/* * AAC decoder * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com> * * AAC LATM decoder * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz> * Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net> * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * AAC decoder * @author Oded Shimon ( ods15 ods15 dyndns org ) * @author Maxim Gavrilov ( maxim.gavrilov gmail com ) */ #define FFT_FLOAT 1 #define USE_FIXED 0 #include "libavutil/float_dsp.h" #include "libavutil/opt.h" #include "avcodec.h" #include "codec_internal.h" #include "get_bits.h" #include "fft.h" #include "mdct15.h" #include "lpc.h" #include "kbdwin.h" #include "sinewin.h" #include "aac.h" #include "aactab.h" #include "aacdectab.h" #include "adts_header.h" #include "cbrt_data.h" #include "sbr.h" #include "aacsbr.h" #include "mpeg4audio.h" #include "profiles.h" #include "libavutil/intfloat.h" #include <errno.h> #include <math.h> #include <stdint.h> #include <string.h> #if ARCH_ARM # include "arm/aac.h" #elif ARCH_MIPS # include "mips/aacdec_mips.h" #endif DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(sine_120))[120]; DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(sine_960))[960]; DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(aac_kbd_long_960))[960]; DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(aac_kbd_short_120))[120]; static av_always_inline 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; } #ifndef VMUL2 static inline float *VMUL2(float *dst, const float *v, unsigned idx, const float *scale) { float s = *scale; *dst++ = v[idx & 15] * s; *dst++ = v[idx>>4 & 15] * s; return dst; } #endif #ifndef VMUL4 static inline float *VMUL4(float *dst, const float *v, unsigned idx, const float *scale) { float s = *scale; *dst++ = v[idx & 3] * s; *dst++ = v[idx>>2 & 3] * s; *dst++ = v[idx>>4 & 3] * s; *dst++ = v[idx>>6 & 3] * s; return dst; } #endif #ifndef VMUL2S static inline float *VMUL2S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale) { union av_intfloat32 s0, s1; s0.f = s1.f = *scale; s0.i ^= sign >> 1 << 31; s1.i ^= sign << 31; *dst++ = v[idx & 15] * s0.f; *dst++ = v[idx>>4 & 15] * s1.f; return dst; } #endif #ifndef VMUL4S static inline float *VMUL4S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale) { unsigned nz = idx >> 12; union av_intfloat32 s = { .f = *scale }; union av_intfloat32 t; t.i = s.i ^ (sign & 1U<<31); *dst++ = v[idx & 3] * t.f; sign <<= nz & 1; nz >>= 1; t.i = s.i ^ (sign & 1U<<31); *dst++ = v[idx>>2 & 3] * t.f; sign <<= nz & 1; nz >>= 1; t.i = s.i ^ (sign & 1U<<31); *dst++ = v[idx>>4 & 3] * t.f; sign <<= nz & 1; t.i = s.i ^ (sign & 1U<<31); *dst++ = v[idx>>6 & 3] * t.f; return dst; } #endif static av_always_inline float flt16_round(float pf) { union av_intfloat32 tmp; tmp.f = pf; tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U; return tmp.f; } static av_always_inline float flt16_even(float pf) { union av_intfloat32 tmp; tmp.f = pf; tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U; return tmp.f; } static av_always_inline float flt16_trunc(float pf) { union av_intfloat32 pun; pun.f = pf; pun.i &= 0xFFFF0000U; return pun.f; } static av_always_inline void predict(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; float r0 = ps->r0, r1 = ps->r1; float cor0 = ps->cor0, cor1 = ps->cor1; float var0 = ps->var0, var1 = ps->var1; k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0; k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0; pv = flt16_round(k1 * r0 + k2 * r1); if (output_enable) *coef += pv; e0 = *coef; e1 = e0 - k1 * r0; ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1); ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1)); ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0); ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0)); ps->r1 = flt16_trunc(a * (r0 - k1 * e0)); ps->r0 = flt16_trunc(a * e0); } /** * Apply dependent channel coupling (applied before IMDCT). * * @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; int g, i, group, k, idx = 0; if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) { av_log(ac->avctx, AV_LOG_ERROR, "Dependent coupling is not supported together with LTP\n"); return; } for (g = 0; g < ics->num_window_groups; g++) { for (i = 0; i < ics->max_sfb; i++, idx++) { 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++) { // FIXME: SIMDify dest[group * 128 + k] += gain * src[group * 128 + k]; } } } } dest += ics->group_len[g] * 128; src += ics->group_len[g] * 128; } } /** * Apply