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ffmpeg/libavcodec/libopusenc.c
Anton Khirnov 2df0c32ea1 lavc: use a separate field for exporting audio encoder padding 
Currently, the amount of padding inserted at the beginning by some audio
encoders, is exported through AVCodecContext.delay. However
- the term 'delay' is heavily overloaded and can have multiple different
  meanings even in the case of audio encoding.
- this field has entirely different meanings, depending on whether the
  codec context is used for encoding or decoding (and has yet another
  different meaning for video), preventing generic handling of the codec
  context.

Therefore, add a new field -- AVCodecContext.initial_padding. It could
conceivably be used for decoding as well at a later point.
2014-10-13 19:09:01 +00:00

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/*
* Opus encoder using libopus
* Copyright (c) 2012 Nathan Caldwell
*
* This file is part of libav.
*
* libav 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.
*
* libav 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 libav; 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 },
};
/* libav to libopus channel order mapping, passed to libopus */
static const uint8_t libav_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->initial_padding); /* 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;
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 int av_cold 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 = libav_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->initial_padding));
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;
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_packet(avpkt, (1275 * 3 + 7) * opus->stream_count)) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
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);
*got_packet_ptr = 1;
return 0;
}
static int av_cold 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 = 20.0 }, 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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