path: root/libhb/platform/macosx/encca_aac.c

/* encca_aac.c

   Copyright (c) 2003-2018 HandBrake Team
   This file is part of the HandBrake source code
   Homepage: <http://handbrake.fr/>.
   It may be used under the terms of the GNU General Public License v2.
   For full terms see the file COPYING file or visit http://www.gnu.org/licenses/gpl-2.0.html
 */

#include "hb.h"
#include "audio_remap.h"
#include <AudioToolbox/AudioToolbox.h>
#include <CoreAudio/CoreAudio.h>

enum AAC_MODE { AAC_MODE_LC, AAC_MODE_HE };

int  encCoreAudioInitLC(hb_work_object_t*, hb_job_t*);
int  encCoreAudioInitHE(hb_work_object_t*, hb_job_t*);
int  encCoreAudioInit(hb_work_object_t*, hb_job_t*, enum AAC_MODE mode);
int  encCoreAudioWork(hb_work_object_t*, hb_buffer_t**, hb_buffer_t**);
void encCoreAudioClose(hb_work_object_t*);

hb_work_object_t hb_encca_aac =
{
    WORK_ENC_CA_AAC,
    "AAC encoder (Apple)",
    encCoreAudioInitLC,
    encCoreAudioWork,
    encCoreAudioClose
};

hb_work_object_t hb_encca_haac =
{
    WORK_ENC_CA_HAAC,
    "HE-AAC encoder (Apple)",
    encCoreAudioInitHE,
    encCoreAudioWork,
    encCoreAudioClose
};

struct hb_work_private_s
{
    uint8_t *buf;
    hb_job_t *job;
    hb_list_t *list;

    AudioConverterRef converter;
    unsigned long isamples, isamplesiz, omaxpacket, nchannels;
    int64_t first_pts;
    int64_t delay;
    uint64_t samples, ibytes;
    Float64 osamplerate;

    int input_done;

    hb_audio_remap_t *remap;
};

#define MP4ESDescrTag                   0x03
#define MP4DecConfigDescrTag            0x04
#define MP4DecSpecificDescrTag          0x05

// based off of mov_mp4_read_descr_len from mov.c in ffmpeg's libavformat
static int readDescrLen(UInt8 **buffer)
{
    int len = 0;
    int count = 4;
    while (count--)
    {
        int c = *(*buffer)++;
        len = (len << 7) | (c & 0x7f);
        if (!(c & 0x80))
            break;
    }
    return len;
}

// based off of mov_mp4_read_descr from mov.c in ffmpeg's libavformat
static int readDescr(UInt8 **buffer, int *tag)
{
    *tag = *(*buffer)++;
    return readDescrLen(buffer);
}

// based off of mov_read_esds from mov.c in ffmpeg's libavformat
static long ReadESDSDescExt(void* descExt, UInt8 **buffer, UInt32 *size, int versionFlags)
{
    UInt8 *esds = (UInt8*)descExt;
    int tag, len;
    *size = 0;

    if (versionFlags)
        esds += 4; // version + flags
    readDescr(&esds, &tag);
    esds += 2;     // ID
    if (tag == MP4ESDescrTag)
        esds++;    // priority

    readDescr(&esds, &tag);
    if (tag == MP4DecConfigDescrTag)
    {
        esds++;    // object type id
        esds++;    // stream type
        esds += 3; // buffer size db
        esds += 4; // max bitrate
        esds += 4; // average bitrate

        len = readDescr(&esds, &tag);
        if (tag == MP4DecSpecificDescrTag)
        {
            *buffer = calloc(1, len + 8);
            if (*buffer)
            {
                memcpy(*buffer, esds, len);
                *size = len;
            }
        }
    }

    return noErr;
}

/***********************************************************************
 * hb_work_encCoreAudio_init switches
 ***********************************************************************
 *
 **********************************************************************/
int encCoreAudioInitLC(hb_work_object_t *w, hb_job_t *job)
{
    return encCoreAudioInit(w, job, AAC_MODE_LC);
}

int encCoreAudioInitHE(hb_work_object_t *w, hb_job_t *job)
{
    return encCoreAudioInit(w, job, AAC_MODE_HE);
}

/***********************************************************************
 * hb_work_encCoreAudio_init
 ***********************************************************************
 *
 **********************************************************************/
int encCoreAudioInit(hb_work_object_t *w, hb_job_t *job, enum AAC_MODE mode)
{
    hb_work_private_t *pv = calloc(1, sizeof(hb_work_private_t));
    hb_audio_t *audio = w->audio;
    AudioStreamBasicDescription input, output;
    UInt32 tmp, tmpsiz = sizeof(tmp);
    OSStatus err;

    w->private_data = pv;
    pv->job = job;
    pv->first_pts = AV_NOPTS_VALUE;

