy.c aacpsy.h
by kostya 31 Jul '08

31 Jul '08

Author: kostya Date: Thu Jul 31 18:57:15 2008 New Revision: 2957 Log: Move borrowed declarations to aac.h, it will be easier to merge with decoder in future Added: aacenc/aac.h Modified: aacenc/aacenc.c aacenc/aacpsy.c aacenc/aacpsy.h Added: aacenc/aac.h ============================================================================== --- (empty file) +++ aacenc/aac.h Thu Jul 31 18:57:15 2008 @@ -0,0 +1,167 @@ +/* this file is a mere collection of things that were borrowed from + * GSoC AAC decoder and should be put into common aac.h + * while the merge is not done, declarations should reside here + */ +#ifndef ERSATZ_AAC_H +#define ERSATZ_AAC_H + +#define MAX_SWB_SIZE 51 + +DECLARE_ALIGNED_16(static float, kbd_long_1024[1024]); +DECLARE_ALIGNED_16(static float, kbd_short_128[128]); +DECLARE_ALIGNED_16(static float, sine_long_1024[1024]); +DECLARE_ALIGNED_16(static float, sine_short_128[128]); + +/** + * window sequences + */ +enum WindowSequence { + ONLY_LONG_SEQUENCE, + LONG_START_SEQUENCE, + EIGHT_SHORT_SEQUENCE, + LONG_STOP_SEQUENCE, +}; + +/** + * IDs for raw_data_block + */ +enum { + ID_SCE = 0x0, + ID_CPE, + ID_CCE, + ID_LFE, + ID_DSE, + ID_PCE, + ID_FIL, + ID_END +}; + +/** + * special codebooks + */ +enum Codebook { + ZERO_HCB = 0, + FIRST_PAIR_HCB = 5, + ESC_HCB = 11, + NOISE_HCB = 13, + INTENSITY_HCB2 = 14, + INTENSITY_HCB = 15, + ESC_FLAG = 16, +}; + +/** + * pulse tool + */ +typedef struct { + int present; + int num_pulse; + int start; + int offset[4]; + int amp[4]; +} Pulse; + +#define MAX_TAGID 16 + +/** + * Program configuration - describes how channels are arranged. Either read from + * stream (ID_PCE) or created based on a default fixed channel arrangement. + */ +typedef struct { + int che_type[4][MAX_TAGID]; ///< channel element type with the first index as the first 4 raw_data_block IDs + int mono_mixdown; ///< The SCE tag to use if user requests mono output, -1 if not available. + int stereo_mixdown; ///< The CPE tag to use if user requests stereo output, -1 if not available. + int matrix_mixdown; ///< The CPE tag to use if user requests matrixed stereo output, -1 if not available. + int mixdown_coeff_index; ///< 0-3 + int pseudo_surround; ///< Mix surround channels out of phase. +} ProgramConfig; + + +/** + * Individual Channel Stream + */ +typedef struct { + int intensity_present; + uint8_t max_sfb; ///< number of scalefactor bands per group + enum WindowSequence window_sequence; + enum WindowSequence window_sequence_prev; + uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window, otherwise use a sinus window. + int num_window_groups; + uint8_t grouping; + uint8_t group_len[8]; + const uint8_t *swb_sizes; + int num_swb; + int num_windows; + int tns_max_bands; +} IndividualChannelStream; + +#define TNS_MAX_ORDER 20 +/** + * Temporal Noise Shaping + */ +typedef struct { + int present; + int n_filt[8]; + int length[8][4]; + int direction[8][4]; + int order[8][4]; + int coef_res[8]; + int coef_compress[8][4]; + int coef_len[8][4]; + const float *tmp2_map[8][4]; + int coef[8][4][TNS_MAX_ORDER]; +} TemporalNoiseShaping; + +/** + * M/S joint channel coding + */ +typedef struct { + int present; + uint8_t mask[8][64]; +} MidSideStereo; + +/** + * Single Channel Element + * Used for both SCE and LFE elements + */ +typedef struct { + int gain; /**< Channel gain (not used by AAC bitstream). + * Note that this is applied before joint stereo decoding. + * Thus, when used inside CPE