From 9cc04edff9fff54416bc9fe0f55dd65ed5deed60 Mon Sep 17 00:00:00 2001 From: Robert Swain Date: Mon, 11 Aug 2008 11:16:06 +0000 Subject: More OKed hunks of the AAC decoder from SoC Originally committed as revision 14694 to svn://svn.ffmpeg.org/ffmpeg/trunk --- libavcodec/aac.c | 408 +++++++++++++++++++++++++++++++++++++++++++++++++++- libavcodec/aac.h | 48 +++++++ libavcodec/aactab.c | 15 ++ libavcodec/aactab.h | 12 ++ 4 files changed, 482 insertions(+), 1 deletion(-) diff --git a/libavcodec/aac.c b/libavcodec/aac.c index cd59c2031a..d28e1d7df9 100644 --- a/libavcodec/aac.c +++ b/libavcodec/aac.c @@ -99,6 +99,40 @@ static VLC vlc_scalefactors; static VLC vlc_spectral[11]; +/** + * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit. + * + * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present. + * @param sce_map mono (Single Channel Element) map + * @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 *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map + map[get_bits(gb, 4)] = type; + } +} + +/** + * Decode program configuration element; reference: table 4.2. + * + * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. + * + * @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) { + int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc; + + skip_bits(gb, 2); // object_type + + ac->m4ac.sampling_index = get_bits(gb, 4); + if(ac->m4ac.sampling_index > 11) { + av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); + return -1; + } + ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index]; num_front = get_bits(gb, 4); num_side = get_bits(gb, 4); num_back = get_bits(gb, 4); @@ -130,6 +164,131 @@ static VLC vlc_spectral[11]; return 0; } +/** + * Set up channel positions based on a default channel configuration + * as specified in table 1.17. + * + * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. + * + * @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) +{ + if(channel_config < 1 || channel_config > 7) { + av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", + channel_config); + return -1; + } + + /* default channel configurations: + * + * 1ch : front center (mono) + * 2ch : L + R (stereo) + * 3ch : front center + L + R + * 4ch : front center + L + R + back center + * 5ch : front center + L + R + back stereo + * 6ch : front center + L + R + back stereo + LFE + * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE + */ + + if(channel_config != 2) + new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) + if(channel_config > 1) + new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) + if(channel_config == 4) + new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center + if(channel_config > 4) + new_che_pos[TYPE_CPE][(channel_config == 7) + 1] + = AAC_CHANNEL_BACK; // back stereo + if(channel_config > 5) + new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE + if(channel_config == 7) + new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right + + return 0; +} + + return -1; + } + + if (get_bits1(gb)) // dependsOnCoreCoder + 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) + 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))) + return ret; + } else { + 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))) + 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 + * aacScalefactorDataResilienceFlag + * aacSpectralDataResilienceFlag + */ + break; + } + skip_bits1(gb); // extensionFlag3 (TBD in version 3) + } + return 0; +} + +/** + * Decode audio specific configuration; reference: table 1.13. + * + * @param data pointer to AVCodecContext extradata + * @param data_size size of AVCCodecContext extradata + * + * @return Returns error status. 0 - OK, !0 - error + */ +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) + return -1; + if(ac->m4ac.sampling_index > 11) { + av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); + return -1; + } + + skip_bits_long(&gb, i); + + switch (ac->m4ac.object_type) { + case AOT_AAC_LC: + if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config)) + return -1; + break; + default: + av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", + ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); + return -1; + } + return 0; +} + static av_cold int aac_decode_init(AVCodecContext * avccontext) { AACContext * ac = avccontext->priv_data; int i; @@ -140,6 +299,7 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) { decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size)) return -1; + avccontext->sample_fmt = SAMPLE_FMT_S16; avccontext->sample_rate = ac->m4ac.sample_rate; avccontext->frame_size = 1024; @@ -157,6 +317,8 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) { dsputil_init(&ac->dsp, avccontext); + ac->random_state = 0x1f2e3d4c; + // -1024 - Compensate wrong IMDCT method. // 32768 - Required to scale values to the correct range for the bias method // for float to int16 conversion. @@ -188,6 +350,10 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) { return 0; } +/** + * Skip data_stream_element; reference: table 4.10. + */ +static void skip_data_stream_element(GetBitContext * gb) { int byte_align = get_bits1(gb); int count = get_bits(gb, 8); if (count == 255) @@ -197,6 +363,27 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) { skip_bits_long(gb, 8 * count); } +/** + * Decode Individual Channel Stream info; reference: table 4.6. + * + * @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) { + if (get_bits1(gb)) { + av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n"); + memset(ics, 0, sizeof(IndividualChannelStream)); + return -1; + } + 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; + + return 0; +} + /** * inverse quantization * @@ -210,6 +397,15 @@ static inline float ivquant(int a) { return cbrtf(fabsf(a)) * a; } +/** + * Decode band types (section_data payload); reference: table 4.46. + * + * @param band_type array of the used band type + * @param band_type_run_end array of the last scalefactor band of a band type run + * + * @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) { int g, idx = 0; const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; @@ -232,7 +428,13 @@ static inline float ivquant(int a) { sect_len, ics->max_sfb); return -1; } + } + } + return 0; +} +/** + * Decode scalefactors; reference: table 4.47. * * @param mix_gain channel gain (Not used by AAC bitstream.) * @param global_gain first scalefactor value as scalefactors are differentially coded @@ -313,6 +515,16 @@ static void decode_pulses(Pulse * pulse, GetBitContext * gb) { } } +/** + * Decode Mid/Side data; reference: table 4.54. + * + * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; + * [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) { + /** * Add pulses with particular amplitudes to the quantized spectral data; reference: 4.6.3.3. * @@ -330,10 +542,109 @@ static void add_pulses(int icoef[1024], const Pulse * pulse, const IndividualCha } /** - * Parse Spectral Band Replication extension data; reference: table 4.55. + * Decode an individual_channel_stream payload; reference: table 4.44. + * + * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. + * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.) + * + * @return Returns error status. 0 - OK, !0 - error + */ +static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) { + int icoeffs[1024]; + Pulse pulse; + TemporalNoiseShaping * tns = &sce->tns; + IndividualChannelStream * ics = &sce->ics; + float * out = sce->coeffs; + int global_gain, pulse_present = 0; + + /* These two assignments are to silence some GCC warnings about the + * variables being used uninitialised when in fact they always are. + */ + pulse.num_pulse = 0; + pulse.start = 0; + + global_gain = get_bits(gb, 8); + + if (!common_window && !scale_flag) { + if (decode_ics_info(ac, ics, gb, 0) < 0) + return -1; + } + + if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) + return -1; + if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) + return -1; + + pulse_present = 0; + if (!scale_flag) { + if ((pulse_present = get_bits1(gb))) { + 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; + } + decode_pulses(&pulse, gb); + } + if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics)) + return -1; + if (get_bits1(gb)) { + av_log_missing_feature(ac->avccontext, "SSR", 1); + return -1; + } + } + + if (decode_spectrum(ac, icoeffs, gb, ics, sce->band_type) < 0) + return -1; + if (pulse_present) + add_pulses(icoeffs, &pulse, ics); + dequant(ac, out, icoeffs, sce->sf, ics, sce->band_type); + return 0; +} + +/** + * Decode a channel_pair_element; reference: table 4.4. + * + * @param elem_id Identifies the instance of a syntax element. + * + * @return Returns error status. 0 - OK, !0 - error + */ +static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) { + int i, ret, common_window, ms_present = 0; + ChannelElement * cpe; + + cpe = ac->che[TYPE_CPE][elem_id]; + common_window = get_bits1(gb); + if (common_window) { + if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) + return -1; + i = cpe->ch[1].ics.use_kb_window[0]; + 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) { + av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); + return -1; + } else if(ms_present) + decode_mid_side_stereo(cpe, gb, ms_present); + } + if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) + return ret; + if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) + return ret; + + if (common_window && ms_present) + apply_mid_side_stereo(cpe); + + if (cpe->ch[1].ics.intensity_present) + apply_intensity_stereo(cpe, ms_present); + return 0; +} + +/** + * Decode Spectral Band Replication extension data; reference: table 4.55. * * @param crc flag indicating the presence of CRC checksum * @param cnt length of TYPE_FIL syntactic element in bytes + * * @return Returns number of bytes consumed from the TYPE_FIL element. */ static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) { @@ -343,6 +654,66 @@ static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, in return cnt; } +/** + * Decode dynamic range information; reference: table 4.52. + * + * @param cnt length of TYPE_FIL syntactic element in bytes + * + * @return Returns number of bytes consumed. + */ +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)) { + che_drc->pce_instance_tag = get_bits(gb, 4); + skip_bits(gb, 4); // tag_reserved_bits + n++; + } + + /* excluded_chns_present? */ + if(get_bits1(gb)) { + n += decode_drc_channel_exclusions(che_drc, gb); + } + + /* drc_bands_present? */ + if (get_bits1(gb)) { + che_drc->band_incr = get_bits(gb, 