Java Code Examples for android.media.AudioFormat#ENCODING_PCM_8BIT
The following examples show how to use
android.media.AudioFormat#ENCODING_PCM_8BIT .
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Example 1
| Source File: MicrophoneCollector.java From sensordatacollector with GNU General Public License v2.0 | 6 votes |
|
private AudioRecord findAudioRecord()
{
for(int rate : mSampleRates) {
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_DEFAULT, AudioFormat.CHANNEL_IN_MONO }) {
try {
Log.d("MicrophoneCollector", "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.DEFAULT, rate, channelConfig, audioFormat, bufferSize);
if(recorder.getState() == AudioRecord.STATE_INITIALIZED)
return recorder;
}
} catch(Exception e) {
Log.e("MicrophoneCollector", rate + " Exception, keep trying.", e);
}
}
}
}
return null;
}
Example 2
| Source File: Microphone.java From ssj with GNU General Public License v3.0 | 6 votes |
|
public static Cons.Type audioFormatSampleType(int f)
{
switch (f)
{
case AudioFormat.ENCODING_PCM_8BIT:
return Cons.Type.CHAR;
case AudioFormat.ENCODING_PCM_16BIT:
case AudioFormat.ENCODING_DEFAULT:
return Cons.Type.SHORT;
case AudioFormat.ENCODING_PCM_FLOAT:
return Cons.Type.FLOAT;
case AudioFormat.ENCODING_INVALID:
default:
return Cons.Type.UNDEF;
}
}
Example 3
| Source File: Microphone.java From ssj with GNU General Public License v3.0 | 6 votes |
|
public static int audioFormatSampleBytes(int f)
{
switch (f)
{
case AudioFormat.ENCODING_PCM_8BIT:
return 1;
case AudioFormat.ENCODING_PCM_16BIT:
case AudioFormat.ENCODING_DEFAULT:
return 2;
case AudioFormat.ENCODING_PCM_FLOAT:
return 4;
case AudioFormat.ENCODING_INVALID:
default:
return 0;
}
}
Example 4
| Source File: AudioTrackPlayerImpl.java From dcs-sdk-java with Apache License 2.0 | 6 votes |
|
private int getMinBufferSize(int sampleRate, int channelConfig, int audioFormat) {
minBufferSize = AudioTrack.getMinBufferSize(sampleRate, channelConfig, audioFormat);
// 解决异常IllegalArgumentException: Invalid audio buffer size
int channelCount = 1;
switch (channelConfig) {
// AudioFormat.CHANNEL_CONFIGURATION_DEFAULT
case AudioFormat.CHANNEL_OUT_DEFAULT:
case AudioFormat.CHANNEL_OUT_MONO:
case AudioFormat.CHANNEL_CONFIGURATION_MONO:
channelCount = 1;
break;
case AudioFormat.CHANNEL_OUT_STEREO:
case AudioFormat.CHANNEL_CONFIGURATION_STEREO:
channelCount = 2;
break;
default:
channelCount = Integer.bitCount(channelConfig);
}
// 判断minBufferSize是否在范围内，如果不在设定默认值为1152
int frameSizeInBytes = channelCount * (audioFormat == AudioFormat.ENCODING_PCM_8BIT ? 1 : 2);
if ((minBufferSize % frameSizeInBytes != 0) || (minBufferSize < 1)) {
minBufferSize = 1152;
}
return minBufferSize;
}
Example 5
| Source File: AudioSaveHelper.java From Android-AudioRecorder-App with Apache License 2.0 | 5 votes |
|
/**
* Writes the proper 44-byte RIFF/WAVE header to/for the given stream
* Two size fields are left empty/null since we do not yet know the final stream size
*
* @param out The stream to write the header to
* @param channelMask An AudioFormat.CHANNEL_* mask
* @param sampleRate The sample rate in hertz
* @param encoding An AudioFormat.ENCODING_PCM_* value
* @throws IOException
*/
private void writeWavHeader(OutputStream out, int channelMask, int sampleRate, int encoding)