elements, both channels must have equal gain. + */ + IndividualChannelStream ics; + TemporalNoiseShaping tns; + Pulse pulse; + int zeroes[8][64]; + int sf_idx[8][64]; + enum Codebook cb[8][64]; ///< codebooks + int cb_run_end[8][64]; ///< codebook run end points + float sf[8][64]; ///< scalefactors + DECLARE_ALIGNED_16(float, coeffs[1024]); ///< coefficients for IMDCT + DECLARE_ALIGNED_16(float, saved[1024]); ///< overlap + DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output + DECLARE_ALIGNED_16(int, icoefs[1024]); ///< integer coefficients for coding +} SingleChannelElement; + +/** + * channel element - generic struct for SCE/CPE/CCE/LFE + */ +typedef struct { + // CPE specific + int common_window; ///< Set if channels share a common 'IndividualChannelStream' in bitstream. + MidSideStereo ms; + // shared + SingleChannelElement ch[2]; + // CCE specific +// ChannelCoupling coup; +} ChannelElement; + +//my stuff + +#define SCALE_ONE_POS 140 ///< scalefactor index that corresponds to scale=1.0 +#define SCALE_MAX_POS 255 ///< scalefactor index maximum value +#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard +#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference + + +#endif + Modified: aacenc/aacenc.c ============================================================================== --- aacenc/aacenc.c (original) +++ aacenc/aacenc.c Thu Jul 31 18:57:15 2008 @@ -30,13 +30,7 @@ #include "mpeg4audio.h" #include "aacpsy.h" - -// XXX: borrowed from aac.c, move to some header eventually -DECLARE_ALIGNED_16(static float, kbd_long_1024[1024]); -DECLARE_ALIGNED_16(static float, kbd_short_128[128]); -DECLARE_ALIGNED_16(static float, sine_long_1024[1024]); -DECLARE_ALIGNED_16(static float, sine_short_128[128]); - +#include "aac.h" #include "aactab.h" static const uint8_t swb_size_1024_96[] = { @@ -118,11 +112,6 @@ static const uint8_t *swb_size_128[] = { swb_size_128_16, swb_size_128_16, swb_size_128_8 }; - -#define MAX_SWB_SIZE 51 - -//borrowed data ends here - #define CB_UNSIGNED 0x01 ///< coefficients are coded as absolute values #define CB_PAIRS 0x02 ///< coefficients are grouped into pairs before coding (quads by default) #define CB_ESCAPE 0x04 ///< codebook allows escapes @@ -174,11 +163,6 @@ typedef struct { AACPsyContext psy; } AACEncContext; -#define SCALE_ONE_POS 140 ///< scalefactor index that corresponds to scale=1.0 -#define SCALE_MAX_POS 255 ///< scalefactor index maximum value -#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard -#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference - /** * Make AAC audio config object. * @see 1.6.2.1 Modified: aacenc/aacpsy.c ============================================================================== --- aacenc/aacpsy.c (original) +++ aacenc/aacpsy.c Thu Jul 31 18:57:15 2008 @@ -32,11 +32,6 @@ static float pow2sf_tab[340]; -#define SCALE_ONE_POS 140 -#define SCALE_MAX_POS 255 -#define SCALE_MAX_DIFF 60 - - /** * Convert coefficients to integers. * @return sum of coefficients Modified: aacenc/aacpsy.h ============================================================================== --- aacenc/aacpsy.h (original) +++ aacenc/aacpsy.h Thu Jul 31 18:57:15 2008 @@ -24,6 +24,7 @@ #include "avcodec.h" #include "dsputil.h" +#include "aac.h" enum AACPsyModelType{ AAC_PSY_NULL, // do nothing on frequencies @@ -33,151 +34,6 @@ enum AACPsyModelType{ AAC_NB_PSY_MODELS }; -// data structures borrowed from aac.c with some minor modifications -/** - * window sequences - */ -enum WindowSequence { - ONLY_LONG_SEQUENCE, - LONG_START_SEQUENCE, - EIGHT_SHORT_SEQUENCE, - LONG_STOP_SEQUENCE, -}; - -/** - * IDs for raw_data_block - */ -enum { - ID_SCE = 0x0, - ID_CPE, - ID_CCE, - ID_LFE, - ID_DSE, - ID_PCE, - ID_FIL, - ID_END -}; - -/** - * special codebooks - */ -enum Codebook { - ZERO_HCB = 0, - FIRST_PAIR_HCB = 5, - ESC_HCB = 11, - NOISE_HCB = 13, - INTENSITY_HCB2 = 14, - INTENSITY_HCB = 15, - ESC_FLAG = 16, -}; - -/** - * pulse tool - */ -typedef struct { - int present; - int num_pulse; - int start; - int offset[4]; - int amp[4]; -} Pulse; - -#define MAX_TAGID 16 - -/** - * Program configuration - describes how channels are arranged. Either read from - * stream (ID_PCE) or created based on a default fixed channel arrangement. - */ -typedef struct { - int che_type[4][MAX_TAGID]; ///< channel element type with the first index as the first 4 raw_data_block IDs - int mono_mixdown; ///< The SCE tag to use if user requests mono output, -1 if not available. - int stereo_mixdown; ///< The CPE tag to use if user requests stereo output, -1 if not available. - int matrix_mixdown; ///< The CPE tag to use if user requests matrixed stereo output, -1 if not available. - int mixdown_coeff_index; ///< 0-3 - int pseudo_surround; ///< Mix surround channels out of phase. -} ProgramConfig; - -/** - * Individual Channel Stream - */ -typedef struct { - int intensity_present; - uint8_t max_sfb; ///< number of scalefactor bands per group - enum WindowSequence window_sequence; - enum WindowSequence window_sequence_prev; - uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window, otherwise use a sinus window. - int num_window_groups; - uint8_t grouping; - uint8_t group_len[8]; - const uint8_t *swb_sizes; - int num_swb; - int num_windows; - int tns_max_bands; -} IndividualChannelStream; - -#define TNS_MAX_ORDER 20 -/** - * Temporal Noise Shaping - */ -typedef struct { - int present; - int n_filt[8]; - int length[8][4]; - int direction[8][4]; - int order[8][4]; - int coef_res[8]; - int coef_compress[8][4]; - int coef_len[8][4]; - const float *tmp2_map[8][4]; - int coef[8][4][TNS_MAX_ORDER]; -} TemporalNoiseShaping; - -/** - * M/S joint channel coding - */ -typedef struct { - int present; - uint8_t mask[8][64]; -} MidSideStereo; - -/** - * Single Channel Element - * Used for both SCE and LFE elements - */ -typedef struct { - int gain; /**< Channel gain (not used by AAC bitstream). - * Note that this is applied before joint stereo decoding. - * Thus, when used inside CPE elements, both channels must have equal gain. - */ - IndividualChannelStream ics; - TemporalNoiseShaping tns; - Pulse pulse; - int zeroes[8][64]; - int sf_idx[8][64]; - enum Codebook cb[8][64]; ///< codebooks - int cb_run_end[8][64]; ///< codebook run end points - float sf[8][64]; ///< scalefactors - DECLARE_ALIGNED_16(float, coeffs[1024]); ///< coefficients for IMDCT - DECLARE_ALIGNED_16(float, saved[1024]); ///< overlap - DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output - DECLARE_ALIGNED_16(int, icoefs[1024]); ///< integer coefficients for coding -} SingleChannelElement; - -/** - * channel element - generic struct for SCE/CPE/CCE/LFE - */ -typedef struct { - // CPE specific - int common_window; ///< Set if channels share a common 'IndividualChannelStream' in bitstream. - MidSideStereo ms; - // shared - SingleChannelElement ch[2]; - // CCE specific -// ChannelCoupling coup; -} ChannelElement; - -// borrowing temporarily ends here - /** * context used by psychoacoustic model */

