path: root/libhb/encavcodecaudio.c

/* $Id: encavcodeca.c,v 1.23 2005/10/13 23:47:06 titer Exp $

   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. */

#include "hb.h"
#include "hbffmpeg.h"
#include "downmix.h"
#include "libavcodec/audioconvert.h"

struct hb_work_private_s
{
    hb_job_t       * job;
    AVCodecContext * context;

    int              out_discrete_channels;
    int              samples_per_frame;
    int              layout;
    unsigned long    input_samples;
    unsigned long    output_bytes;
    hb_list_t      * list;
    uint8_t        * buf;
};

static int  encavcodecaInit( hb_work_object_t *, hb_job_t * );
static int  encavcodecaWork( hb_work_object_t *, hb_buffer_t **, hb_buffer_t ** );
static void encavcodecaClose( hb_work_object_t * );

hb_work_object_t hb_encavcodeca =
{
    WORK_ENCAVCODEC_AUDIO,
    "AVCodec Audio encoder (libavcodec)",
    encavcodecaInit,
    encavcodecaWork,
    encavcodecaClose
};

static int encavcodecaInit( hb_work_object_t * w, hb_job_t * job )
{
    AVCodec * codec;
    AVCodecContext * context;
    hb_audio_t * audio = w->audio;

    hb_work_private_t * pv = calloc( 1, sizeof( hb_work_private_t ) );
    w->private_data = pv;

    pv->job = job;

    pv->out_discrete_channels = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(audio->config.out.mixdown);

    codec = avcodec_find_encoder( w->codec_param );
    if( !codec )
    {
        hb_log( "encavcodecaInit: avcodec_find_encoder "
                "failed" );
        return 1;
    }
    context = avcodec_alloc_context3(codec);

    AVDictionary *av_opts = NULL;
    if ( w->codec_param == CODEC_ID_AAC )
    {
        av_dict_set( &av_opts, "stereo_mode", "ms_off", 0 );
    }
    if ( w->codec_param == CODEC_ID_AC3 )
    {
        if( audio->config.out.mixdown == HB_AMIXDOWN_DOLBY ||
            audio->config.out.mixdown == HB_AMIXDOWN_DOLBYPLII )
        {
            av_dict_set( &av_opts, "dsur_mode", "on", 0 );
        }
    }

    context->channel_layout = AV_CH_LAYOUT_STEREO;
    switch( audio->config.out.mixdown )
    {
        case HB_AMIXDOWN_MONO:
            context->channel_layout = AV_CH_LAYOUT_MONO;
            pv->layout = HB_INPUT_CH_LAYOUT_MONO;
            break;

        case HB_AMIXDOWN_STEREO:
        case HB_AMIXDOWN_DOLBY:
        case HB_AMIXDOWN_DOLBYPLII:
            context->channel_layout = AV_CH_LAYOUT_STEREO;
            pv->layout = HB_INPUT_CH_LAYOUT_STEREO;
            break;

        case HB_AMIXDOWN_6CH:
            context->channel_layout = AV_CH_LAYOUT_5POINT1;
            pv->layout = HB_INPUT_CH_LAYOUT_3F2R | HB_INPUT_CH_LAYOUT_HAS_LFE;
            break;

        default:
            hb_log("encavcodecaInit: bad mixdown" );
            break;
    }

    if( audio->config.out.bitrate > 0 )
        context->bit_rate = audio->config.out.bitrate * 1000;
    else if( audio->config.out.quality >= 0 )
    {
        context->global_quality = audio->config.out.quality * FF_QP2LAMBDA;
        context->flags |= CODEC_FLAG_QSCALE;
    }

    if( audio->config.out.compression_level >= 0 )
        context->compression_level = audio->config.out.compression_level;

    context->sample_rate = audio->config.out.samplerate;
    context->channels = pv->out_discrete_channels;
    // Try to set format to float.  Fallback to whatever is supported.
    hb_ff_set_sample_fmt( context, codec );

    if( hb_avcodec_open( context, codec, &av_opts, 0 ) )
    {
        hb_log( "encavcodecaInit: avcodec_open failed" );
        return 1;
    }
    // avcodec_open populates the opts dictionary with the
    // things it didn't recognize.
    AVDictionaryEntry *t = NULL;
    while( ( t = av_dict_get( av_opts, "", t, AV_DICT_IGNORE_SUFFIX ) ) )
    {
        hb_log( "encavcodecaInit: Unknown avcodec option %s", t->key );
    }
    av_dict_free( &av_opts );

    pv->context = context;

    audio->config.out.samples_per_frame = pv->samples_per_frame = context->frame_size;
    pv->input_samples = pv->samples_per_frame * pv->out_discrete_channels;

    // Set a reasonable maximum output size
    pv->output_bytes = context->frame_size * 
        av_get_bytes_per_sample(context->sample_fmt) * 
        context->channels;

    pv->buf = malloc( pv->input_samples * sizeof( float ) );

    pv->list = hb_list_init();

    if ( context->extradata )
    {
        memcpy( w->config->extradata.bytes, context->extradata, context->extradata_size );
        w->config->extradata.length = context->extradata_size;
    }

    return 0;
}

/***********************************************************************
 * Close
 ***********************************************************************
 *
 **********************************************************************/
// Some encoders (e.g. flac) require a final NULL encode in order to
// finalize things.
static void Finalize( hb_work_object_t * w )
{
    hb_work_private_t * pv = w->private_data;
    hb_buffer_t * buf = hb_buffer_init( pv->output_bytes );

