/*初始化*/
jlong Java_com_vvku_aacencoder_heaacEncInterface_init(JNIEnv* env, jobject thiz, jint samplerate, jint channels, jint bitrate, jint bandwidth, jlongArray param_out, jstring input_file)
{
__android_log_print(ANDROID_LOG_INFO, "encoderInterface native", "begin init");
jlong *info = (jlong*)(*env)->GetLongArrayElements(env, param_out, 0);
Encoder * en = (Encoder *) malloc(sizeof(Encoder));
en->inputInfo = (WavInfo *) malloc(sizeof(WavInfo));
en->bitrate = 16000;
en->sampleRateAAC = 44100;
unsigned char* wav_file = (char*)(*env)->GetStringUTFChars(env, input_file, 0);
FILE *inputfile;
inputfile = AuChannelOpen(wav_file, en->inputInfo);
if(bitrate > 0){
en->bitrate = bitrate * 1000;
}
if(!inputfile){
en->inputInfo->nChannels = 2;
en->inputInfo->sampleRate = 32000;
}
if(samplerate > 0){
en->inputInfo->sampleRate = samplerate;
}
if(channels > 0){
en->inputInfo->nChannels = channels;
}
en->hEncoder = aacplusEncOpen(en->inputInfo->sampleRate,
en->inputInfo->nChannels,
&en->inputSamples,
&en->maxOutputBytes);
info[0] = en->inputSamples*2;
info[1] = en->maxOutputBytes;
en->cfg = aacplusEncGetCurrentConfiguration(en->hEncoder);
en->cfg->bitRate = en->bitrate;
en->cfg->bandWidth = 0;
en->cfg->outputFormat = 0; // 设置为1的话，会加上adts头，直接保存成aac文件的时候需要
en->cfg->nChannelsOut = en->inputInfo->nChannels;
//en->cfg->inputFormat = AACPLUS_INPUT_FLOAT;
int ret = 0;
if((ret = aacplusEncSetConfiguration(en->hEncoder, en->cfg)) == 0) {
__android_log_print(ANDROID_LOG_INFO, "encoderInterface native", "Init failed.");
if(inputfile) AuChannelClose(inputfile);
(*env)->ReleaseLongArrayElements(env, param_out, info, 0);
(*env)->ReleaseStringUTFChars(env, input_file, wav_file);
return -2;
}
if(inputfile) AuChannelClose(inputfile);
(*env)->ReleaseLongArrayElements(env, param_out, info, 0);
(*env)->ReleaseStringUTFChars(env, input_file, wav_file);
__android_log_print(ANDROID_LOG_INFO, "encoderInterface native", "init success.");
return (jlong) en;
}
LRESULT CMainDlg::OnBtnTest(WORD wNotifyCode, WORD wID, HWND hWndCtl,
BOOL& bHandled) {
HANDLE_MP4_FILE hMp4File;
hAudioChannel inputFile = NULL;
AuChanInfo inputInfo;
AuChanMode auFlags = AU_CHAN_READ;
AuChanType auType = TYPE_AUTODETECT ; /* must be set */
inputInfo.bitsPerSample = 16 ; /* only relevant if valid == 1 */
inputInfo.sampleRate = 44100 ; /* only relevant if valid == 1 */
inputInfo.nChannels = 2 ; /* only relevant if valid == 1 */
inputInfo.nSamples = 0 ; /* only relevant if valid == 1 */
inputInfo.isLittleEndian = 1;
inputInfo.fpScaleFactor = AACENC_PCM_LEVEL ; /* must be set */
inputInfo.valid = 1 ; /* must be set */
inputInfo.useWaveExt = 0;
char file_path[MAX_PATH] = {0};
GetDlgItemTextA(m_hWnd, IDC_EDT_SOURCE, file_path, MAX_PATH);
int ret = AuChannelOpen (&inputFile, file_path, auFlags, &auType, &inputInfo);
struct aac_encoder_t *encoder = 0;
open_aac_encoder(encoder, inputInfo.sampleRate, inputInfo.sampleRate,
inputInfo.nChannels, inputInfo.nChannels);
//AACENC_CONFIG config;
//AacInitDefaultConfig(&config);
//unsigned char ASConfigBuffer[80];
//unsigned int nConfigBits;
//unsigned int nConfigBytes;
//
//memset (ASConfigBuffer, 0, 80);
//if (GetMPEG4ASConfig(22050,
// 1,
// ASConfigBuffer,
// &nConfigBits,
// 1,
// 1) ) {
// fprintf(stderr, "\nCould not initialize Audio Specific Config\n");
// exit(10);
//}
if (encoder)
close_aac_encoder(encoder);
if (inputFile)
AuChannelClose (inputFile);
return 0;
}
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;
}
int main(int argc, char* argv[]) {
HANDLE_MP4_FILE hMp4File = 0;
FILE *pcm_fp = 0;
hAudioChannel inputFile = 0;
AuChanInfo inputInfo;
AuChanType auType = TYPE_AUTODETECT ; /* must be set */
AuChanMode auFlags = AU_CHAN_READ;
short TimeDataPcm[AACENC_BLOCKSIZE * 2 * MAX_CHANNELS];
inputInfo.bitsPerSample = 16 ; /* only relevant if valid == 1 */
inputInfo.sampleRate = 44100 ; /* only relevant if valid == 1 */
inputInfo.nChannels = 2 ; /* only relevant if valid == 1 */
inputInfo.nSamples = 0 ; /* only relevant if valid == 1 */
inputInfo.isLittleEndian = 1;
inputInfo.fpScaleFactor = AACENC_PCM_LEVEL ; /* must be set */
inputInfo.valid = 1 ; /* must be set */
inputInfo.useWaveExt = 0;
pcm_fp = fopen("f:/temp/im/aac_pcm.pcm", "wb");
int ret = AuChannelOpen (&inputFile, "c:\\audio\\ave_maria_48.wav", auFlags,
&auType, &inputInfo);
IHEAAC_ENC* enc = CreateAACEnc();
assert(enc && enc->Init(inputInfo.sampleRate, inputInfo.nChannels, 32000));
IHEAAC_DEC* dec = CreateAACDec();
assert(dec && dec->Init(inputInfo.sampleRate, 26000));
int inSamples = AACENC_BLOCKSIZE * inputInfo.nChannels;
int numSamplesRead = 0;
inSamples = enc->GetSampleCount();
std::string xxx, out_pcm;
while (1) {
if (AuChannelReadShort(inputFile, TimeDataPcm, inSamples,
&numSamplesRead) != AU_CHAN_OK) {
fprintf(stderr, "failed to read source file!\n");
return 0;
}
enc->Enc(TimeDataPcm, xxx);
dec->Dec(xxx, out_pcm, true);
fwrite(out_pcm.c_str(), 1, out_pcm.length(), pcm_fp);
}
if (enc)
DestroyAACEnc(enc);
if (inputFile)
AuChannelClose(inputFile);
if (pcm_fp)
fclose(pcm_fp);
system("pause");
return 0;
}