    // pass the number of channels used into the private work data
    pv->nchannels =
        hb_mixdown_get_discrete_channel_count(audio->config.out.mixdown);

    bzero(&input, sizeof(AudioStreamBasicDescription));
    input.mSampleRate = (Float64)audio->config.out.samplerate;
    input.mFormatID = kAudioFormatLinearPCM;
    input.mFormatFlags = (kLinearPCMFormatFlagIsFloat|kAudioFormatFlagsNativeEndian);
    input.mBytesPerPacket = 4 * pv->nchannels;
    input.mFramesPerPacket = 1;
    input.mBytesPerFrame = input.mBytesPerPacket * input.mFramesPerPacket;
    input.mChannelsPerFrame = pv->nchannels;
    input.mBitsPerChannel = 32;

    bzero(&output, sizeof(AudioStreamBasicDescription));
    switch (mode)
    {
        case AAC_MODE_HE:
            output.mFormatID = kAudioFormatMPEG4AAC_HE;
            break;
        case AAC_MODE_LC:
        default:
            output.mFormatID = kAudioFormatMPEG4AAC;
            break;
    }
    output.mSampleRate = (Float64)audio->config.out.samplerate;
    output.mChannelsPerFrame = pv->nchannels;
    // let CoreAudio decide the rest

    // initialise encoder
    err = AudioConverterNew(&input, &output, &pv->converter);
    if (err != noErr)
    {
        // Retry without the samplerate
        bzero(&output, sizeof(AudioStreamBasicDescription));
        switch (mode)
        {
            case AAC_MODE_HE:
                output.mFormatID = kAudioFormatMPEG4AAC_HE;
                break;
            case AAC_MODE_LC:
            default:
                output.mFormatID = kAudioFormatMPEG4AAC;
                break;
        }
        output.mChannelsPerFrame = pv->nchannels;

        err = AudioConverterNew(&input, &output, &pv->converter);

        if (err != noErr)
        {
            hb_log("Error creating an AudioConverter err=%"PRId64" output.mBytesPerFrame=%"PRIu64"",
                   (int64_t)err, (uint64_t)output.mBytesPerFrame);
            *job->done_error = HB_ERROR_UNKNOWN;
            *job->die = 1;
            return -1;
        }
    }

    // set encoder quality to maximum
    tmp = kAudioConverterQuality_Max;
    AudioConverterSetProperty(pv->converter, kAudioConverterCodecQuality,
                              sizeof(tmp), &tmp);

    if (audio->config.out.bitrate > 0)
    {
        // set encoder bitrate control mode to constrained variable
        tmp = kAudioCodecBitRateControlMode_VariableConstrained;
        AudioConverterSetProperty(pv->converter,
                                  kAudioCodecPropertyBitRateControlMode,
                                  sizeof(tmp), &tmp);

        // get available bitrates
        AudioValueRange *bitrates;
        ssize_t bitrateCounts;
        AudioConverterGetPropertyInfo(pv->converter,
                                      kAudioConverterApplicableEncodeBitRates,
                                      &tmpsiz, NULL);
        bitrates = malloc(tmpsiz);
        AudioConverterGetProperty(pv->converter,
                                  kAudioConverterApplicableEncodeBitRates,
                                  &tmpsiz, bitrates);
        bitrateCounts = tmpsiz / sizeof(AudioValueRange);

        // set bitrate
        tmp = audio->config.out.bitrate * 1000;
        if (tmp < bitrates[0].mMinimum)
            tmp = bitrates[0].mMinimum;
        if (tmp > bitrates[bitrateCounts-1].mMinimum)
            tmp = bitrates[bitrateCounts-1].mMinimum;
        free(bitrates);
        if (tmp != audio->config.out.bitrate * 1000)
        {
            hb_log("encCoreAudioInit: sanitizing track %d audio bitrate %d to %"PRIu32"",
                   audio->config.out.track, audio->config.out.bitrate, tmp / 1000);
        }
        AudioConverterSetProperty(pv->converter,
                                  kAudioConverterEncodeBitRate,
                                  sizeof(tmp), &tmp);
    }
    else if (audio->config.out.quality >= 0)
    {
        if (mode != AAC_MODE_LC)
        {
            hb_error("encCoreAudioInit: internal error, VBR set but not applicable");
            return 1;
        }
        // set encoder bitrate control mode to variable
        tmp = kAudioCodecBitRateControlMode_Variable;
        AudioConverterSetProperty(pv->converter,
                                  kAudioCodecPropertyBitRateControlMode,
                                  sizeof(tmp), &tmp);

