|  | /*
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
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