libfdk-aacenc.c

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/*
 * AAC encoder wrapper
 * Copyright (c) 2012 Martin Storsjo
 *
 * 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 <fdk-aac/aacenc_lib.h>

#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "internal.h"

typedef struct AACContext {
    const AVClass *class;
    HANDLE_AACENCODER handle;
    int afterburner;
    int eld_sbr;
    int signaling;
    int latm;
    int header_period;
    int vbr;

    AudioFrameQueue afq;
} AACContext;

static const AVOption aac_enc_options[] = {
    { "afterburner", "Afterburner (improved quality)", offsetof(AACContext, afterburner), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
    { "eld_sbr", "Enable SBR for ELD (for SBR in other configurations, use the -profile parameter)", offsetof(AACContext, eld_sbr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
    { "signaling", "SBR/PS signaling style", offsetof(AACContext, signaling), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 2, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
    { "default", "Choose signaling implicitly (explicit hierarchical by default, implicit if global header is disabled)", 0, AV_OPT_TYPE_CONST, { .i64 = -1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
    { "implicit", "Implicit backwards compatible signaling", 0, AV_OPT_TYPE_CONST, { .i64 = 0 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
    { "explicit_sbr", "Explicit SBR, implicit PS signaling", 0, AV_OPT_TYPE_CONST, { .i64 = 1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
    { "explicit_hierarchical", "Explicit hierarchical signaling", 0, AV_OPT_TYPE_CONST, { .i64 = 2 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
    { "latm", "Output LATM/LOAS encapsulated data", offsetof(AACContext, latm), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
    { "header_period", "StreamMuxConfig and PCE repetition period (in frames)", offsetof(AACContext, header_period), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 0xffff, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
    { "vbr", "VBR mode (1-5)", offsetof(AACContext, vbr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 5, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
    { NULL }
};

static const AVClass aac_enc_class = {
    "libfdk_aac", av_default_item_name, aac_enc_options, LIBAVUTIL_VERSION_INT
};

static const char *aac_get_error(AACENC_ERROR err)
{
    switch (err) {
    case AACENC_OK:
        return "No error";
    case AACENC_INVALID_HANDLE:
        return "Invalid handle";
    case AACENC_MEMORY_ERROR:
        return "Memory allocation error";
    case AACENC_UNSUPPORTED_PARAMETER:
        return "Unsupported parameter";
    case AACENC_INVALID_CONFIG:
        return "Invalid config";
    case AACENC_INIT_ERROR:
        return "Initialization error";
    case AACENC_INIT_AAC_ERROR:
        return "AAC library initialization error";
    case AACENC_INIT_SBR_ERROR:
        return "SBR library initialization error";
    case AACENC_INIT_TP_ERROR:
        return "Transport library initialization error";
    case AACENC_INIT_META_ERROR:
        return "Metadata library initialization error";
    case AACENC_ENCODE_ERROR:
        return "Encoding error";
    case AACENC_ENCODE_EOF:
        return "End of file";
    default:
        return "Unknown error";
    }
}

static int aac_encode_close(AVCodecContext *avctx)
{
    AACContext *s = avctx->priv_data;

    if (s->handle)
        aacEncClose(&s->handle);
#if FF_API_OLD_ENCODE_AUDIO
    av_freep(&avctx->coded_frame);
#endif
    av_freep(&avctx->extradata);
    ff_af_queue_close(&s->afq);

    return 0;
}

static av_cold int aac_encode_init(AVCodecContext *avctx)
{
    AACContext *s = avctx->priv_data;
    int ret = AVERROR(EINVAL);
    AACENC_InfoStruct info = { 0 };
    CHANNEL_MODE mode;
    AACENC_ERROR err;
    int aot = FF_PROFILE_AAC_LOW + 1;
    int sce = 0, cpe = 0;

    if ((err = aacEncOpen(&s->handle, 0, avctx->channels)) != AACENC_OK) {
        av_log(avctx, AV_LOG_ERROR, "Unable to open the encoder: %s\n",
               aac_get_error(err));
        goto error;
    }

    if (avctx->profile != FF_PROFILE_UNKNOWN)
        aot = avctx->profile + 1;