4); + che_drc->interpolation_scheme = get_bits(gb, 4); + n++; + drc_num_bands += che_drc->band_incr; + for (i = 0; i < drc_num_bands; i++) { + che_drc->band_top[i] = get_bits(gb, 8); + n++; + } + } + + /* prog_ref_level_present? */ + if (get_bits1(gb)) { + che_drc->prog_ref_level = get_bits(gb, 7); + skip_bits1(gb); // prog_ref_level_reserved_bits + n++; + } + + for (i = 0; i < drc_num_bands; i++) { + che_drc->dyn_rng_sgn[i] = get_bits1(gb); + che_drc->dyn_rng_ctl[i] = get_bits(gb, 7); + n++; + } + + return n; +} + +/** + * Decode extension data (incomplete); reference: table 4.51. + * + * @param cnt length of TYPE_FIL syntactic element in bytes + * + * @return Returns number of bytes consumed + */ +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 @@ -364,6 +735,21 @@ static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, in return res; } +/** + * 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 * lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024; + const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128; + const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024; + const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128; + float * buf = ac->buf_mdct; + int i; + /** * Apply dependent channel coupling (applied before IMDCT). * @@ -409,6 +795,26 @@ static void apply_independent_coupling(AACContext * ac, SingleChannelElement * s sce->ret[i] += gain * (cc->ch[0].ret[i] - ac->add_bias); } + if (!ac->is_saved) { + ac->is_saved = 1; + *data_size = 0; + return 0; + } + + data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t); + 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); + return -1; + } + *data_size = data_size_tmp; + + ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels); + + return buf_size; +} + static av_cold int aac_decode_close(AVCodecContext * avccontext) { AACContext * ac = avccontext->priv_data; int i, j; diff --git a/libavcodec/aac.h b/libavcodec/aac.h index ebf2218c8a..aafb8261be 100644 --- a/libavcodec/aac.h +++ b/libavcodec/aac.h @@ -43,6 +43,7 @@ size); #define MAX_CHANNELS 64 +#define MAX_ELEM_ID 16 #define IVQUANT_SIZE 1024 @@ -76,6 +77,17 @@ enum AudioObjectType { AOT_SSC, ///< N SinuSoidal Coding }; +enum RawDataBlockType { + TYPE_SCE, + TYPE_CPE, + TYPE_CCE, + TYPE_LFE, + TYPE_DSE, + TYPE_PCE, + TYPE_FIL, + TYPE_END, +}; + enum ExtensionPayloadID { EXT_FILL, EXT_FILL_DATA, @@ -111,6 +123,35 @@ enum ChannelPosition { AAC_CHANNEL_CC = 5, }; +/** + * The point during decoding at which channel coupling is applied. + */ +enum CouplingPoint { + BEFORE_TNS, + BETWEEN_TNS_AND_IMDCT, + AFTER_IMDCT = 3, +}; + +/** + * Individual Channel Stream + */ + +/** + * Dynamic Range Control - decoded from the bitstream but not processed further. + */ +typedef struct { + 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 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; + typedef struct { int num_pulse; int start; @@ -134,9 +175,15 @@ typedef struct { int is_saved; ///< Set if elements have stored overlap from previous frame. DynamicRangeControl che_drc; + /** + * @defgroup elements + * @{ + */ enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the * first index as the first 4 raw data block types */ + ChannelElement * che[4][MAX_ELEM_ID]; + /** @} */ /** * @defgroup tables Computed / set up during initialization. @@ -145,6 +192,7 @@ typedef struct { MDCTContext mdct; MDCTContext mdct_small; DSPContext dsp; + int random_state; /** @} */ /** diff --git a/libavcodec/aactab.c b/libavcodec/aactab.c index a404567eea..ac1d59f0c2 100644 --- a/libavcodec/aactab.c +++ b/libavcodec/aactab.c @@ -32,6 +32,14 @@ #include +const uint8_t ff_aac_num_swb_1024[] = { + 41, 41, 47, 49, 49, 51, 47, 47, 43, 43, 43, 40 +}; + +const uint8_t ff_aac_num_swb_128[] = { + 12, 12, 12, 14, 14, 14, 15, 15, 15, 15, 15, 15 +}; + const uint32_t ff_aac_scalefactor_code[121] = { 0x3ffe8, 0x3ffe6, 0x3ffe7, 0x3ffe5, 0x7fff5, 0x7fff1, 0x7ffed, 0x7fff6, 0x7ffee, 0x7ffef, 0x7fff0, 0x7fffc, 0x7fffd, 0x7ffff, 0x7fffe, 0x7fff7, @@ -796,6 +804,13 @@ const float ff_aac_ivquant_tab[IVQUANT_SIZE] = { 4064.0312908, 4074.6805676, 4085.3368071, 4096.0000000, }; +/** + * Table of pow(2, (i - 200)/4.) used for different purposes depending on the + * range of indices to the table: + * [ 0, 255] scale factor decoding when using C dsp.float_to_int16 + * [60, 315] scale factor decoding when using SIMD dsp.float_to_int16 + * [45, 300] intensity stereo position decoding mapped in reverse order i.e. 0->300, 1->299, ..., 254->46, 255->45 + */ const float ff_aac_pow2sf_tab[316] = { 8.88178420e-16, 1.05622810e-15, 1.25607397e-15, 1.49373210e-15, 1.77635684e-15, 2.11245619e-15, 2.51214793e-15, 2.98746420e-15, diff --git a/libavcodec/aactab.h b/libavcodec/aactab.h index e7f14e6ed6..198b61c81a 100644 --- a/libavcodec/aactab.h +++ b/libavcodec/aactab.h @@ -35,6 +35,18 @@ #include +/* NOTE: + * Tables in this file are used by the AAC decoder and will be used by the AAC + * encoder. + */ + +/* @name number of scalefactor window bands for long and short transform windows respectively + * @{ + */ +extern const uint8_t ff_aac_num_swb_1024[]; +extern const uint8_t ff_aac_num_swb_128 []; +// @} + extern const uint32_t ff_aac_scalefactor_code[121]; extern const uint8_t ff_aac_scalefactor_bits[121]; -- cgit v1.2.3