361 lines
13 KiB
C
Executable file
361 lines
13 KiB
C
Executable file
/*
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* Bluetooth low-complexity, subband codec (SBC)
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*
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* Copyright (C) 2017 Aurelien Jacobs <aurel@gnuage.org>
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* Copyright (C) 2012-2013 Intel Corporation
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* Copyright (C) 2008-2010 Nokia Corporation
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* Copyright (C) 2004-2010 Marcel Holtmann <marcel@holtmann.org>
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* Copyright (C) 2004-2005 Henryk Ploetz <henryk@ploetzli.ch>
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* Copyright (C) 2005-2008 Brad Midgley <bmidgley@xmission.com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* SBC encoder implementation
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*/
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "internal.h"
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#include "profiles.h"
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#include "put_bits.h"
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#include "sbc.h"
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#include "sbcdsp.h"
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typedef struct SBCEncContext {
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AVClass *class;
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int64_t max_delay;
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int msbc;
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DECLARE_ALIGNED(SBC_ALIGN, struct sbc_frame, frame);
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DECLARE_ALIGNED(SBC_ALIGN, SBCDSPContext, dsp);
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} SBCEncContext;
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static int sbc_analyze_audio(SBCDSPContext *s, struct sbc_frame *frame)
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{
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int ch, blk;
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int16_t *x;
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switch (frame->subbands) {
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case 4:
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for (ch = 0; ch < frame->channels; ch++) {
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x = &s->X[ch][s->position - 4 *
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s->increment + frame->blocks * 4];
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for (blk = 0; blk < frame->blocks;
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blk += s->increment) {
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s->sbc_analyze_4s(
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s, x,
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frame->sb_sample_f[blk][ch],
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frame->sb_sample_f[blk + 1][ch] -
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frame->sb_sample_f[blk][ch]);
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x -= 4 * s->increment;
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}
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}
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return frame->blocks * 4;
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case 8:
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for (ch = 0; ch < frame->channels; ch++) {
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x = &s->X[ch][s->position - 8 *
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s->increment + frame->blocks * 8];
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for (blk = 0; blk < frame->blocks;
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blk += s->increment) {
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s->sbc_analyze_8s(
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s, x,
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frame->sb_sample_f[blk][ch],
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frame->sb_sample_f[blk + 1][ch] -
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frame->sb_sample_f[blk][ch]);
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x -= 8 * s->increment;
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}
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}
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return frame->blocks * 8;
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default:
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return AVERROR(EIO);
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}
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}
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/*
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* Packs the SBC frame from frame into the memory in avpkt.
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* Returns the length of the packed frame.
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*/
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static size_t sbc_pack_frame(AVPacket *avpkt, struct sbc_frame *frame,
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int joint, int msbc)
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{
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PutBitContext pb;
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/* Will copy the header parts for CRC-8 calculation here */
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uint8_t crc_header[11] = { 0 };
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int crc_pos;
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uint32_t audio_sample;
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int ch, sb, blk; /* channel, subband, block and bit counters */
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int bits[2][8]; /* bits distribution */
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uint32_t levels[2][8]; /* levels are derived from that */
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uint32_t sb_sample_delta[2][8];
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if (msbc) {
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avpkt->data[0] = MSBC_SYNCWORD;
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avpkt->data[1] = 0;
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avpkt->data[2] = 0;
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} else {
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avpkt->data[0] = SBC_SYNCWORD;
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avpkt->data[1] = (frame->frequency & 0x03) << 6;
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avpkt->data[1] |= (((frame->blocks >> 2) - 1) & 0x03) << 4;
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avpkt->data[1] |= (frame->mode & 0x03) << 2;
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avpkt->data[1] |= (frame->allocation & 0x01) << 1;
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avpkt->data[1] |= ((frame->subbands == 8) & 0x01) << 0;
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avpkt->data[2] = frame->bitpool;
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if (frame->bitpool > frame->subbands << (4 + (frame->mode == STEREO
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|| frame->mode == JOINT_STEREO)))
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return -5;
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}
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/* Can't fill in crc yet */
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crc_header[0] = avpkt->data[1];
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crc_header[1] = avpkt->data[2];
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crc_pos = 16;
