forked from KolibriOS/kolibrios
447 lines
16 KiB
C
447 lines
16 KiB
C
/*
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* Opus encoder using libopus
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* Copyright (c) 2012 Nathan Caldwell
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <opus.h>
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#include <opus_multistream.h>
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "bytestream.h"
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#include "internal.h"
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#include "libopus.h"
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#include "vorbis.h"
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#include "audio_frame_queue.h"
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typedef struct LibopusEncOpts {
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int vbr;
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int application;
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int packet_loss;
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int complexity;
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float frame_duration;
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int packet_size;
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int max_bandwidth;
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} LibopusEncOpts;
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typedef struct LibopusEncContext {
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AVClass *class;
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OpusMSEncoder *enc;
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int stream_count;
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uint8_t *samples;
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LibopusEncOpts opts;
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AudioFrameQueue afq;
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} LibopusEncContext;
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static const uint8_t opus_coupled_streams[8] = {
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0, 1, 1, 2, 2, 2, 2, 3
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};
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/* Opus internal to Vorbis channel order mapping written in the header */
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static const uint8_t opus_vorbis_channel_map[8][8] = {
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{ 0 },
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{ 0, 1 },
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{ 0, 2, 1 },
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{ 0, 1, 2, 3 },
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{ 0, 4, 1, 2, 3 },
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{ 0, 4, 1, 2, 3, 5 },
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{ 0, 4, 1, 2, 3, 5, 6 },
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{ 0, 6, 1, 2, 3, 4, 5, 7 },
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};
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/* libavcodec to libopus channel order mapping, passed to libopus */
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static const uint8_t libavcodec_libopus_channel_map[8][8] = {
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{ 0 },
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{ 0, 1 },
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{ 0, 1, 2 },
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{ 0, 1, 2, 3 },
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{ 0, 1, 3, 4, 2 },
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{ 0, 1, 4, 5, 2, 3 },
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{ 0, 1, 5, 6, 2, 4, 3 },
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{ 0, 1, 6, 7, 4, 5, 2, 3 },
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};
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static void libopus_write_header(AVCodecContext *avctx, int stream_count,
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int coupled_stream_count,
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const uint8_t *channel_mapping)
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{
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uint8_t *p = avctx->extradata;
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int channels = avctx->channels;
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bytestream_put_buffer(&p, "OpusHead", 8);
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bytestream_put_byte(&p, 1); /* Version */
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bytestream_put_byte(&p, channels);
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bytestream_put_le16(&p, avctx->delay); /* Lookahead samples at 48kHz */
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bytestream_put_le32(&p, avctx->sample_rate); /* Original sample rate */
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bytestream_put_le16(&p, 0); /* Gain of 0dB is recommended. */
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/* Channel mapping */
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if (channels > 2) {
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bytestream_put_byte(&p, channels <= 8 ? 1 : 255);
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bytestream_put_byte(&p, stream_count);
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bytestream_put_byte(&p, coupled_stream_count);
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bytestream_put_buffer(&p, channel_mapping, channels);
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} else {
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bytestream_put_byte(&p, 0);
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}
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}
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static int libopus_configure_encoder(AVCodecContext *avctx, OpusMSEncoder *enc,
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LibopusEncOpts *opts)
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{
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int ret;
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if (avctx->global_quality) {
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av_log(avctx, AV_LOG_ERROR,
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"Quality-based encoding not supported, "
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"please specify a bitrate and VBR setting.\n");
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return AVERROR(EINVAL);
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}
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ret = opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(avctx->bit_rate));
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if (ret != OPUS_OK) {
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av_log(avctx, AV_LOG_ERROR,
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"Failed to set bitrate: %s\n", opus_strerror(ret));
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return ret;
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}
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ret = opus_multistream_encoder_ctl(enc,
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OPUS_SET_COMPLEXITY(opts->complexity));
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if (ret != OPUS_OK)
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av_log(avctx, AV_LOG_WARNING,
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"Unable to set complexity: %s\n", opus_strerror(ret));
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ret = opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(!!opts->vbr));
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if (ret != OPUS_OK)
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av_log(avctx, AV_LOG_WARNING,
