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author | Aurelien Jacobs <aurel@gnuage.org> | 2017-12-17 19:59:30 +0100 |
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committer | Aurelien Jacobs <aurel@gnuage.org> | 2018-03-07 22:26:53 +0100 |
commit | ff4600d95471a653073a961ec77f32e2f946684a (patch) | |
tree | 403810f4aeb160c06803c31a10c6e02c65b86fb8 /libavcodec/sbcenc.c | |
parent | 2e08de08159df2079f1db2a7d8fe66e2ad2238d5 (diff) | |
download | ffmpeg-ff4600d95471a653073a961ec77f32e2f946684a.tar.gz |
sbc: implement SBC encoder (low-complexity subband codec)
This was originally based on libsbc, and was fully integrated into ffmpeg.
Diffstat (limited to 'libavcodec/sbcenc.c')
-rw-r--r-- | libavcodec/sbcenc.c | 361 |
1 files changed, 361 insertions, 0 deletions
diff --git a/libavcodec/sbcenc.c b/libavcodec/sbcenc.c new file mode 100644 index 0000000000..a0064c0e8c --- /dev/null +++ b/libavcodec/sbcenc.c @@ -0,0 +1,361 @@ +/* + * 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 <stdbool.h> +#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, bool 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] = 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 }, + { 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), +}; 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