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