我一直在关注使用jni在Android上使用LAME mp3的this tutorial。录音似乎正在工作，我得到一个输出为mp3，但在播放时音频已经减慢并放下。
我试图将所有相关代码放在下面。有关为何发生这种情况的任何指导？在此先感谢您的帮助。
编辑：确定以便检查我是否将原始数据导入Audacity并且播放效果正常，因此在编码阶段这一定是一个问题。
Java类：
public class Record extends Activity implements OnClickListener {
static {
System.loadLibrary("mp3lame");
}
private native void initEncoder(int numChannels, int sampleRate, int bitRate, int mode, int quality);
private native void destroyEncoder();
private native int encodeFile(String sourcePath, String targetPath);
private static final int RECORDER_BPP = 16;
private static final String AUDIO_RECORDER_FILE_EXT_WAV = ".wav";
private static final String AUDIO_RECORDER_FOLDER = "AberdeenSoundsites";
private static final String AUDIO_RECORDER_TEMP_FILE = "record_temp.raw";
private static final int[] RECORDER_SAMPLERATES = {44100, 22050, 11025, 8000};
private static final int RECORDER_CHANNELS = AudioFormat.CHANNEL_IN_STEREO;
private static final int RECORDER_AUDIO_ENCODING = AudioFormat.ENCODING_PCM_16BIT;
public static final int NUM_CHANNELS = 2;
public static final int SAMPLE_RATE = 44100;
public static final int BITRATE = 320;
public static final int MODE = 1;
public static final int QUALITY = 2;
private short[] mBuffer;
private File rawFile;
private File encodedFile;
private int sampleRate;
private String filename;
private AudioRecord recorder = null;
private int bufferSize = 0;
private Thread recordingThread = null;
private boolean isRecording = false;
@Override
public void onCreate(Bundle savedInstanceState) {
super.onCreate(savedInstanceState);
setContentView(R.layout.record);
initEncoder(NUM_CHANNELS, SAMPLE_RATE, BITRATE, MODE, QUALITY);
stopButton = (Button) findViewById(R.id.stop_button);
stopButton.setOnClickListener(this);
timer = (TextView) findViewById(R.id.recording_time);
bufferSize = AudioRecord.getMinBufferSize(44100, RECORDER_CHANNELS, RECORDER_AUDIO_ENCODING);
}
private void startRecording() {
stopped = false;
stopButton.setText(R.string.stop_button_label);
// Set up and start audio recording
recorder = findAudioRecord();
recorder.startRecording();
isRecording = true;
rawFile = getFile("raw");
mBuffer = new short[bufferSize];
startBufferedWrite(rawFile);
}
private void stopRecording() {
mHandler.removeCallbacks(startTimer);
stopped = true;
if(recorder != null){
isRecording = false;
recorder.stop();
recorder.release();
recorder = null;
recordingThread = null;
}
encodedFile = getFile("mp3");
int result = encodeFile(rawFile.getAbsolutePath(), encodedFile.getAbsolutePath());
if (result == 0) {
Toast.makeText(Record.this, "Encoded to " + encodedFile.getName(), Toast.LENGTH_SHORT)
.show();
}
}
private void startBufferedWrite(final File file) {
new Thread(new Runnable() {
@Override
public void run() {
Looper.prepare();
DataOutputStream output = null;
try {
output = new DataOutputStream(new BufferedOutputStream(new FileOutputStream(file)));
while (isRecording) {
int readSize = recorder.read(mBuffer, 0, mBuffer.length);
for (int i = 0; i < readSize; i++) {
output.writeShort(mBuffer[i]);
}
}
} catch (IOException e) {
Toast.makeText(Record.this, e.getMessage(), Toast.LENGTH_SHORT).show();
} finally {
if (output != null) {
try {
output.flush();
} catch (IOException e) {
Toast.makeText(Record.this, e.getMessage(), Toast.LENGTH_SHORT).show();
} finally {
try {
output.close();
} catch (IOException e) {
Toast.makeText(Record.this, e.getMessage(), Toast.LENGTH_SHORT).show();
}
}
}
}
}
}).start();
}
private File getFile(final String suffix) {
Time time = new Time();
time.setToNow();
return new File(Environment.getExternalStorageDirectory()+"/MyAppFolder", time.format("%Y%m%d%H%M%S") + "." + suffix);
}
public AudioRecord findAudioRecord() {
for (int rate : RECORDER_SAMPLERATES) {
for (short audioFormat : new short[] { AudioFormat.ENCODING_PCM_16BIT, AudioFormat.ENCODING_PCM_8BIT }) {
for (short channelConfig : new short[] { AudioFormat.CHANNEL_IN_STEREO, AudioFormat.CHANNEL_IN_MONO }) {
try {
Log.d("AberdeenSoundsites", "Attempting rate " + rate + "Hz, bits: " + audioFormat + ", channel: "
+ channelConfig);
int bufferSize = AudioRecord.getMinBufferSize(rate, channelConfig, audioFormat);
if (bufferSize != AudioRecord.ERROR_BAD_VALUE) {
// check if we can instantiate and have a success