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"Unable to set VBR: %s\n", opus_strerror(ret));
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ret = opus_multistream_encoder_ctl(enc,
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OPUS_SET_VBR_CONSTRAINT(opts->vbr == 2));
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if (ret != OPUS_OK)
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av_log(avctx, AV_LOG_WARNING,
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"Unable to set constrained VBR: %s\n", opus_strerror(ret));
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ret = opus_multistream_encoder_ctl(enc,
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OPUS_SET_PACKET_LOSS_PERC(opts->packet_loss));
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if (ret != OPUS_OK)
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av_log(avctx, AV_LOG_WARNING,
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"Unable to set expected packet loss percentage: %s\n",
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opus_strerror(ret));
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if (avctx->cutoff) {
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ret = opus_multistream_encoder_ctl(enc,
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OPUS_SET_MAX_BANDWIDTH(opts->max_bandwidth));
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if (ret != OPUS_OK)
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av_log(avctx, AV_LOG_WARNING,
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"Unable to set maximum bandwidth: %s\n", opus_strerror(ret));
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}
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return OPUS_OK;
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}
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static av_cold int libopus_encode_init(AVCodecContext *avctx)
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{
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LibopusEncContext *opus = avctx->priv_data;
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const uint8_t *channel_mapping;
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OpusMSEncoder *enc;
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int ret = OPUS_OK;
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int coupled_stream_count, header_size, frame_size;
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coupled_stream_count = opus_coupled_streams[avctx->channels - 1];
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opus->stream_count = avctx->channels - coupled_stream_count;
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channel_mapping = libavcodec_libopus_channel_map[avctx->channels - 1];
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/* FIXME: Opus can handle up to 255 channels. However, the mapping for
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* anything greater than 8 is undefined. */
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if (avctx->channels > 8)
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av_log(avctx, AV_LOG_WARNING,
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"Channel layout undefined for %d channels.\n", avctx->channels);
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if (!avctx->bit_rate) {
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/* Sane default copied from opusenc */
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avctx->bit_rate = 64000 * opus->stream_count +
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32000 * coupled_stream_count;
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av_log(avctx, AV_LOG_WARNING,
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"No bit rate set. Defaulting to %d bps.\n", avctx->bit_rate);
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}
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if (avctx->bit_rate < 500 || avctx->bit_rate > 256000 * avctx->channels) {
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av_log(avctx, AV_LOG_ERROR, "The bit rate %d bps is unsupported. "
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"Please choose a value between 500 and %d.\n", avctx->bit_rate,
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256000 * avctx->channels);
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return AVERROR(EINVAL);
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}
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frame_size = opus->opts.frame_duration * 48000 / 1000;
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switch (frame_size) {
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case 120:
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case 240:
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if (opus->opts.application != OPUS_APPLICATION_RESTRICTED_LOWDELAY)
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av_log(avctx, AV_LOG_WARNING,
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"LPC mode cannot be used with a frame duration of less "
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"than 10ms. Enabling restricted low-delay mode.\n"
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"Use a longer frame duration if this is not what you want.\n");
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/* Frame sizes less than 10 ms can only use MDCT mode, so switching to
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* RESTRICTED_LOWDELAY avoids an unnecessary extra 2.5ms lookahead. */
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opus->opts.application = OPUS_APPLICATION_RESTRICTED_LOWDELAY;
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case 480:
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case 960:
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case 1920:
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case 2880:
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opus->opts.packet_size =
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avctx->frame_size = frame_size * avctx->sample_rate / 48000;
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break;
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default:
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av_log(avctx, AV_LOG_ERROR, "Invalid frame duration: %g.\n"
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"Frame duration must be exactly one of: 2.5, 5, 10, 20, 40 or 60.\n",
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opus->opts.frame_duration);
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return AVERROR(EINVAL);
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}
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if (avctx->compression_level < 0 || avctx->compression_level > 10) {
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av_log(avctx, AV_LOG_WARNING,
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"Compression level must be in the range 0 to 10. "
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"Defaulting to 10.\n");
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opus->opts.complexity = 10;
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} else {
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opus->opts.complexity = avctx->compression_level;
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}
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if (avctx->cutoff) {
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switch (avctx->cutoff) {
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case 4000:
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opus->opts.max_bandwidth = OPUS_BANDWIDTH_NARROWBAND;
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break;
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case 6000:
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opus->opts.max_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