    // Finalize with NULL input needed by FLAC to generate md5sum
    // in context extradata
    avcodec_encode_audio( pv->context, buf->data, buf->alloc, NULL );
    hb_buffer_close( &buf );
}

static void encavcodecaClose( hb_work_object_t * w )
{
    hb_work_private_t * pv = w->private_data;

    if ( pv )
    {
        if( pv->context )
        {
            Finalize( w );
            hb_deep_log( 2, "encavcodeca: closing libavcodec" );
            if ( pv->context->codec )
                avcodec_flush_buffers( pv->context );
            hb_avcodec_close( pv->context );
        }

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

        if ( pv->list )
            hb_list_empty( &pv->list );

        free( pv );
        w->private_data = NULL;
    }
}

static hb_buffer_t * Encode( hb_work_object_t * w )
{
    hb_work_private_t * pv = w->private_data;
    uint64_t pts, pos;
    hb_audio_t * audio = w->audio;
    hb_buffer_t * buf;

    if( hb_list_bytes( pv->list ) < pv->input_samples * sizeof( float ) )
    {
        return NULL;
    }

    hb_list_getbytes( pv->list, pv->buf, pv->input_samples * sizeof( float ),
                      &pts, &pos);

    // XXX: ffaac fails to remap from the internal libav* channel map (SMPTE) to the native AAC channel map
    //      do it here - this hack should be removed if Libav fixes the bug
    hb_chan_map_t * out_map = ( w->codec_param == CODEC_ID_AAC ) ? &hb_qt_chan_map : &hb_smpte_chan_map;

    if ( audio->config.in.channel_map != out_map )
    {
        hb_layout_remap( audio->config.in.channel_map, out_map, pv->layout,
                         (float*)pv->buf, pv->samples_per_frame );
    }

    // Do we need to convert our internal float format?
    if ( pv->context->sample_fmt != AV_SAMPLE_FMT_FLT )
    {
        int isamp, osamp;
        AVAudioConvert *ctx;

        isamp = av_get_bytes_per_sample( AV_SAMPLE_FMT_FLT );
        osamp = av_get_bytes_per_sample( pv->context->sample_fmt );
        ctx = av_audio_convert_alloc( pv->context->sample_fmt, 1,
                                      AV_SAMPLE_FMT_FLT, 1,
                                      NULL, 0 );

        // get output buffer size then malloc a buffer
        //nsamples = out_size / isamp;
        //buffer = av_malloc( nsamples * sizeof(hb_sample_t) );

        // we're doing straight sample format conversion which 
        // behaves as if there were only one channel.
        const void * const ibuf[6] = { pv->buf };
        void * const obuf[6] = { pv->buf };
        const int istride[6] = { isamp };
        const int ostride[6] = { osamp };

        av_audio_convert( ctx, obuf, ostride, ibuf, istride, pv->input_samples );
        av_audio_convert_free( ctx );
    }
    
    buf = hb_buffer_init( pv->output_bytes );
    buf->size = avcodec_encode_audio( pv->context, buf->data, buf->alloc,
                                      (short*)pv->buf );

    buf->start = pts + 90000 * pos / pv->out_discrete_channels / sizeof( float ) / audio->config.out.samplerate;
    buf->stop  = buf->start + 90000 * pv->samples_per_frame / audio->config.out.samplerate;

    buf->frametype = HB_FRAME_AUDIO;

    if ( !buf->size )
    {
        hb_buffer_close( &buf );
        return Encode( w );
    }
    else if (buf->size < 0)
    {
        hb_log( "encavcodeca: avcodec_encode_audio failed" );
        hb_buffer_close( &buf );
        return NULL;
    }

    return buf;
}

static hb_buffer_t * Flush( hb_work_object_t * w )
{
    hb_work_private_t * pv = w->private_data;
    hb_buffer_t *first, *buf, *last;

    first = last = buf = Encode( w );
    while( buf )
    {
        last = buf;
        buf->next = Encode( w );
        buf = buf->next;
    }

    if( last )
    {
        last->next = hb_buffer_init( pv->output_bytes );
        buf = last->next;
    }
    else
    {
        first = buf = hb_buffer_init( pv->output_bytes );
    }
    // Finalize with NULL input needed by FLAC to generate md5sum
    // in context extradata
    avcodec_encode_audio( pv->context, buf->data, buf->alloc, NULL );
    buf->size = 0;
    return first;
}

/***********************************************************************
 * Work
 ***********************************************************************
 *
 **********************************************************************/
static int encavcodecaWork( 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, * buf;

    if ( in->size <= 0 )
    {
        /* EOF on input - send it downstream & say we're done */
        *buf_out = Flush( w );
        return HB_WORK_DONE;
    }

    if ( pv->context == NULL || pv->context->codec == NULL )
    {
        // No encoder context. Nothing we can do.
        return HB_WORK_OK;
    }

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

    *buf_out = buf = Encode( w );

    while ( buf )
    {
        buf->next = Encode( w );
        buf = buf->next;
    }

    return HB_WORK_OK;
}