JackAC3Encoder.cpp

/*
Copyright (C) 2006 Jesse Chappell <jesse@essej.net> (AC3Jack)
Copyright (C) 2012 Grame

This program is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.

This program 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 General Public License for more details.

You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.

*/

#include "JackAC3Encoder.h"
#include "JackError.h"
#include <unistd.h>
#include <string.h>
#include <stdio.h>

#define max(x,y) (((x)>(y)) ? (x) : (y))
#define min(x,y) (((x)<(y)) ? (x) : (y))

namespace Jack
{

#ifndef __ppc__

JackAC3Encoder::JackAC3Encoder(const JackAC3EncoderParams& params)
{
        aften_set_defaults(&fAftenContext);

        fAftenContext.channels = params.channels;
        fAftenContext.samplerate = params.sample_rate;
        fAftenContext.params.bitrate = params.bitrate;

        int acmod = A52_ACMOD_MONO;
        int lfe = params.lfe;

        switch (params.channels) {
                case 1: acmod = A52_ACMOD_MONO; break;
                case 2: acmod = A52_ACMOD_STEREO; break;
                case 3: acmod = A52_ACMOD_3_0; break;
                case 4: acmod = A52_ACMOD_2_2; break;
                case 5: acmod = A52_ACMOD_3_2; break;
                default:
                        break;
        }

        if (lfe) {
                fAftenContext.channels += 1;
        }

        fAftenContext.acmod = acmod;
        fAftenContext.lfe = lfe;
        fAftenContext.sample_format = A52_SAMPLE_FMT_FLT;
        fAftenContext.verbose = 1;

        fAftenContext.system.n_threads = 1;

        // create interleaved framebuffer for MAX_AC3_CHANNELS
        fSampleBuffer = new float[MAX_AC3_CHANNELS * A52_SAMPLES_PER_FRAME];

        // create AC3 buffer
        fAC3Buffer = new unsigned char[A52_MAX_CODED_FRAME_SIZE];
        memset(fAC3Buffer, 0, A52_MAX_CODED_FRAME_SIZE);

        fZeroBuffer = new unsigned char[SPDIF_FRAME_SIZE];
        memset(fZeroBuffer, 0, SPDIF_FRAME_SIZE);

        fRingBuffer = jack_ringbuffer_create(32768);
        
        fOutSizeByte = 0;
        fFramePos = 0;

        fSampleRate = 0;
        fByteRate = 0;
}

bool JackAC3Encoder::Init(jack_nframes_t sample_rate)
{
        fSampleRate = sample_rate;
        fByteRate = fSampleRate * sizeof(short) * 2;
        return (aften_encode_init(&fAftenContext) == 0);
}

JackAC3Encoder::~JackAC3Encoder()
{
        aften_encode_close(&fAftenContext);

        delete [] fSampleBuffer;
        delete [] fAC3Buffer;
        delete [] fZeroBuffer;

        if (fRingBuffer) {
                jack_ringbuffer_free(fRingBuffer);
        }
}

void JackAC3Encoder::Process(float** inputs_buffer, float** outputs_buffer, int nframes)
{
        // fill and process frame buffers as appropriate
        jack_nframes_t frames_left = A52_SAMPLES_PER_FRAME - fFramePos;
        jack_nframes_t offset = 0;

        while (offset < nframes)
        {
                if ((nframes - offset) >= frames_left) {
                        // copy only frames_left more data
                        jack_nframes_t pos = fFramePos * fAftenContext.channels;
                        for (jack_nframes_t spos = offset; spos < offset + frames_left; ++spos) {
                                for (size_t i = 0; i < fAftenContext.channels; ++i) {
                                        fSampleBuffer[pos + i] = inputs_buffer[i][spos];
                                }
                                pos += fAftenContext.channels;
                        }  

