我正在尝试将音频文件解码为 PCM 以与 AudioTrack 一起使用。音频是吱吱作响的，嘈杂的，只是简单的胡言乱语，随机的一秒钟听起来应该是这样，但主要是完全混乱。我不确定我的错误在哪里，是数组如何传递回 playSound 吗？
提前谢谢你，我真的很感谢你在这件事上的帮助，因为它已经让我很恼火了一段时间。
这是我的java代码：
public void init() {
int bufSize = AudioTrack.getMinBufferSize(44100,
AudioFormat.CHANNEL_OUT_STEREO,
AudioFormat.ENCODING_PCM_16BIT);
track = new AudioTrack(AudioManager.STREAM_MUSIC,
44100,
AudioFormat.CHANNEL_OUT_STEREO,
AudioFormat.ENCODING_PCM_16BIT,
bufSize,
AudioTrack.MODE_STREAM);
log("STARTING!!! _________________________ <--");
byte[] array = new byte[bufSize];
try {
fos = new FileOutputStream("/sdcard/acdc.bin");
} catch (IOException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
decoder("/sdcard/acdc.ogg", array);
}
void playSound(byte[] buf, int size) {
try {
fos.write(buf, 0, size);
} catch (IOException e) {
e.printStackTrace();
}
if(track.getPlayState()!=AudioTrack.PLAYSTATE_PLAYING) {
track.play();
}
int wrote = track.write(buf, 0, size);
if (wrote != size)
log("WRITING: " + wrote + " but size was: " + size);
}
这是我的 c 函数：
void Java_com_example_ffmpegsample_MainActivity_decoder(JNIEnv* env, jobject obj,jstring file,jbyteArray array)
{
jboolean isfilenameCopy;
const char * filename = (*env)->GetStringUTFChars(env, file, &isfilenameCopy);
AVCodec *codec;
AVCodecContext *c= NULL;
AVFormatContext *pFormatCtx;
AVCodecContext *pCodecCtx;
int out_size, len;
FILE *f, *outfile;
uint8_t *outbuf;
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
LOGI("HERE");
jclass cls = (*env)->GetObjectClass(env, obj);
LOGI(cls);
jmethodID play = (*env)->GetMethodID(env, cls, "playSound", "([BI)V");//At the begining of your main function
av_init_packet(&avpkt);
av_register_all();
LOGI("AUDIO DECODER");
printf("Audio decoding\n");
int err;
err = av_open_input_file(&pFormatCtx, filename, NULL, 0, NULL);
if (err!=0) {
LOGI("COULD NOT AV_OPEN file");
return;
}
if(av_find_stream_info(pFormatCtx)<0) {
LOGE("Unable to get stream info");
return;
}
int audioStream = -1;
int i;
for (i=0; i<pFormatCtx->nb_streams; i++) {
if(pFormatCtx->streams[i]->codec->codec_type==CODEC_TYPE_AUDIO) {
audioStream = i;
break;
}
}
if(audioStream==-1) {
LOGE("Unable to find audio stream");
return;
}
LOGI("Audio stream is [%d]", audioStream);
pCodecCtx=pFormatCtx->streams[audioStream]->codec;
codec = avcodec_find_decoder(pCodecCtx->codec_id);
/* find the mpeg audio decoder */
// codec = avcodec_find_decoder(CODEC_ID_AAC);
if (!codec) {
LOGI("NO CODEC");
fprintf(stderr, "codec not found\n");
return;
}
//c= avcodec_alloc_context();
c = pCodecCtx;
/* open it */
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
LOGI("NOT LOADING CODEC");
return;
}
outbuf = malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "could not open %s\n", filename);
LOGI("COULD NOT OPEN FILE");
return;
}
/* decode until eof */
avpkt.data = inbuf;
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (avpkt.size > 0) {
LOGI("............................." + avpkt.size);
out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
len = avcodec_decode_audio3(c, (short *)outbuf, &out_size, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding\n");
LOGI("ERROR DECODING, error: %d", len);
return;
}
if (out_size > 0) {
/* if a frame has been decoded, output it */
jbyte *bytes = (*env)->GetByteArrayElements(env, array, NULL);
memcpy(bytes, outbuf, out_size); //
(*env)->ReleaseByteArrayElements(env, array, bytes, 0);
(*env)->CallVoidMethod(env, obj, play, array, out_size);
}
avpkt.size -= len;
avpkt.data += len;
if (avpkt.size < AUDIO_REFILL_THRESH) {
/* Refill the input buffer, to avoid trying to decode
* incomplete frames. Instead of this, one could also use
* a parser, or use a proper container format through
* libavformat. */
memmove(inbuf, avpkt.data, avpkt.size);
avpkt.data = inbuf;
len = fread(avpkt.data + avpkt.size, 1,
AUDIO_INBUF_SIZE - avpkt.size, f);
if (len > 0)
avpkt.size += len;
}
}
fclose(f);
free(outbuf);
avcodec_close(c);
av_free(c);
}