throws IOException {
short channels;
switch (channelMask) {
case AudioFormat.CHANNEL_IN_MONO:
channels = 1;
break;
case AudioFormat.CHANNEL_IN_STEREO:
channels = 2;
break;
default:
throw new IllegalArgumentException("Unacceptable channel mask");
}
short bitDepth;
switch (encoding) {
case AudioFormat.ENCODING_PCM_8BIT:
bitDepth = 8;
break;
case AudioFormat.ENCODING_PCM_16BIT:
bitDepth = 16;
break;
case AudioFormat.ENCODING_PCM_FLOAT:
bitDepth = 32;
break;
default:
throw new IllegalArgumentException("Unacceptable encoding");
}
writeWavHeader(out, channels, sampleRate, bitDepth);
}
Example 6
| Source File: AudioRecordConfig.java From OmRecorder with Apache License 2.0 | 5 votes |
|
@Override public byte bitsPerSample() {
if (audioEncoding == AudioFormat.ENCODING_PCM_16BIT) {
return 16;
} else if (audioEncoding == AudioFormat.ENCODING_PCM_8BIT) {
return 8;
} else {
return 16;
}
}
Example 7
| Source File: RecordAudioTest.java From AndPermission with Apache License 2.0 | 5 votes |
|
public static int[] findAudioParameters() {
for (int rate : RATES) {
for (int channel : new int[]{AudioFormat.CHANNEL_IN_MONO, AudioFormat.CHANNEL_IN_STEREO}) {
for (int format : new int[]{AudioFormat.ENCODING_PCM_8BIT, AudioFormat.ENCODING_PCM_16BIT}) {
int buffer = AudioRecord.getMinBufferSize(rate, channel, format);
if (buffer != AudioRecord.ERROR_BAD_VALUE) {
return new int[]{rate, channel, format, buffer};
}
}
}
}
return null;
}
Example 8
| Source File: WavFileHelper.java From video-quickstart-android with MIT License | 5 votes |
|
/**
* Writes the proper 44-byte RIFF/WAVE header to/for the given stream Two size fields are left
* empty/null since we do not yet know the final stream size
*
* @param out The stream to write the header to
* @param channelMask An AudioFormat.CHANNEL_* mask
* @param sampleRate The sample rate in hertz
* @param encoding An AudioFormat.ENCODING_PCM_* value
* @throws IOException
*/
private static void writeWavHeader(
OutputStream out, int channelMask, int sampleRate, int encoding) throws IOException {
short channels;
switch (channelMask) {
case AudioFormat.CHANNEL_IN_MONO:
channels = 1;
break;
case AudioFormat.CHANNEL_IN_STEREO:
channels = 2;
break;
default:
throw new IllegalArgumentException("Unacceptable channel mask");
}
short bitDepth;
switch (encoding) {
case AudioFormat.ENCODING_PCM_8BIT:
bitDepth = 8;
break;
case AudioFormat.ENCODING_PCM_16BIT:
bitDepth = 16;
break;
case AudioFormat.ENCODING_PCM_FLOAT:
bitDepth = 32;
break;
default:
throw new IllegalArgumentException("Unacceptable encoding");
}
writeWavHeader(out, channels, sampleRate, bitDepth);
}
Example 9
| Source File: MediaAudioEncoder.java From EZFilter with MIT License | 5 votes |
|
private int getBitsPerSample(int audioFormat) {
int bitsPerSample;
switch (audioFormat) {
case AudioFormat.ENCODING_PCM_16BIT:
bitsPerSample = 16;
break;
case AudioFormat.ENCODING_PCM_8BIT:
bitsPerSample = 8;
break;
default:
bitsPerSample = 16;
break;
}
return bitsPerSample;
}
Example 10
| Source File: MediaAudioEncoder.java From EZFilter with MIT License | 5 votes |
|
/**
* 查找可用的音频录制器
*
* @return
*/
private AudioRecord findAudioRecord() {
int[] samplingRates = new int[]{44100, 22050, 11025, 8000};