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break;
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case 8000:
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opus->opts.max_bandwidth = OPUS_BANDWIDTH_WIDEBAND;
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break;
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case 12000:
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opus->opts.max_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND;
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break;
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case 20000:
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opus->opts.max_bandwidth = OPUS_BANDWIDTH_FULLBAND;
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break;
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default:
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av_log(avctx, AV_LOG_WARNING,
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"Invalid frequency cutoff: %d. Using default maximum bandwidth.\n"
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"Cutoff frequency must be exactly one of: 4000, 6000, 8000, 12000 or 20000.\n",
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avctx->cutoff);
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avctx->cutoff = 0;
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}
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}
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enc = opus_multistream_encoder_create(avctx->sample_rate, avctx->channels,
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opus->stream_count,
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coupled_stream_count,
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channel_mapping,
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opus->opts.application, &ret);
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if (ret != OPUS_OK) {
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av_log(avctx, AV_LOG_ERROR,
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"Failed to create encoder: %s\n", opus_strerror(ret));
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return ff_opus_error_to_averror(ret);
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}
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ret = libopus_configure_encoder(avctx, enc, &opus->opts);
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if (ret != OPUS_OK) {
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ret = ff_opus_error_to_averror(ret);
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goto fail;
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}
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header_size = 19 + (avctx->channels > 2 ? 2 + avctx->channels : 0);
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avctx->extradata = av_malloc(header_size + FF_INPUT_BUFFER_PADDING_SIZE);
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if (!avctx->extradata) {
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av_log(avctx, AV_LOG_ERROR, "Failed to allocate extradata.\n");
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ret = AVERROR(ENOMEM);
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goto fail;
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}
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avctx->extradata_size = header_size;
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opus->samples = av_mallocz(frame_size * avctx->channels *
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av_get_bytes_per_sample(avctx->sample_fmt));
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if (!opus->samples) {
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av_log(avctx, AV_LOG_ERROR, "Failed to allocate samples buffer.\n");
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ret = AVERROR(ENOMEM);
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goto fail;
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}
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ret = opus_multistream_encoder_ctl(enc, OPUS_GET_LOOKAHEAD(&avctx->delay));
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if (ret != OPUS_OK)
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av_log(avctx, AV_LOG_WARNING,
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"Unable to get number of lookahead samples: %s\n",
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opus_strerror(ret));
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libopus_write_header(avctx, opus->stream_count, coupled_stream_count,
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opus_vorbis_channel_map[avctx->channels - 1]);
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ff_af_queue_init(avctx, &opus->afq);
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opus->enc = enc;
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return 0;
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fail:
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opus_multistream_encoder_destroy(enc);
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av_freep(&avctx->extradata);
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return ret;
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}
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static int libopus_encode(AVCodecContext *avctx, AVPacket *avpkt,
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const AVFrame *frame, int *got_packet_ptr)
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{
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LibopusEncContext *opus = avctx->priv_data;
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const int sample_size = avctx->channels *
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av_get_bytes_per_sample(avctx->sample_fmt);
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uint8_t *audio;
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int ret;
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int discard_padding;
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if (frame) {
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ff_af_queue_add(&opus->afq, frame);
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if (frame->nb_samples < opus->opts.packet_size) {
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audio = opus->samples;
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memcpy(audio, frame->data[0], frame->nb_samples * sample_size);
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} else
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audio = frame->data[0];
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} else {
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if (!opus->afq.remaining_samples)
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return 0;
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audio = opus->samples;
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memset(audio, 0, opus->opts.packet_size * sample_size);
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}
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/* Maximum packet size taken from opusenc in opus-tools. 60ms packets
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* consist of 3 frames in one packet. The maximum frame size is 1275
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* bytes along with the largest possible packet header of 7 bytes. */
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if ((ret = ff_alloc_packet2(avctx, avpkt, (1275 * 3 + 7) * opus->stream_count)) < 0)
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return ret;
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if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
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ret = opus_multistream_encode_float(opus->enc, (float *)audio,
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opus->opts.packet_size,
