static int aacPlus_encode_frame(AVCodecContext *avctx,
unsigned char *frame, int buf_size, void *data)
{
aacPlusAudioContext *s = avctx->priv_data;
int bytes_written;
bytes_written = aacplusEncEncode(s->aacplus_handle,
data,
avctx->frame_size * avctx->channels,
frame,
buf_size);
return bytes_written;
}
/*编码一帧数据*/
jint Java_com_vvku_aacencoder_heaacEncInterface_encodeframe(JNIEnv* env, jobject thiz, jlong handle, jbyteArray in, jint samplesRead, jbyteArray out)
{
Encoder * en = (Encoder *) handle;
jbyte * pcmbuf = (jbyte*)(*env)->GetByteArrayElements(env, in, 0);
jbyte * outputBuffer= (jbyte*)(*env)->GetByteArrayElements(env, out, 0);
short *in_buf = (short*) pcmbuf;
char out_buf[en->maxOutputBytes];
memset(out_buf, 0, sizeof(out_buf));
int numOutBytes=0;
numOutBytes = aacplusEncEncode(en->hEncoder, (int32_t *)in_buf, en->inputSamples, out_buf, en->maxOutputBytes);
memcpy(outputBuffer, out_buf, numOutBytes);
(*env)->ReleaseByteArrayElements(env,out,outputBuffer,0); //释放内存，防止泄露
(*env)->ReleaseByteArrayElements(env,in,pcmbuf,0);
return numOutBytes;
}
/*------------------------------------------------------------------------------
* Write data to the encoder
*----------------------------------------------------------------------------*/
unsigned int
aacPlusEncoder :: write ( const void * buf,
unsigned int len ) throw ( Exception )
{
if ( !isOpen() || len == 0) {
return 0;
}
unsigned int channels = getInChannel();
unsigned int bitsPerSample = getInBitsPerSample();
unsigned int sampleSize = (bitsPerSample / 8) * channels;
unsigned char * b = (unsigned char*) buf;
unsigned int processed = len - (len % sampleSize);
unsigned int nSamples = processed / sampleSize;
unsigned char * aacplusBuf = new unsigned char[maxOutputBytes];
int samples = (int) nSamples * channels;
int processedSamples = 0;
if ( converter ) {
unsigned int converted;
#ifdef HAVE_SRC_LIB
src_short_to_float_array ((short *) b, converterData.data_in, samples);
converterData.input_frames = nSamples;
converterData.data_out = resampledOffset + (resampledOffsetSize * channels);
int srcError = src_process (converter, &converterData);
if (srcError)
throw Exception (__FILE__, __LINE__, "libsamplerate error: ", src_strerror (srcError));
converted = converterData.output_frames_gen;
#else
int inCount = nSamples;
short int * shortBuffer = new short int[samples];
int outCount = (int) (inCount * resampleRatio);
Util::conv( bitsPerSample, b, processed, shortBuffer, isInBigEndian());
converted = converter->resample( inCount,
outCount+1,
shortBuffer,
&resampledOffset[resampledOffsetSize*channels]);
delete[] shortBuffer;
#endif
resampledOffsetSize += converted;
// encode samples (if enough)
while(resampledOffsetSize - processedSamples >= inputSamples/channels) {
int outputBytes;
#ifdef HAVE_SRC_LIB
short *shortData = new short[inputSamples];
src_float_to_short_array(resampledOffset + (processedSamples * channels),
shortData, inputSamples) ;
outputBytes = aacplusEncEncode(encoderHandle,
(int32_t*) shortData,
inputSamples,
aacplusBuf,
maxOutputBytes);
delete [] shortData;
#else
outputBytes = aacplusEncEncode(encoderHandle,
(int32_t*) &resampledOffset[processedSamples*channels],
inputSamples,
aacplusBuf,
maxOutputBytes);
#endif
getSink()->write(aacplusBuf, outputBytes);
processedSamples+=inputSamples/channels;
}
if (processedSamples && (int) resampledOffsetSize >= processedSamples) {
resampledOffsetSize -= processedSamples;
//move least part of resampled data to beginning
if(resampledOffsetSize)
#ifdef HAVE_SRC_LIB
resampledOffset = (float *) memmove(resampledOffset, &resampledOffset[processedSamples*channels],
resampledOffsetSize*channels*sizeof(float));