independent channel coupling (applied after IMDCT). * * @param index index into coupling gain array */ static void apply_independent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index) { const float gain = cce->coup.gain[index][0]; const float *src = cce->ch[0].ret; float *dest = target->ret; const int len = 1024 << (ac->oc[1].m4ac.sbr == 1); ac->fdsp->vector_fmac_scalar(dest, src, gain, len); } #include "aacdec_template.c" #define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word struct LATMContext { AACContext aac_ctx; ///< containing AACContext int initialized; ///< initialized after a valid extradata was seen // parser data int audio_mux_version_A; ///< LATM syntax version int frame_length_type; ///< 0/1 variable/fixed frame length int frame_length; ///< frame length for fixed frame length }; static inline uint32_t latm_get_value(GetBitContext *b) { int length = get_bits(b, 2); return get_bits_long(b, (length+1)*8); } static int latm_decode_audio_specific_config(struct LATMContext *latmctx, GetBitContext *gb, int asclen) { AACContext *ac = &latmctx->aac_ctx; AVCodecContext *avctx = ac->avctx; MPEG4AudioConfig m4ac = { 0 }; GetBitContext gbc; int config_start_bit = get_bits_count(gb); int sync_extension = 0; int bits_consumed, esize, i; if (asclen > 0) { sync_extension = 1; asclen = FFMIN(asclen, get_bits_left(gb)); init_get_bits(&gbc, gb->buffer, config_start_bit + asclen); skip_bits_long(&gbc, config_start_bit); } else if (asclen == 0) { gbc = *gb; } else { return AVERROR_INVALIDDATA; } if (get_bits_left(gb) <= 0) return AVERROR_INVALIDDATA; bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac, &gbc, config_start_bit, sync_extension); if (bits_consumed < config_start_bit) return AVERROR_INVALIDDATA; bits_consumed -= config_start_bit; if (asclen == 0) asclen = bits_consumed; if (!latmctx->initialized || ac->oc[1].m4ac.sample_rate != m4ac.sample_rate || ac->oc[1].m4ac.chan_config != m4ac.chan_config) { if (latmctx->initialized) { av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", m4ac.sample_rate, m4ac.chan_config); } else { av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n"); } latmctx->initialized = 0; esize = (asclen + 7) / 8; if (avctx->extradata_size < esize) { av_free(avctx->extradata); avctx->extradata = av_malloc(esize + AV_INPUT_BUFFER_PADDING_SIZE); if (!avctx->extradata) return AVERROR(ENOMEM); } avctx->extradata_size = esize; gbc = *gb; for (i = 0; i < esize; i++) { avctx->extradata[i] = get_bits(&gbc, 8); } memset(avctx->extradata+esize, 0, AV_INPUT_BUFFER_PADDING_SIZE); } skip_bits_long(gb, asclen); return 0; } static int read_stream_mux_config(struct LATMContext *latmctx, GetBitContext *gb) { int ret, audio_mux_version = get_bits(gb, 1); latmctx->audio_mux_version_A = 0; if (audio_mux_version) latmctx->audio_mux_version_A = get_bits(gb, 1); if (!latmctx->audio_mux_version_A) { if (audio_mux_version) latm_get_value(gb); // taraFullness skip_bits(gb, 1); // allStreamSameTimeFraming skip_bits(gb, 6); // numSubFrames // numPrograms if (get_bits(gb, 4)) { // numPrograms avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs"); return AVERROR_PATCHWELCOME; } // for each program (which there is only one in DVB) // for each layer (which there is only one in DVB) if (get_bits(gb, 3)) { // numLayer avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers"); return AVERROR_PATCHWELCOME; } // for all but first stream: use_same_config = get_bits(gb, 1); if (!audio_mux_version) { if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0) return ret; } else { int ascLen = latm_get_value(gb); if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0) return ret; } latmctx->frame_length_type = get_bits(gb, 3); switch (latmctx->frame_length_type) { case 0: skip_bits(gb, 8); // latmBufferFullness break; case 1: latmctx->frame_length = get_bits(gb, 9); break; case 3: case 4: case 5: skip_bits(gb, 6); // CELP frame length table index break; case 6: case 7: skip_bits(gb, 1); // HVXC frame length table index break; } if (get_bits(gb, 1)) { // other data if (audio_mux_version) { latm_get_value(gb); // other_data_bits } else { int esc; do { if (get_bits_left(gb) < 9) return AVERROR_INVALIDDATA; esc = get_bits(gb, 1); skip_bits(gb, 8); } while (esc); } } if (get_bits(gb, 1)) // crc present skip_bits(gb, 8); // config_crc } return 0; } static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb) { uint8_t tmp; if (ctx->frame_length_type == 0) { int