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ffmpeg/libavcodec/sbcenc.c

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/*
* 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 },
{ 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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