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init_put_bits(&pb, avpkt->data + 4, avpkt->size);
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if (frame->mode == JOINT_STEREO) {
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put_bits(&pb, frame->subbands, joint);
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crc_header[crc_pos >> 3] = joint;
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crc_pos += frame->subbands;
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}
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for (ch = 0; ch < frame->channels; ch++) {
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for (sb = 0; sb < frame->subbands; sb++) {
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put_bits(&pb, 4, frame->scale_factor[ch][sb] & 0x0F);
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crc_header[crc_pos >> 3] <<= 4;
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crc_header[crc_pos >> 3] |= frame->scale_factor[ch][sb] & 0x0F;
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crc_pos += 4;
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}
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}
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/* align the last crc byte */
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if (crc_pos % 8)
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crc_header[crc_pos >> 3] <<= 8 - (crc_pos % 8);
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avpkt->data[3] = ff_sbc_crc8(frame->crc_ctx, crc_header, crc_pos);
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ff_sbc_calculate_bits(frame, bits);
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for (ch = 0; ch < frame->channels; ch++) {
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for (sb = 0; sb < frame->subbands; sb++) {
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levels[ch][sb] = ((1 << bits[ch][sb]) - 1) <<
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(32 - (frame->scale_factor[ch][sb] +
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SCALE_OUT_BITS + 2));
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sb_sample_delta[ch][sb] = (uint32_t) 1 <<
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(frame->scale_factor[ch][sb] +
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SCALE_OUT_BITS + 1);
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}
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}
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for (blk = 0; blk < frame->blocks; blk++) {
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for (ch = 0; ch < frame->channels; ch++) {
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for (sb = 0; sb < frame->subbands; sb++) {
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if (bits[ch][sb] == 0)
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continue;
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audio_sample = ((uint64_t) levels[ch][sb] *
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(sb_sample_delta[ch][sb] +
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frame->sb_sample_f[blk][ch][sb])) >> 32;
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put_bits(&pb, bits[ch][sb], audio_sample);
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}
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}
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}
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flush_put_bits(&pb);
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return (put_bits_count(&pb) + 7) / 8;
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}
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static int sbc_encode_init(AVCodecContext *avctx)
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{
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SBCEncContext *sbc = avctx->priv_data;
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struct sbc_frame *frame = &sbc->frame;
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if (avctx->profile == FF_PROFILE_SBC_MSBC)
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sbc->msbc = 1;
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if (sbc->msbc) {
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if (avctx->channels != 1) {
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av_log(avctx, AV_LOG_ERROR, "mSBC require mono channel.\n");
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return AVERROR(EINVAL);
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}
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if (avctx->sample_rate != 16000) {
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av_log(avctx, AV_LOG_ERROR, "mSBC require 16 kHz samplerate.\n");
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return AVERROR(EINVAL);
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}
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frame->mode = SBC_MODE_MONO;
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frame->subbands = 8;
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frame->blocks = MSBC_BLOCKS;
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frame->allocation = SBC_AM_LOUDNESS;
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frame->bitpool = 26;
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avctx->frame_size = 8 * MSBC_BLOCKS;
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} else {
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int d;
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if (avctx->global_quality > 255*FF_QP2LAMBDA) {
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av_log(avctx, AV_LOG_ERROR, "bitpool > 255 is not allowed.\n");
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return AVERROR(EINVAL);
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}
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if (avctx->channels == 1) {
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frame->mode = SBC_MODE_MONO;
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if (sbc->max_delay <= 3000 || avctx->bit_rate > 270000)
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frame->subbands = 4;
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else
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frame->subbands = 8;
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} else {
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if (avctx->bit_rate < 180000 || avctx->bit_rate > 420000)
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frame->mode = SBC_MODE_JOINT_STEREO;
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else
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frame->mode = SBC_MODE_STEREO;
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if (sbc->max_delay <= 4000 || avctx->bit_rate > 420000)
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frame->subbands = 4;
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else
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frame->subbands = 8;
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}
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/* sbc algorithmic delay is ((blocks + 10) * subbands - 2) / sample_rate */
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frame->blocks = av_clip(((sbc->max_delay * avctx->sample_rate + 2)
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/ (1000000 * frame->subbands)) - 10, 4, 16) & ~3;
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frame->allocation = SBC_AM_LOUDNESS;
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d = frame->blocks * ((frame->mode == SBC_MODE_DUAL_CHANNEL) + 1);
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frame->bitpool = (((avctx->bit_rate * frame->subbands * frame->blocks) / avctx->sample_rate)
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- 4 * frame->subbands * avctx->channels
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- (frame->mode == SBC_MODE_JOINT_STEREO)*frame->subbands - 32 + d/2) / d;