AudioRecord recorder = new AudioRecord(MediaRecorder.AudioSource.MIC, rate, channelConfig, audioFormat, bufferSize);
sampleRate = rate;
if (recorder.getState() == AudioRecord.STATE_INITIALIZED)
return recorder;
}
} catch (Exception e) {
Log.e("MyApp", rate + "Exception, keep trying.",e);
}
}
}
}
Log.e("MyApp", "No settings worked :(");
return null;
}
C包装器：
#include <stdio.h>
#include <stdlib.h>
#include <jni.h>
#include <android/log.h>
#include "libmp3lame/lame.h"
#define LOG_TAG "LAME ENCODER"
#define LOGD(format, args...) __android_log_print(ANDROID_LOG_DEBUG, LOG_TAG, format, ##args);
#define BUFFER_SIZE 8192
#define be_short(s) ((short) ((unsigned short) (s) << 8) | ((unsigned short) (s) >> 8))
lame_t lame;
int read_samples(FILE *input_file, short *input) {
int nb_read;
nb_read = fread(input, 1, sizeof(short), input_file) / sizeof(short);
int i = 0;
while (i < nb_read) {
input[i] = be_short(input[i]);
i++;
}
return nb_read;
}
void Java_myPacakage_myApp_Record_initEncoder(JNIEnv *env,
jobject jobj, jint in_num_channels, jint in_samplerate, jint in_brate,
jint in_mode, jint in_quality) {
lame = lame_init();
LOGD("Init parameters:");
lame_set_num_channels(lame, in_num_channels);
LOGD("Number of channels: %d", in_num_channels);
lame_set_in_samplerate(lame, in_samplerate);
LOGD("Sample rate: %d", in_samplerate);
lame_set_brate(lame, in_brate);
LOGD("Bitrate: %d", in_brate);
lame_set_mode(lame, in_mode);
LOGD("Mode: %d", in_mode);
lame_set_quality(lame, in_quality);
LOGD("Quality: %d", in_quality);
int res = lame_init_params(lame);
LOGD("Init returned: %d", res);
}
void Java_myPacakage_myApp_Record_destroyEncoder(
JNIEnv *env, jobject jobj) {
int res = lame_close(lame);
LOGD("Deinit returned: %d", res);
}
void Java_myPacakage_myApp_Record_encodeFile(JNIEnv *env,
jobject jobj, jstring in_source_path, jstring in_target_path) {
const char *source_path, *target_path;
source_path = (*env)->GetStringUTFChars(env, in_source_path, NULL);
target_path = (*env)->GetStringUTFChars(env, in_target_path, NULL);
FILE *input_file, *output_file;
input_file = fopen(source_path, "rb");
output_file = fopen(target_path, "wb");
short input[BUFFER_SIZE];
char output[BUFFER_SIZE];
int nb_read = 0;
int nb_write = 0;
int nb_total = 0;
LOGD("Encoding started");
while (nb_read = read_samples(input_file, input)) {
nb_write = lame_encode_buffer(lame, input, input, nb_read, output,
BUFFER_SIZE);
fwrite(output, nb_write, 1, output_file);
nb_total += nb_write;
}
LOGD("Encoded %d bytes", nb_total);
nb_write = lame_encode_flush(lame, output, BUFFER_SIZE);
fwrite(output, nb_write, 1, output_file);
LOGD("Flushed %d bytes", nb_write);
fclose(input_file);
fclose(output_file);
}
编辑 - 好吧，出于兴趣，我下载了教程提供给我手机的apk并运行它。这很好。因此，这表明教程中的问题较少，而且我已经完成了更多的事情。当我有一些时间可用时，我将重新审视这一点，看看我是否能确定出错的地方
答案 0 :(得分：3)
为了给你一个指针，你需要调用lame_encode_buffer_interleaved（）如果你使用2个通道（.stereo）来记录。
我花了几天时间弄明白，这是你可以使用的代码：
if (lame_get_num_channels(glf) == 2)
{
result = lame_encode_buffer_interleaved(glf, j_buffer_l, samples/2, j_mp3buf, mp3buf_size);
}
else
{
result = lame_encode_buffer(glf, j_buffer_l, j_buffer_r, samples, j_mp3buf, mp3buf_size);
}
答案 1 :(得分：3)
您使用 2 频道调用 initEncoder ，并使用STEREO和MONO初始化 AudioRecord ，但wrapper.c只能处理1个频道：
nb_write = lame_encode_buffer(lame, input, input, nb_read, output, BUFFER_SIZE);
上述代码要求源音频为1声道单声道。如果你想支持STEREO，请注意 lame_encode_buffer 方法
int CDECL lame_encode_buffer (
lame_global_flags* gfp, /* global context handle */
const short int buffer_l [], /* PCM data for left channel */
const short int buffer_r [], /* PCM data for right channel */
const int nsamples, /* number of samples per channel */
unsigned char* mp3buf, /* pointer to encoded MP3 stream */
const int mp3buf_size ); /* number of valid octets in this
stream */
答案 2 :(得分：1)
你刺激我再次看我的问题，我找到了问题。也许这就是为你而发生的事情。检查您正在使用的wav文件的采样率。我假设或看得太快，并认为它说44100;但它是48000！我解决了我的问题：
lame_set_in_samplerate(lame, 48000);
lame_set_out_samplerate(lame, 44100);
由于某些奇怪的原因，您的代码可能没有读取正确的采样率？
答案 3 :(得分：0)
你可以改写
nb_write = lame_encode_buffer(lame, input, input, nb_read, output, BUFFER_SIZE);
到
nb_write = lame_encode_buffer(lame, input1, input2, nb_read, output, BUFFER_SIZE);
并使用2个单声道原始文件作为输入。当然，您必须调整您的encodeFile - 函数，以便它需要两个字符串作为源并处理所有内容两次。