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[soc]: r2956 - mxf/mxfenc.c
by spyfeng 31 Jul '08

31 Jul '08

Author: spyfeng Date: Thu Jul 31 18:33:52 2008 New Revision: 2956 Log: modified mxf_get_essence_container_ul(). remove the first useless param Modified: mxf/mxfenc.c Modified: mxf/mxfenc.c ============================================================================== --- mxf/mxfenc.c (original) +++ mxf/mxfenc.c Thu Jul 31 18:33:52 2008 @@ -285,8 +285,9 @@ static int klv_encode_ber_length(ByteIOC return 0; } -static const MXFCodecUL *mxf_get_essence_container_ul(const MXFCodecUL *uls, enum CodecID type) +static const MXFCodecUL *mxf_get_essence_container_ul(enum CodecID type) { + const MXFCodecUL *uls = mxf_essence_container_uls; while (uls->id != CODEC_ID_NONE) { if (uls->id == type) break; @@ -811,7 +812,7 @@ static int mxf_write_mpeg_video_desc(AVF av_log(s, AV_LOG_DEBUG, "linked track ID:%d\n", stream_index); #endif - codec_ul = mxf_get_essence_container_ul(mxf_essence_container_uls, st->codec->codec_id); + codec_ul = mxf_get_essence_container_ul(st->codec->codec_id); mxf_write_local_tag(pb, 16, 0x3004); put_buffer(pb, codec_ul->uid, 16); @@ -852,7 +853,7 @@ static int mxf_write_wav_desc(AVFormatCo #ifdef DEBUG PRINT_KEY(s, "wav desc uid", (*refs->track)[stream_index]); #endif - codec_ul = mxf_get_essence_container_ul(mxf_essence_container_uls, st->codec->codec_id); + codec_ul = mxf_get_essence_container_ul(st->codec->codec_id); mxf_write_local_tag(pb, 16, 0x3004); put_buffer(pb, codec_ul->uid, 16);

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[soc]: r2955 - mxf/mxfenc.c
by spyfeng 31 Jul '08

31 Jul '08

Author: spyfeng Date: Thu Jul 31 18:25:09 2008 New Revision: 2955 Log: build essence container uls by checking mxf_essence_container_uls table instead of streams Modified: mxf/mxfenc.c Modified: mxf/mxfenc.c ============================================================================== --- mxf/mxfenc.c (original) +++ mxf/mxfenc.c Thu Jul 31 18:25:09 2008 @@ -941,10 +941,6 @@ static int mxf_write_header_metadata_set static int mxf_add_essence_container_ul(MXFContext *mxf, const MXFCodecUL *codec_ul) { int i; - for (i = 0; i < mxf->essence_container_count; i++) { - if (!memcmp(mxf->essence_container_uls[i], codec_ul->uid, 16)) - return 0; - } mxf->essence_container_uls = av_realloc(mxf->essence_container_uls, (mxf->essence_container_count + 1) * 16); if (!mxf->essence_container_uls) return -1; @@ -960,14 +956,15 @@ static int mxf_build_essence_container_r int i; const MXFCodecUL *codec_ul = NULL; - for (i = 0; i < s->nb_streams; i++) { - st = s->streams[i]; - codec_ul = mxf_get_essence_container_ul(mxf_essence_container_uls, st->codec->codec_id); - if (codec_ul) { - if (mxf_add_essence_container_ul(mxf, codec_ul) < 0 ) - return -1; - } else - return -1; + for (codec_ul = mxf_essence_container_uls; codec_ul->id; codec_ul++) { + for (i = 0; i < s->nb_streams; i++) { + st = s->streams[i]; + if (st->codec->codec_id == codec_ul->id) { + if (mxf_add_essence_container_ul(mxf, codec_ul) < 0 ) + return -1; + break; + } + } } return 0; }

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