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if (avctx->global_quality > 0)
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frame->bitpool = avctx->global_quality / FF_QP2LAMBDA;
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avctx->frame_size = 4*((frame->subbands >> 3) + 1) * 4*(frame->blocks >> 2);
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}
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for (int i = 0; avctx->codec->supported_samplerates[i]; i++)
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if (avctx->sample_rate == avctx->codec->supported_samplerates[i])
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frame->frequency = i;
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frame->channels = avctx->channels;
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frame->codesize = frame->subbands * frame->blocks * avctx->channels * 2;
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frame->crc_ctx = av_crc_get_table(AV_CRC_8_EBU);
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memset(&sbc->dsp.X, 0, sizeof(sbc->dsp.X));
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sbc->dsp.position = (SBC_X_BUFFER_SIZE - frame->subbands * 9) & ~7;
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sbc->dsp.increment = sbc->msbc ? 1 : 4;
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ff_sbcdsp_init(&sbc->dsp);
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return 0;
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}
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static int sbc_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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const AVFrame *av_frame, int *got_packet_ptr)
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{
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SBCEncContext *sbc = avctx->priv_data;
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struct sbc_frame *frame = &sbc->frame;
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uint8_t joint = frame->mode == SBC_MODE_JOINT_STEREO;
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uint8_t dual = frame->mode == SBC_MODE_DUAL_CHANNEL;
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int ret, j = 0;
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int frame_length = 4 + (4 * frame->subbands * frame->channels) / 8
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+ ((frame->blocks * frame->bitpool * (1 + dual)
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+ joint * frame->subbands) + 7) / 8;
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/* input must be large enough to encode a complete frame */
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if (av_frame->nb_samples * frame->channels * 2 < frame->codesize)
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return 0;
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if ((ret = ff_alloc_packet2(avctx, avpkt, frame_length, 0)) < 0)
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return ret;
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/* Select the needed input data processing function and call it */
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if (frame->subbands == 8)
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sbc->dsp.position = sbc->dsp.sbc_enc_process_input_8s(
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sbc->dsp.position, av_frame->data[0], sbc->dsp.X,
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frame->subbands * frame->blocks, frame->channels);
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else
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sbc->dsp.position = sbc->dsp.sbc_enc_process_input_4s(
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sbc->dsp.position, av_frame->data[0], sbc->dsp.X,
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frame->subbands * frame->blocks, frame->channels);
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sbc_analyze_audio(&sbc->dsp, &sbc->frame);
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if (frame->mode == JOINT_STEREO)
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j = sbc->dsp.sbc_calc_scalefactors_j(frame->sb_sample_f,
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frame->scale_factor,
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frame->blocks,
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frame->subbands);
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else
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sbc->dsp.sbc_calc_scalefactors(frame->sb_sample_f,
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frame->scale_factor,
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frame->blocks,
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frame->channels,
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frame->subbands);
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emms_c();
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sbc_pack_frame(avpkt, frame, j, sbc->msbc);
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*got_packet_ptr = 1;
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return 0;
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}
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#define OFFSET(x) offsetof(SBCEncContext, x)
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#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
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static const AVOption options[] = {
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{ "sbc_delay", "set maximum algorithmic latency",
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OFFSET(max_delay), AV_OPT_TYPE_DURATION, {.i64 = 13000}, 1000,13000, AE },
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{ "msbc", "use mSBC mode (wideband speech mono SBC)",
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OFFSET(msbc), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AE },
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FF_AVCTX_PROFILE_OPTION("msbc", NULL, AUDIO, FF_PROFILE_SBC_MSBC)
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{ NULL },
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};
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static const AVClass sbc_class = {
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.class_name = "sbc encoder",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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AVCodec ff_sbc_encoder = {
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.name = "sbc",
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.long_name = NULL_IF_CONFIG_SMALL("SBC (low-complexity subband codec)"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_SBC,
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.priv_data_size = sizeof(SBCEncContext),
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.init = sbc_encode_init,
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.encode2 = sbc_encode_frame,
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.capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME,
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.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
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.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
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AV_CH_LAYOUT_STEREO, 0},
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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.supported_samplerates = (const int[]) { 16000, 32000, 44100, 48000, 0 },
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.priv_class = &sbc_class,
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.profiles = NULL_IF_CONFIG_SMALL(ff_sbc_profiles),
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};
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