                        // use interleaved version
#ifdef HAVE_AFTEN_NEW_API
                        // note additional parameter 'nframes'
                        // added in commit e1cbb66628de8aa496a75092d8d694234c67aa95 git://aften.git.sourceforge.net/gitroot/aften/aften
                        int res = aften_encode_frame(&fAftenContext, fAC3Buffer + SPDIF_HEADER_SIZE, fSampleBuffer, nframes);
#else
                        // released version 0.0.8 hasn't the 'count' parameter
                        int res = aften_encode_frame(&fAftenContext, fAC3Buffer + SPDIF_HEADER_SIZE, fSampleBuffer);
#endif

                        if (res < 0) {
                                jack_error("aften_encode_frame error !!");
                                return;
                        }

                        fOutSizeByte = res;

                        FillSpdifHeader(fAC3Buffer, fOutSizeByte + SPDIF_HEADER_SIZE);

                        // push AC3 output to SPDIF ring buffer
                        float calc_ac3byterate = (fOutSizeByte * fSampleRate / (float) A52_SAMPLES_PER_FRAME);  
                        jack_nframes_t silencebytes = (jack_nframes_t) (fOutSizeByte * (fByteRate / calc_ac3byterate)) - fOutSizeByte - SPDIF_HEADER_SIZE;

                        jack_ringbuffer_write(fRingBuffer, (const char *)fAC3Buffer, fOutSizeByte + SPDIF_HEADER_SIZE);

                        // write the proper remainder of zero padding (inefficient, should be memsetting)
                        jack_ringbuffer_write(fRingBuffer, (const char *)fZeroBuffer, silencebytes);

                        offset += frames_left;
                        frames_left = A52_SAMPLES_PER_FRAME;
                        fFramePos = 0;
                } else {
                        // copy incoming data into frame buffers without processing
                        jack_nframes_t pos = fFramePos * fAftenContext.channels;
                        for (jack_nframes_t spos = offset; spos < nframes; ++spos) {
                                for (size_t i = 0; i < fAftenContext.channels; ++i) {
                                        fSampleBuffer[pos + i] = inputs_buffer[i][spos];
                                }
                                pos += fAftenContext.channels;
                        }  

                        fFramePos += (nframes - offset);
                        offset += (nframes-offset);
                }
        }

        Output2Driver(outputs_buffer, nframes);
}

void JackAC3Encoder::FillSpdifHeader(unsigned char* buf, int outsize)
{
        // todo, use outsize and not assume the fixed frame size?
        int ac3outsize = outsize - SPDIF_HEADER_SIZE;

        buf[0] = 0x72; buf[1] = 0xf8;   /* spdif syncword */
        buf[2] = 0x1f; buf[3] = 0x4e;   /* .............. */
        buf[4] = 0x01;                  /* AC3 data */
        buf[5] = buf[13] & 7;           /* bsmod, stream = 0 */
        buf[6] = (ac3outsize << 3) & 0xff;
        buf[7] = (ac3outsize >> 5) & 0xff;

#if !IS_BIGENDIAN
        swab(buf+SPDIF_HEADER_SIZE, buf + SPDIF_HEADER_SIZE, ac3outsize);
#endif
}

int JackAC3Encoder::Output2Driver(float** outputs, jack_nframes_t nframes)
{
        int wrotebytes = 0;
        jack_nframes_t nframes_left = nframes;

        if (jack_ringbuffer_read_space(fRingBuffer) == 0) {

                // just write silence
                memset(outputs[0], 0, nframes * sizeof(jack_default_audio_sample_t));
                memset(outputs[1], 0, nframes * sizeof(jack_default_audio_sample_t));   

        } else {

                jack_ringbuffer_data_t rb_data[2];

                jack_ringbuffer_get_read_vector(fRingBuffer, rb_data);

                while (nframes_left > 0 && rb_data[0].len > 4) {

                        jack_nframes_t towrite_frames = (rb_data[0].len) / (sizeof(short) * 2);
                        towrite_frames = min(towrite_frames, nframes_left);