        // set quality
        tmp = audio->config.out.quality;
        AudioConverterSetProperty(pv->converter,
                                  kAudioCodecPropertySoundQualityForVBR,
                                  sizeof(tmp), &tmp);
    }
    else
    {
        hb_error("encCoreAudioInit: internal error, bitrate/quality not set");
        return 1;
    }

    // get real input
    tmpsiz = sizeof(input);
    AudioConverterGetProperty(pv->converter,
                              kAudioConverterCurrentInputStreamDescription,
                              &tmpsiz, &input);
    // get real output
    tmpsiz = sizeof(output);
    AudioConverterGetProperty(pv->converter,
                              kAudioConverterCurrentOutputStreamDescription,
                              &tmpsiz, &output);

    // set sizes
    pv->isamplesiz  = input.mBytesPerPacket;
    pv->isamples    = output.mFramesPerPacket;
    pv->osamplerate = output.mSampleRate;
    audio->config.out.samples_per_frame = pv->isamples;

    // channel remapping
    pv->remap = hb_audio_remap_init(AV_SAMPLE_FMT_FLT, &hb_aac_chan_map,
                                    audio->config.in.channel_map);
    if (pv->remap == NULL)
    {
        hb_error("encCoreAudioInit: hb_audio_remap_init() failed");
    }
    uint64_t layout = hb_ff_mixdown_xlat(audio->config.out.mixdown, NULL);
    hb_audio_remap_set_channel_layout(pv->remap, layout);

    // get maximum output size
    AudioConverterGetProperty(pv->converter,
                              kAudioConverterPropertyMaximumOutputPacketSize,
                              &tmpsiz, &tmp);
    pv->omaxpacket = tmp;

    // get magic cookie (elementary stream descriptor)
    tmp = HB_CONFIG_MAX_SIZE;
    AudioConverterGetProperty(pv->converter,
                              kAudioConverterCompressionMagicCookie,
                              &tmp, w->config->extradata.bytes);
    // CoreAudio returns a complete ESDS, but we only need
    // the DecoderSpecific info.
    UInt8* buffer = NULL;
    ReadESDSDescExt(w->config->extradata.bytes, &buffer, &tmpsiz, 0);
    w->config->extradata.length = tmpsiz;
    memmove(w->config->extradata.bytes, buffer, w->config->extradata.length);
    free(buffer);

    AudioConverterPrimeInfo primeInfo;
    UInt32 piSize = sizeof(primeInfo);
    bzero(&primeInfo, piSize);
    AudioConverterGetProperty(pv->converter,
                              kAudioConverterPrimeInfo,
                              &piSize, &primeInfo);

    pv->delay = primeInfo.leadingFrames * 90000LL / pv->osamplerate;
    w->config->init_delay = pv->delay;

    pv->list = hb_list_init();
    pv->buf = NULL;

    return 0;
}

/***********************************************************************
 * Close
 ***********************************************************************
 *
 **********************************************************************/
void encCoreAudioClose(hb_work_object_t *w)
{
    hb_work_private_t *pv = w->private_data;

    if (pv != NULL)
    {
        if (pv->converter)
        {
            AudioConverterDispose(pv->converter);
        }
        if (pv->buf != NULL)
        {
            free(pv->buf);
        }
        if (pv->remap != NULL)
        {
            hb_audio_remap_free(pv->remap);
        }
        hb_list_empty(&pv->list);
        free(pv);
        w->private_data = NULL;
    }
}

/* Called whenever necessary by AudioConverterFillComplexBuffer */
static OSStatus inInputDataProc(AudioConverterRef converter, UInt32 *npackets,
                                AudioBufferList *buffers,
                                AudioStreamPacketDescription **ignored,
                                void *userdata)
{
    hb_work_private_t *pv = userdata;

    if (!pv->ibytes)
    {
        *npackets = 0;
        if (pv->input_done)
        {
            // EOF on input
            buffers->mBuffers[0].mDataByteSize = 0;
            return noErr;
        }
        else
        {
            // Not enough data available
            return 1;
        }
    }

    if (pv->buf != NULL)
    {
        free(pv->buf);
    }

    buffers->mBuffers[0].mDataByteSize = MIN(pv->ibytes,
                                             pv->isamplesiz * *npackets);
    pv->buf = calloc(1, buffers->mBuffers[0].mDataByteSize);
    buffers->mBuffers[0].mData = pv->buf;

    if (hb_list_bytes(pv->list) >= buffers->mBuffers[0].mDataByteSize)
    {
        hb_list_getbytes(pv->list, buffers->mBuffers[0].mData,
                         buffers->mBuffers[0].mDataByteSize, NULL, NULL);
    }
    else
    {
        *npackets = 0;
        return 1;
    }