    if ((err = aacEncoder_SetParam(s->handle, AACENC_AOT, aot)) != AACENC_OK) {
        av_log(avctx, AV_LOG_ERROR, "Unable to set the AOT %d: %s\n",
               aot, aac_get_error(err));
        goto error;
    }

    if (aot == FF_PROFILE_AAC_ELD + 1 && s->eld_sbr) {
        if ((err = aacEncoder_SetParam(s->handle, AACENC_SBR_MODE,
                                       1)) != AACENC_OK) {
            av_log(avctx, AV_LOG_ERROR, "Unable to enable SBR for ELD: %s\n",
                   aac_get_error(err));
            goto error;
        }
    }

    if ((err = aacEncoder_SetParam(s->handle, AACENC_SAMPLERATE,
                                   avctx->sample_rate)) != AACENC_OK) {
        av_log(avctx, AV_LOG_ERROR, "Unable to set the sample rate %d: %s\n",
               avctx->sample_rate, aac_get_error(err));
        goto error;
    }

    switch (avctx->channels) {
    case 1: mode = MODE_1;       sce = 1; cpe = 0; break;
    case 2: mode = MODE_2;       sce = 0; cpe = 1; break;
    case 3: mode = MODE_1_2;     sce = 1; cpe = 1; break;
    case 4: mode = MODE_1_2_1;   sce = 2; cpe = 1; break;
    case 5: mode = MODE_1_2_2;   sce = 1; cpe = 2; break;
    case 6: mode = MODE_1_2_2_1; sce = 2; cpe = 2; break;
    default:
        av_log(avctx, AV_LOG_ERROR,
               "Unsupported number of channels %d\n", avctx->channels);
        goto error;
    }

    if ((err = aacEncoder_SetParam(s->handle, AACENC_CHANNELMODE,
                                   mode)) != AACENC_OK) {
        av_log(avctx, AV_LOG_ERROR,
               "Unable to set channel mode %d: %s\n", mode, aac_get_error(err));
        goto error;
    }

    if ((err = aacEncoder_SetParam(s->handle, AACENC_CHANNELORDER,
                                   1)) != AACENC_OK) {
        av_log(avctx, AV_LOG_ERROR,
               "Unable to set wav channel order %d: %s\n",
               mode, aac_get_error(err));
        goto error;
    }

    if (avctx->flags & CODEC_FLAG_QSCALE || s->vbr) {
        int mode = s->vbr ? s->vbr : avctx->global_quality;
        if (mode <  1 || mode > 5) {
            av_log(avctx, AV_LOG_WARNING,
                   "VBR quality %d out of range, should be 1-5\n", mode);
            mode = av_clip(mode, 1, 5);
        }
        av_log(avctx, AV_LOG_WARNING,
               "Note, the VBR setting is unsupported and only works with "
               "some parameter combinations\n");
        if ((err = aacEncoder_SetParam(s->handle, AACENC_BITRATEMODE,
                                       mode)) != AACENC_OK) {
            av_log(avctx, AV_LOG_ERROR, "Unable to set the VBR bitrate mode %d: %s\n",
                   mode, aac_get_error(err));
            goto error;
        }
    } else {
        if (avctx->bit_rate <= 0) {
            if (avctx->profile == FF_PROFILE_AAC_HE_V2) {
                sce = 1;
                cpe = 0;
            }
            avctx->bit_rate = (96*sce + 128*cpe) * avctx->sample_rate / 44;
            if (avctx->profile == FF_PROFILE_AAC_HE ||
                avctx->profile == FF_PROFILE_AAC_HE_V2 ||
                s->eld_sbr)
                avctx->bit_rate /= 2;
        }
        if ((err = aacEncoder_SetParam(s->handle, AACENC_BITRATE,
                                       avctx->bit_rate)) != AACENC_OK) {
            av_log(avctx, AV_LOG_ERROR, "Unable to set the bitrate %d: %s\n",
                   avctx->bit_rate, aac_get_error(err));
            goto error;
        }
    }