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avpkt->data, avpkt->size);
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else
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ret = opus_multistream_encode(opus->enc, (opus_int16 *)audio,
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opus->opts.packet_size,
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avpkt->data, avpkt->size);
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if (ret < 0) {
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av_log(avctx, AV_LOG_ERROR,
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"Error encoding frame: %s\n", opus_strerror(ret));
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return ff_opus_error_to_averror(ret);
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}
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av_shrink_packet(avpkt, ret);
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ff_af_queue_remove(&opus->afq, opus->opts.packet_size,
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&avpkt->pts, &avpkt->duration);
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discard_padding = opus->opts.packet_size - avpkt->duration;
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// Check if subtraction resulted in an overflow
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if ((discard_padding < opus->opts.packet_size) != (avpkt->duration > 0)) {
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av_free_packet(avpkt);
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av_free(avpkt);
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return AVERROR(EINVAL);
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}
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if (discard_padding > 0) {
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uint8_t* side_data = av_packet_new_side_data(avpkt,
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AV_PKT_DATA_SKIP_SAMPLES,
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10);
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if(side_data == NULL) {
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av_free_packet(avpkt);
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av_free(avpkt);
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return AVERROR(ENOMEM);
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}
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AV_WL32(side_data + 4, discard_padding);
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}
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*got_packet_ptr = 1;
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return 0;
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}
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static av_cold int libopus_encode_close(AVCodecContext *avctx)
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{
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LibopusEncContext *opus = avctx->priv_data;
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opus_multistream_encoder_destroy(opus->enc);
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ff_af_queue_close(&opus->afq);
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av_freep(&opus->samples);
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av_freep(&avctx->extradata);
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return 0;
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}
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#define OFFSET(x) offsetof(LibopusEncContext, opts.x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
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static const AVOption libopus_options[] = {
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{ "application", "Intended application type", OFFSET(application), AV_OPT_TYPE_INT, { .i64 = OPUS_APPLICATION_AUDIO }, OPUS_APPLICATION_VOIP, OPUS_APPLICATION_RESTRICTED_LOWDELAY, FLAGS, "application" },
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{ "voip", "Favor improved speech intelligibility", 0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_VOIP }, 0, 0, FLAGS, "application" },
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{ "audio", "Favor faithfulness to the input", 0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_AUDIO }, 0, 0, FLAGS, "application" },
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{ "lowdelay", "Restrict to only the lowest delay modes", 0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_RESTRICTED_LOWDELAY }, 0, 0, FLAGS, "application" },
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{ "frame_duration", "Duration of a frame in milliseconds", OFFSET(frame_duration), AV_OPT_TYPE_FLOAT, { .dbl = 10.0 }, 2.5, 60.0, FLAGS },
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{ "packet_loss", "Expected packet loss percentage", OFFSET(packet_loss), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 100, FLAGS },
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{ "vbr", "Variable bit rate mode", OFFSET(vbr), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 2, FLAGS, "vbr" },
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{ "off", "Use constant bit rate", 0, AV_OPT_TYPE_CONST, { .i64 = 0 }, 0, 0, FLAGS, "vbr" },
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{ "on", "Use variable bit rate", 0, AV_OPT_TYPE_CONST, { .i64 = 1 }, 0, 0, FLAGS, "vbr" },
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{ "constrained", "Use constrained VBR", 0, AV_OPT_TYPE_CONST, { .i64 = 2 }, 0, 0, FLAGS, "vbr" },
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{ NULL },
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};
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static const AVClass libopus_class = {
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.class_name = "libopus",
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.item_name = av_default_item_name,
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.option = libopus_options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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static const AVCodecDefault libopus_defaults[] = {
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{ "b", "0" },
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{ "compression_level", "10" },
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{ NULL },
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};
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static const int libopus_sample_rates[] = {
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48000, 24000, 16000, 12000, 8000, 0,
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};
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AVCodec ff_libopus_encoder = {
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.name = "libopus",
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.long_name = NULL_IF_CONFIG_SMALL("libopus Opus"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_OPUS,
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.priv_data_size = sizeof(LibopusEncContext),
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.init = libopus_encode_init,
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.encode2 = libopus_encode,
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.close = libopus_encode_close,
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.capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_NONE },
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.channel_layouts = ff_vorbis_channel_layouts,
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.supported_samplerates = libopus_sample_rates,
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.priv_class = &libopus_class,
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.defaults = libopus_defaults,
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};
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