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