    *npackets = buffers->mBuffers[0].mDataByteSize / pv->isamplesiz;
    pv->ibytes -= buffers->mBuffers[0].mDataByteSize;

    hb_audio_remap(pv->remap, (uint8_t**)(&buffers->mBuffers[0].mData),
                   (int)(*npackets));

    return noErr;
}

/***********************************************************************
 * Encode
 ***********************************************************************
 *
 **********************************************************************/
static hb_buffer_t* Encode(hb_work_object_t *w)
{
    hb_work_private_t *pv = w->private_data;
    UInt32 npackets = 1;

    /* check if we need more data or we have already got to EOF
       if so, we need to call the audio converter again even
       without data to get out the remaining packets.
     */
    if (pv->input_done != 1 &&
        (pv->ibytes = hb_list_bytes(pv->list)) < pv->isamples * pv->isamplesiz)
    {
        return NULL;
    }

    hb_buffer_t *obuf;
    AudioStreamPacketDescription odesc = { 0 };
    AudioBufferList obuflist =
    {
        .mNumberBuffers = 1,
        .mBuffers = { { .mNumberChannels = pv->nchannels } },
    };

    obuf = hb_buffer_init(pv->omaxpacket);
    obuflist.mBuffers[0].mDataByteSize = obuf->size;
    obuflist.mBuffers[0].mData = obuf->data;

    OSStatus err = AudioConverterFillComplexBuffer(pv->converter,
                                                   inInputDataProc, pv,
                                                   &npackets, &obuflist, &odesc);

    if (err != noErr && err != 1)
    {
        hb_log("encCoreAudio: unexpected error in AudioConverterFillComplexBuffer()");
    }
    // only drop the output buffer if it's actually empty
    if (!npackets || odesc.mDataByteSize <= 0)
    {
        hb_buffer_close(&obuf);
        return NULL;
    }

    obuf->size        = odesc.mDataByteSize;
    obuf->s.start     = pv->first_pts - pv->delay +
                        90000LL * pv->samples / pv->osamplerate;
    pv->samples      += pv->isamples;
    obuf->s.stop      = pv->first_pts - pv->delay +
                        90000LL * pv->samples / pv->osamplerate;
    obuf->s.duration  = (double)90000 * pv->isamples / pv->osamplerate;
    obuf->s.type      = AUDIO_BUF;
    obuf->s.frametype = HB_FRAME_AUDIO;

    return obuf;
}

static void Flush(hb_work_object_t *w, hb_buffer_list_t * list)
{
    hb_buffer_t *buf = Encode(w);
    while (buf)
    {
        hb_buffer_list_append(list, buf);
        buf = Encode(w);
    }
}

/***********************************************************************
 * Work
 ***********************************************************************
 *
 **********************************************************************/
int encCoreAudioWork(hb_work_object_t *w, hb_buffer_t **buf_in,
                     hb_buffer_t **buf_out)
{
    hb_work_private_t *pv = w->private_data;
    hb_buffer_t      * in = *buf_in;
    hb_buffer_t      * buf;
    hb_buffer_list_t   list;

    hb_buffer_list_clear(&list);
    if (in->s.flags & HB_BUF_FLAG_EOF)
    {
        /* EOF on input - send it downstream & say we're done */
        pv->input_done = 1;
        Flush(w, &list);
        hb_buffer_list_append(&list, hb_buffer_eof_init());
        *buf_out = hb_buffer_list_clear(&list);
        return HB_WORK_DONE;
    }

    if (pv->first_pts == AV_NOPTS_VALUE)
    {
        pv->first_pts = in->s.start;
    }

    hb_list_add(pv->list, in);
    *buf_in = NULL;

    buf = Encode(w);
    while (buf)
    {
        hb_buffer_list_append(&list, buf);
        buf = Encode(w);
    }

    *buf_out = hb_buffer_list_clear(&list);
    return HB_WORK_OK;
}