                        // write and deinterleave into the two channels
#if 1
                        sample_move_dS_s16(outputs[0] + (nframes - nframes_left), (char *) rb_data[0].buf, towrite_frames, sizeof(short) * 2);
                        sample_move_dS_s16(outputs[1] + (nframes - nframes_left), (char *) rb_data[0].buf + sizeof(short), towrite_frames, sizeof(short) * 2);
#else
                        sample_move_dS_s16_24ph(outputs[0] + (nframes - nframes_left), (char *) rb_data[0].buf, towrite_frames, sizeof(short) * 2);
                        sample_move_dS_s16_24ph(outputs[1] + (nframes - nframes_left), (char *) rb_data[0].buf + sizeof(short), towrite_frames, sizeof(short) * 2);
#endif                  
                        wrotebytes = towrite_frames * sizeof(short) * 2;
                        nframes_left -= towrite_frames;

                        jack_ringbuffer_read_advance(fRingBuffer, wrotebytes);
                        jack_ringbuffer_get_read_vector(fRingBuffer, rb_data);
                }

                if (nframes_left > 0) {
                        // write silence
                        memset(outputs[0] + (nframes - nframes_left), 0, (nframes_left) * sizeof(jack_default_audio_sample_t));
                        memset(outputs[1] + (nframes - nframes_left), 0, (nframes_left) * sizeof(jack_default_audio_sample_t));         
                }
        }

        return wrotebytes;
}

void JackAC3Encoder::sample_move_dS_s16(jack_default_audio_sample_t* dst, char *src, jack_nframes_t nsamples, unsigned long src_skip) 
{
        /* ALERT: signed sign-extension portability !!! */
        while (nsamples--) {
                *dst = (*((short *) src)) / SAMPLE_MAX_16BIT;
                dst++;
                src += src_skip;
        }
}       

void JackAC3Encoder::sample_move_dS_s16_24ph(jack_default_audio_sample_t* dst, char *src, jack_nframes_t nsamples, unsigned long src_skip) 
{
        /* ALERT: signed sign-extension portability !!! */
        while (nsamples--) {
                *dst = (((int)(*((short *) src))) << 8) / SAMPLE_MAX_24BIT;
                dst++;
                src += src_skip;
        }
}

void JackAC3Encoder::GetChannelName(const char* name, const char* alias, char* portname, int channel)
{
        /*
         * 2 channels = L, R
         * 3 channels = L, C, R
         * 4 channels = L, R, LS, RS
         * 5 ch       = L, C, R,  LS, RS
         * 6 ch       = L, C, R,  LS, RS, LFE
         */

        const char* AC3_name = "";

        switch (channel) {

                case 0:
                        AC3_name = "AC3_1_Left";
                        break;

                case 1:
                        if (fAftenContext.channels == 2 || fAftenContext.channels == 4) {
                                AC3_name = "AC3_2_Right";
                        } else {
                                AC3_name = "AC3_2_Center";
                        }
                        break;

                case 2:
                        if (fAftenContext.channels == 4) {
                                AC3_name = "AC3_3_LeftSurround";
                        } else {
                                AC3_name = "AC3_3_Right";
                        }
                        break;

                case 3:
                        if (fAftenContext.channels == 4) {
                                AC3_name = "AC3_4_RightSurround";
                        } else {
                                AC3_name = "AC3_4_LeftSurround";
                        }
                        break;

                case 4:
                        if (fAftenContext.channels > 4) {
                                AC3_name = "AC3_5_RightSurround";
                        }
                        break;

                default:
                        break;
        }

        // Last channel
        if (fAftenContext.lfe && (channel == fAftenContext.channels - 1)) {
                sprintf(portname, "%s:%s:AC3_%d_LFE", name, alias, fAftenContext.channels);
        } else {
                sprintf(portname, "%s:%s:%s", name, alias, AC3_name);
        }
}

#endif

} // end of namespace