/*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;
}
/*------------------------------------------------------------------------------
* Open an encoding session
*----------------------------------------------------------------------------*/
bool
aacPlusEncoder :: open ( void )
throw ( Exception )
{
if ( isOpen() ) {
close();
}
// open the underlying sink
if ( !sink->open() ) {
throw Exception( __FILE__, __LINE__,
"aacplus lib opening underlying sink error");
}
reportEvent(1, "Using aacplus codec");
encoderHandle = aacplusEncOpen(getOutSampleRate(),
getInChannel(),
&inputSamples,
&maxOutputBytes);
aacplusEncConfiguration * aacplusConfig;
aacplusConfig = aacplusEncGetCurrentConfiguration(encoderHandle);
aacplusConfig->bitRate = getOutBitrate() * 1000;
aacplusConfig->bandWidth = lowpass;
aacplusConfig->outputFormat = 1;
aacplusConfig->inputFormat = AACPLUS_INPUT_16BIT;
aacplusConfig->nChannelsOut = getOutChannel();
if (!aacplusEncSetConfiguration(encoderHandle, aacplusConfig)) {
throw Exception(__FILE__, __LINE__,
"error configuring libaacplus library");
}
// initialize the resampling coverter if needed
if ( converter ) {
#ifdef HAVE_SRC_LIB
converterData.input_frames = 4096/((getInBitsPerSample() / 8) * getInChannel());
converterData.data_in = new float[converterData.input_frames*getInChannel()];
converterData.output_frames = (int) (converterData.input_frames * resampleRatio + 1);
if ((int) inputSamples > getInChannel() * converterData.output_frames) {
resampledOffset = new float[2 * inputSamples];
} else {
resampledOffset = new float[2 * getInChannel() * converterData.input_frames];
}
converterData.src_ratio = resampleRatio;
converterData.end_of_input = 0;
#else
converter->initialize( resampleRatio, getInChannel());
//needed 2x(converted input samples) to handle offsets
int outCount = 2 * getInChannel() * (inputSamples + 1);
if (resampleRatio > 1)
outCount = (int) (outCount * resampleRatio);
resampledOffset = new short int[outCount];
#endif
resampledOffsetSize = 0;
}
aacplusOpen = true;
reportEvent(10, "nChannelsAAC", aacplusConfig->nChannelsOut);
reportEvent(10, "sampleRateAAC", aacplusConfig->sampleRate);
reportEvent(10, "inSamples", inputSamples);
return true;
}
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;
}
static av_cold int aacPlus_encode_init(AVCodecContext *avctx)
{
aacPlusAudioContext *s = avctx->priv_data;
aacplusEncConfiguration *aacplus_cfg;
unsigned long samples_input, max_bytes_output;
/* number of channels */
if (avctx->channels < 1 || avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed\n", avctx->channels);
return -1;
}
s->aacplus_handle = aacplusEncOpen(avctx->sample_rate,
avctx->channels,
&samples_input, &max_bytes_output);
if(!s->aacplus_handle) {
av_log(avctx, AV_LOG_ERROR, "can't open encoder\n");
return -1;
}
/* check aacplus version */
aacplus_cfg = aacplusEncGetCurrentConfiguration(s->aacplus_handle);
/* put the options in the configuration struct */
if(avctx->profile != FF_PROFILE_AAC_LOW && avctx->profile != FF_PROFILE_UNKNOWN) {
av_log(avctx, AV_LOG_ERROR, "invalid AAC profile: %d, only LC supported\n", avctx->profile);
aacplusEncClose(s->aacplus_handle);
return -1;
}
aacplus_cfg->bitRate = avctx->bit_rate;
aacplus_cfg->bandWidth = avctx->cutoff;
aacplus_cfg->outputFormat = !(avctx->flags & CODEC_FLAG_GLOBAL_HEADER);
aacplus_cfg->inputFormat = AACPLUS_INPUT_16BIT;
if (!aacplusEncSetConfiguration(s->aacplus_handle, aacplus_cfg)) {
av_log(avctx, AV_LOG_ERROR, "libaacplus doesn't support this output format!\n");
return -1;
}
avctx->frame_size = samples_input / avctx->channels;
avctx->coded_frame= avcodec_alloc_frame();
avctx->coded_frame->key_frame= 1;
/* Set decoder specific info */
avctx->extradata_size = 0;
if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
unsigned char *buffer = NULL;
unsigned long decoder_specific_info_size;
if (aacplusEncGetDecoderSpecificInfo(s->aacplus_handle, &buffer,
&decoder_specific_info_size) == 1) {
avctx->extradata = av_malloc(decoder_specific_info_size + FF_INPUT_BUFFER_PADDING_SIZE);
avctx->extradata_size = decoder_specific_info_size;
memcpy(avctx->extradata, buffer, avctx->extradata_size);
}
#undef free
free(buffer);
#define free please_use_av_free
}
return 0;
}