| /* |
| * 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 FFT_FIXED_32 0 |
| #define USE_FIXED 0 |
| |
| #include "libavutil/float_dsp.h" |
| #include "libavutil/opt.h" |
| #include "avcodec.h" |
| #include "internal.h" |
| #include "get_bits.h" |
| #include "fft.h" |
| #include "imdct15.h" |
| #include "lpc.h" |
| #include "kbdwin.h" |
| #include "sinewin.h" |
| |
| #include "aac.h" |
| #include "aactab.h" |
| #include "aacdectab.h" |
| #include "cbrt_tablegen.h" |
| #include "sbr.h" |
| #include "aacsbr.h" |
| #include "mpeg4audio.h" |
| #include "aacadtsdec.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 |
| |
| 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) |
| { |
| int i; |
| 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); |
| |
| for (i = 0; i < len; i++) |
| dest[i] += gain * src[i]; |
| } |
| |
| #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 }; |
| int config_start_bit = get_bits_count(gb); |
| int sync_extension = 0; |
| int bits_consumed, esize; |
| |
| if (asclen) { |
| sync_extension = 1; |
| asclen = FFMIN(asclen, get_bits_left(gb)); |
| } else |
| asclen = get_bits_left(gb); |
| |
| if (config_start_bit % 8) { |
| avpriv_request_sample(latmctx->aac_ctx.avctx, |
| "Non-byte-aligned audio-specific config"); |
| return AVERROR_PATCHWELCOME; |
| } |
| if (asclen <= 0) |
| return AVERROR_INVALIDDATA; |
| bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac, |
| gb->buffer + (config_start_bit / 8), |
| asclen, sync_extension); |
| |
| if (bits_consumed < 0) |
| return AVERROR_INVALIDDATA; |
| |
| 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\n"); |
| } else { |
| av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n"); |
| } |
| latmctx->initialized = 0; |
| |
| esize = (bits_consumed+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; |
| memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize); |
| memset(avctx->extradata+esize, 0, AV_INPUT_BUFFER_PADDING_SIZE); |
| } |
| skip_bits_long(gb, bits_consumed); |
| |
| return bits_consumed; |
| } |
| |
| 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; |
| ascLen -= ret; |
| skip_bits_long(gb, ascLen); |
| } |
| |
| 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 { |
| 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 { |
| 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 AVERROR(EAGAIN); |
| } |
| if (latmctx->audio_mux_version_A == 0) { |
| int mux_slot_length_bytes = read_payload_length_info(latmctx, gb); |
| if (mux_slot_length_bytes * 8 > 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, void *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)) < 0) |
| return err; |
| |
| 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; |
| } |
| |
| AVCodec ff_aac_decoder = { |
| .name = "aac", |
| .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"), |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = AV_CODEC_ID_AAC, |
| .priv_data_size = sizeof(AACContext), |
| .init = aac_decode_init, |
| .close = aac_decode_close, |
| .decode = aac_decode_frame, |
| .sample_fmts = (const enum AVSampleFormat[]) { |
| AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE |
| }, |
| .capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1, |
| .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, |
| .channel_layouts = aac_channel_layout, |
| .flush = flush, |
| .priv_class = &aac_decoder_class, |
| .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. |
| */ |
| AVCodec ff_aac_latm_decoder = { |
| .name = "aac_latm", |
| .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"), |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = AV_CODEC_ID_AAC_LATM, |
| .priv_data_size = sizeof(struct LATMContext), |
| .init = latm_decode_init, |
| .close = aac_decode_close, |
| .decode = latm_decode_frame, |
| .sample_fmts = (const enum AVSampleFormat[]) { |
| AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE |
| }, |
| .capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1, |
| .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, |
| .channel_layouts = aac_channel_layout, |
| .flush = flush, |
| .profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles), |
| }; |