mux_slot_length = 0; do { if (get_bits_left(gb) < 8) return AVERROR_INVALIDDATA; tmp = get_bits(gb, 8); mux_slot_length += tmp; } while (tmp == 255); return mux_slot_length; } else if (ctx->frame_length_type == 1) { return ctx->frame_length; } else if (ctx->frame_length_type == 3 || ctx->frame_length_type == 5 || ctx->frame_length_type == 7) { skip_bits(gb, 2); // mux_slot_length_coded } return 0; } static int read_audio_mux_element(struct LATMContext *latmctx, GetBitContext *gb) { int err; uint8_t use_same_mux = get_bits(gb, 1); if (!use_same_mux) { if ((err = read_stream_mux_config(latmctx, gb)) < 0) return err; } else if (!latmctx->aac_ctx.avctx->extradata) { av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG, "no decoder config found\n"); return 1; } if (latmctx->audio_mux_version_A == 0) { int mux_slot_length_bytes = read_payload_length_info(latmctx, gb); if (mux_slot_length_bytes < 0 || mux_slot_length_bytes * 8LL > get_bits_left(gb)) { av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n"); return AVERROR_INVALIDDATA; } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) { av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "frame length mismatch %d << %d\n", mux_slot_length_bytes * 8, get_bits_left(gb)); return AVERROR_INVALIDDATA; } } return 0; } static int latm_decode_frame(AVCodecContext *avctx, AVFrame *out, int *got_frame_ptr, AVPacket *avpkt) { struct LATMContext *latmctx = avctx->priv_data; int muxlength, err; GetBitContext gb; if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0) return err; // check for LOAS sync word if (get_bits(&gb, 11) != LOAS_SYNC_WORD) return AVERROR_INVALIDDATA; muxlength = get_bits(&gb, 13) + 3; // not enough data, the parser should have sorted this out if (muxlength > avpkt->size) return AVERROR_INVALIDDATA; if ((err = read_audio_mux_element(latmctx, &gb))) return (err < 0) ? err : avpkt->size; if (!latmctx->initialized) { if (!avctx->extradata) { *got_frame_ptr = 0; return avpkt->size; } else { push_output_configuration(&latmctx->aac_ctx); if ((err = decode_audio_specific_config( &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac, avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) { pop_output_configuration(&latmctx->aac_ctx); return err; } latmctx->initialized = 1; } } if (show_bits(&gb, 12) == 0xfff) { av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "ADTS header detected, probably as result of configuration " "misparsing\n"); return AVERROR_INVALIDDATA; } switch (latmctx->aac_ctx.oc[1].m4ac.object_type) { case AOT_ER_AAC_LC: case AOT_ER_AAC_LTP: case AOT_ER_AAC_LD: case AOT_ER_AAC_ELD: err = aac_decode_er_frame(avctx, out, got_frame_ptr, &gb); break; default: err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt); } if (err < 0) return err; return muxlength; } static av_cold int latm_decode_init(AVCodecContext *avctx) { struct LATMContext *latmctx = avctx->priv_data; int ret = aac_decode_init(avctx); if (avctx->extradata_size > 0) latmctx->initialized = !ret; return ret; } const FFCodec ff_aac_decoder = { .p.name = "aac", .p.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"), .p.type = AVMEDIA_TYPE_AUDIO, .p.id = AV_CODEC_ID_AAC, .priv_data_size = sizeof(AACContext), .init = aac_decode_init, .close = aac_decode_close, FF_CODEC_DECODE_CB(aac_decode_frame), .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1, .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, #if FF_API_OLD_CHANNEL_LAYOUT .p.channel_layouts = aac_channel_layout, #endif .p.ch_layouts = aac_ch_layout, .flush = flush, .p.priv_class = &aac_decoder_class, .p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles), }; /* Note: This decoder filter is intended to decode LATM streams transferred in MPEG transport streams which only contain one program. To do a more complex LATM demuxing a separate LATM demuxer should be used. */ const FFCodec ff_aac_latm_decoder = { .p.name = "aac_latm", .p.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"), .p.type = AVMEDIA_TYPE_AUDIO, .p.id = AV_CODEC_ID_AAC_LATM, .priv_data_size = sizeof(struct LATMContext), .init = latm_decode_init, .close = aac_decode_close, FF_CODEC_DECODE_CB(latm_decode_frame), .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1, .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, #if FF_API_OLD_CHANNEL_LAYOUT .p.channel_layouts = aac_channel_layout, #endif .p.ch_layouts = aac_ch_layout, .flush = flush, .p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles), };