    /* Choose bitstream format - if global header is requested, use
     * raw access units, otherwise use ADTS. */
    if ((err = aacEncoder_SetParam(s->handle, AACENC_TRANSMUX,
                                   avctx->flags & CODEC_FLAG_GLOBAL_HEADER ? 0 : s->latm ? 10 : 2)) != AACENC_OK) {
        av_log(avctx, AV_LOG_ERROR, "Unable to set the transmux format: %s\n",
               aac_get_error(err));
        goto error;
    }

    if (s->latm && s->header_period) {
        if ((err = aacEncoder_SetParam(s->handle, AACENC_HEADER_PERIOD,
                                       s->header_period)) != AACENC_OK) {
             av_log(avctx, AV_LOG_ERROR, "Unable to set header period: %s\n",
                    aac_get_error(err));
             goto error;
        }
    }

    /* If no signaling mode is chosen, use explicit hierarchical signaling
     * if using mp4 mode (raw access units, with global header) and
     * implicit signaling if using ADTS. */
    if (s->signaling < 0)
        s->signaling = avctx->flags & CODEC_FLAG_GLOBAL_HEADER ? 2 : 0;

    if ((err = aacEncoder_SetParam(s->handle, AACENC_SIGNALING_MODE,
                                   s->signaling)) != AACENC_OK) {
        av_log(avctx, AV_LOG_ERROR, "Unable to set signaling mode %d: %s\n",
               s->signaling, aac_get_error(err));
        goto error;
    }

    if ((err = aacEncoder_SetParam(s->handle, AACENC_AFTERBURNER,
                                   s->afterburner)) != AACENC_OK) {
        av_log(avctx, AV_LOG_ERROR, "Unable to set afterburner to %d: %s\n",
               s->afterburner, aac_get_error(err));
        goto error;
    }

    if (avctx->cutoff > 0) {
        if (avctx->cutoff < (avctx->sample_rate + 255) >> 8) {
            av_log(avctx, AV_LOG_ERROR, "cutoff valid range is %d-20000\n",
                   (avctx->sample_rate + 255) >> 8);
            goto error;
        }
        if ((err = aacEncoder_SetParam(s->handle, AACENC_BANDWIDTH,
                                       avctx->cutoff)) != AACENC_OK) {
            av_log(avctx, AV_LOG_ERROR, "Unable to set the encoder bandwidth to %d: %s\n",
                   avctx->cutoff, aac_get_error(err));
            goto error;
        }
    }

    if ((err = aacEncEncode(s->handle, NULL, NULL, NULL, NULL)) != AACENC_OK) {
        av_log(avctx, AV_LOG_ERROR, "Unable to initialize the encoder: %s\n",
               aac_get_error(err));
        return AVERROR(EINVAL);
    }

    if ((err = aacEncInfo(s->handle, &info)) != AACENC_OK) {
        av_log(avctx, AV_LOG_ERROR, "Unable to get encoder info: %s\n",
               aac_get_error(err));
        goto error;
    }

#if FF_API_OLD_ENCODE_AUDIO
    avctx->coded_frame = avcodec_alloc_frame();
    if (!avctx->coded_frame) {
        ret = AVERROR(ENOMEM);
        goto error;
    }
#endif
    avctx->frame_size = info.frameLength;
    avctx->delay      = info.encoderDelay;
    ff_af_queue_init(avctx, &s->afq);

    if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
        avctx->extradata_size = info.confSize;
        avctx->extradata      = av_mallocz(avctx->extradata_size +
                                           FF_INPUT_BUFFER_PADDING_SIZE);
        if (!avctx->extradata) {
            ret = AVERROR(ENOMEM);
            goto error;
        }

        memcpy(avctx->extradata, info.confBuf, info.confSize);
    }
    return 0;
error:
    aac_encode_close(avctx);
    return ret;
}

static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                            const AVFrame *frame, int *got_packet_ptr)
{
    AACContext    *s        = avctx->priv_data;
    AACENC_BufDesc in_buf   = { 0 }, out_buf = { 0 };
    AACENC_InArgs  in_args  = { 0 };
    AACENC_OutArgs out_args = { 0 };
    int in_buffer_identifier = IN_AUDIO_DATA;
    int in_buffer_size, in_buffer_element_size;
    int out_buffer_identifier = OUT_BITSTREAM_DATA;
    int out_buffer_size, out_buffer_element_size;
    void *in_ptr, *out_ptr;
    int ret;
    AACENC_ERROR err;

