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ffmpeg/libavcodec/libopusenc.c

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/*
* 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,
};
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