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FFmpeg/libavcodec/sbcenc.c
Andreas Rheinhardt a247ac640d avcodec: Constify AVCodecs
Given that the AVCodec.next pointer has now been removed, most of the
AVCodecs are not modified at all any more and can therefore be made
const (as this patch does); the only exceptions are the very few codecs
for external libraries that have a init_static_data callback.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
2021-04-27 10:43:15 -03:00

362 lines
13 KiB
C

/*
* 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_bytes_output(&pb);
}
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,
};
const 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),
};