    /* handle end-of-stream small frame and flushing */
    if (!frame) {
        in_args.numInSamples = -1;
    } else {
        in_ptr                   = frame->data[0];
        in_buffer_size           = 2 * avctx->channels * frame->nb_samples;
        in_buffer_element_size   = 2;

        in_args.numInSamples     = avctx->channels * frame->nb_samples;
        in_buf.numBufs           = 1;
        in_buf.bufs              = &in_ptr;
        in_buf.bufferIdentifiers = &in_buffer_identifier;
        in_buf.bufSizes          = &in_buffer_size;
        in_buf.bufElSizes        = &in_buffer_element_size;

        /* add current frame to the queue */
        if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
            return ret;
    }

    /* The maximum packet size is 6144 bits aka 768 bytes per channel. */
    if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels))))
        return ret;

    out_ptr                   = avpkt->data;
    out_buffer_size           = avpkt->size;
    out_buffer_element_size   = 1;
    out_buf.numBufs           = 1;
    out_buf.bufs              = &out_ptr;
    out_buf.bufferIdentifiers = &out_buffer_identifier;
    out_buf.bufSizes          = &out_buffer_size;
    out_buf.bufElSizes        = &out_buffer_element_size;

    if ((err = aacEncEncode(s->handle, &in_buf, &out_buf, &in_args,
                            &out_args)) != AACENC_OK) {
        if (!frame && err == AACENC_ENCODE_EOF)
            return 0;
        av_log(avctx, AV_LOG_ERROR, "Unable to encode frame: %s\n",
               aac_get_error(err));
        return AVERROR(EINVAL);
    }

    if (!out_args.numOutBytes)
        return 0;

    /* Get the next frame pts & duration */
    ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
                       &avpkt->duration);

    avpkt->size     = out_args.numOutBytes;
    *got_packet_ptr = 1;
    return 0;
}

static const AVProfile profiles[] = {
    { FF_PROFILE_AAC_LOW,   "LC"       },
    { FF_PROFILE_AAC_HE,    "HE-AAC"   },
    { FF_PROFILE_AAC_HE_V2, "HE-AACv2" },
    { FF_PROFILE_AAC_LD,    "LD"       },
    { FF_PROFILE_AAC_ELD,   "ELD"      },
    { FF_PROFILE_UNKNOWN },
};

static const AVCodecDefault aac_encode_defaults[] = {
    { "b", "0" },
    { NULL }
};

static const uint64_t aac_channel_layout[] = {
    AV_CH_LAYOUT_MONO,
    AV_CH_LAYOUT_STEREO,
    AV_CH_LAYOUT_SURROUND,
    AV_CH_LAYOUT_4POINT0,
    AV_CH_LAYOUT_5POINT0_BACK,
    AV_CH_LAYOUT_5POINT1_BACK,
    0,
};

static const int aac_sample_rates[] = {
    96000, 88200, 64000, 48000, 44100, 32000,
    24000, 22050, 16000, 12000, 11025, 8000, 0
};

AVCodec ff_libfdk_aac_encoder = {
    .name                  = "libfdk_aac",
    .type                  = AVMEDIA_TYPE_AUDIO,
    .id                    = AV_CODEC_ID_AAC,
    .priv_data_size        = sizeof(AACContext),
    .init                  = aac_encode_init,
    .encode2               = aac_encode_frame,
    .close                 = aac_encode_close,
    .capabilities          = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
    .sample_fmts           = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
                                                            AV_SAMPLE_FMT_NONE },
    .long_name             = NULL_IF_CONFIG_SMALL("Fraunhofer FDK AAC"),
    .priv_class            = &aac_enc_class,
    .defaults              = aac_encode_defaults,
    .profiles              = profiles,
    .supported_samplerates = aac_sample_rates,
    .channel_layouts       = aac_channel_layout,
};