#else
resampledOffset = (short *) memmove(resampledOffset, &resampledOffset[processedSamples*channels],
resampledOffsetSize*sampleSize);
#endif
}
} else {
while (processedSamples < samples) {
int outputBytes;
int inSamples = samples - processedSamples < (int) inputSamples
? samples - processedSamples
: inputSamples;
outputBytes = aacplusEncEncode(encoderHandle,
(int32_t*) (b + processedSamples/sampleSize),
inSamples,
aacplusBuf,
maxOutputBytes);
getSink()->write(aacplusBuf, outputBytes);
processedSamples += inSamples;
}
}
delete[] aacplusBuf;
// return processedSamples;
return samples * sampleSize;
}
int main(int argc, char *argv[])
{
WavInfo inputInfo;
FILE *inputFile = NULL;
FILE *hADTSFile;
int error;
int bEncodeMono = 0;
int frmCnt = 0;
/*
* parse command line arguments
*/
if (argc != 5) {
fprintf(stderr, "\nUsage: %s <wav_file> <bitstream_file> <bitrate> <(m)ono/(s)tereo>\n", argv[0]);
fprintf(stderr, "\nExample: %s input.wav out.aac 24000 s\n", argv[0]);
return 0;
}
if ( strcmp (argv[4],"m") == 0 ) {
bEncodeMono = 1;
}
else {
if ( strcmp (argv[4],"s") != 0 ) {
fprintf(stderr, "\nWrong mode %s, use either (m)ono or (s)tereo\n", argv[4]);
return 0;
}
}
fflush(stdout);
inputFile = AuChannelOpen (argv[1], &inputInfo);
if(inputFile == NULL){
fprintf(stderr,"could not open %s\n",argv[1]);
exit(10);
}
if (inputInfo.nChannels==1 && !bEncodeMono) {
fprintf(stderr,"Need stereo input for stereo coding mode !\n");
exit(10);
}
if (strcmp(argv[2],"-")==0)
hADTSFile=stdout;
else
hADTSFile = fopen(argv[2], "wb");
if(!hADTSFile) {
fprintf(stderr, "\nFailed to create ADTS file\n") ;
exit(10);
}
/*
Be verbose
*/
unsigned long inputSamples=0;
unsigned long maxOutputBytes=0;
aacplusEncHandle hEncoder = aacplusEncOpen(inputInfo.sampleRate,
inputInfo.nChannels,
&inputSamples,
&maxOutputBytes);
aacplusEncConfiguration *cfg = aacplusEncGetCurrentConfiguration(hEncoder);
cfg->bitRate = atoi(argv[3]);
cfg->bandWidth = 0;
cfg->outputFormat = 1;
cfg->nChannelsOut = bEncodeMono ? 1 : inputInfo.nChannels;
if(inputInfo.aFmt == WAV_FORMAT_FLOAT){
cfg->inputFormat = AACPLUS_INPUT_FLOAT;
}
fprintf(stdout,"input file %s: \nsr = %d, nc = %d fmt = %d\n\n",
argv[1], inputInfo.sampleRate, inputInfo.nChannels, inputInfo.aFmt);
fprintf(stdout,"output file %s: \nbr = %d inputSamples = %lu maxOutputBytes = %lu nc = %d m = %d\n\n",
argv[2], cfg->bitRate, inputSamples, maxOutputBytes, cfg->nChannelsOut, bEncodeMono);
fflush(stdout);
int ret = 0;
if((ret = aacplusEncSetConfiguration(hEncoder, cfg)) == 0) {
fprintf(stdout,"setting cfg failed\n", ret);
return -1;
}
uint8_t *outputBuffer = malloc(maxOutputBytes);
int32_t *TimeDataPcm;
if(inputInfo.aFmt == WAV_FORMAT_FLOAT) {
TimeDataPcm = calloc(inputSamples, sizeof(float));
} else {
TimeDataPcm = calloc(inputSamples, sizeof(short));
}
int stopLoop = 0;
int bytes = 0;
do {
int numSamplesRead = 0;
if(inputInfo.aFmt == WAV_FORMAT_FLOAT) {
if ( AuChannelReadFloat(inputFile, (float *) TimeDataPcm, inputSamples, &numSamplesRead) > 0) {
stopLoop = 1;
break;
}
} else {
if ( AuChannelReadShort(inputFile, (short *) TimeDataPcm, inputSamples, &numSamplesRead) > 0) {
stopLoop = 1;
break;
}
}
if(numSamplesRead < inputSamples) {
stopLoop = 1;
break;
}
bytes = aacplusEncEncode(hEncoder, (int32_t *) TimeDataPcm, numSamplesRead,
outputBuffer,
maxOutputBytes);
if(bytes > 0) fwrite(outputBuffer, bytes, 1, hADTSFile);
frmCnt++;
fprintf(stderr,"[%d]\r",frmCnt); fflush(stderr);
} while (!stopLoop && bytes >= 0);
fprintf(stderr,"\n");
fflush(stderr);
printf("\nencoding finished\n");
aacplusEncClose(hEncoder);
fclose(hADTSFile);
free(outputBuffer);
free(TimeDataPcm);
return 0;
}
