summaryrefslogtreecommitdiff
path: root/libavcodec/sbcenc.c
diff options
context:
space:
mode:
authorAurelien Jacobs <aurel@gnuage.org>2017-12-17 19:59:30 +0100
committerAurelien Jacobs <aurel@gnuage.org>2018-03-07 22:26:53 +0100
commitff4600d95471a653073a961ec77f32e2f946684a (patch)
tree403810f4aeb160c06803c31a10c6e02c65b86fb8 /libavcodec/sbcenc.c
parent2e08de08159df2079f1db2a7d8fe66e2ad2238d5 (diff)
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.c361
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),
+};