int[] audioFormats = new int[]{
AudioFormat.ENCODING_PCM_16BIT,
AudioFormat.ENCODING_PCM_8BIT};
int[] channelConfigs = new int[]{
AudioFormat.CHANNEL_IN_STEREO,
AudioFormat.CHANNEL_IN_MONO};
for (int rate : samplingRates) {
for (int format : audioFormats) {
for (int config : channelConfigs) {
try {
int bufferSize = AudioRecord.getMinBufferSize(rate, config, format);
if (bufferSize != AudioRecord.ERROR_BAD_VALUE) {
for (int source : AUDIO_SOURCES) {
AudioRecord recorder = new AudioRecord(source, rate, config, format, bufferSize * 4);
if (recorder.getState() == AudioRecord.STATE_INITIALIZED) {
mSamplingRate = rate;
return recorder;
}
}
}
} catch (Exception e) {
Log.e(TAG, "Init AudioRecord Error." + Log.getStackTraceString(e));
}
}
}
}
return null;
}
Example 11
| Source File: BaseAudioDecoder.java From sdl_java_suite with BSD 3-Clause "New" or "Revised" License | 5 votes |
|
protected void onOutputFormatChanged(@NonNull MediaFormat mediaFormat) {
if (mediaFormat.containsKey(MediaFormat.KEY_CHANNEL_COUNT)) {
outputChannelCount = mediaFormat.getInteger(MediaFormat.KEY_CHANNEL_COUNT);
}
if (mediaFormat.containsKey(MediaFormat.KEY_SAMPLE_RATE)) {
outputSampleRate = mediaFormat.getInteger(MediaFormat.KEY_SAMPLE_RATE);
}
if (android.os.Build.VERSION.SDK_INT >= android.os.Build.VERSION_CODES.N && mediaFormat.containsKey(MediaFormat.KEY_PCM_ENCODING)) {
int key = mediaFormat.getInteger(MediaFormat.KEY_PCM_ENCODING);
switch (key) {
case AudioFormat.ENCODING_PCM_8BIT:
outputSampleType = SampleType.UNSIGNED_8_BIT;
break;
case AudioFormat.ENCODING_PCM_FLOAT:
outputSampleType = SampleType.FLOAT;
break;
case AudioFormat.ENCODING_PCM_16BIT:
default:
// by default we fallback to signed 16 bit samples
outputSampleType = SampleType.SIGNED_16_BIT;
break;
}
} else {
outputSampleType = SampleType.SIGNED_16_BIT;
}
}
Example 12
| Source File: AudioProcess.java From NoiseCapture with GNU General Public License v3.0 | 5 votes |
|
/**
* Constructor
* @param recording Recording state
* @param canceled Canceled state
* @param customLeqProcessing Custom receiver of sound signals
*/
public AudioProcess(AtomicBoolean recording, AtomicBoolean canceled, ProcessingThread
customLeqProcessing) {
this.recording = recording;
this.canceled = canceled;
this.customLeqProcessing = customLeqProcessing;
final int[] mSampleRates = new int[] {44100}; // AWeigting coefficient are based on 44100
// Hz sampling rate, so we do not support other samplings (22050, 16000, 11025,8000)
final int[] encodings = new int[] { AudioFormat.ENCODING_PCM_16BIT , AudioFormat.ENCODING_PCM_8BIT };
final short[] audioChannels = new short[] { AudioFormat.CHANNEL_IN_MONO, AudioFormat.CHANNEL_IN_STEREO };
for (int tryRate : mSampleRates) {
for (int tryEncoding : encodings) {
for(short tryAudioChannel : audioChannels) {
int tryBufferSize = AudioRecord.getMinBufferSize(tryRate,
tryAudioChannel, tryEncoding);
if (tryBufferSize != AudioRecord.ERROR_BAD_VALUE) {
// Take a higher buffer size in order to get a smooth recording under load
// avoiding Buffer overflow error on AudioRecord side.
bufferSize = Math.max(tryBufferSize,
(int)(AcousticIndicators.TIMEPERIOD_FAST * tryRate));
encoding = tryEncoding;
audioChannel = tryAudioChannel;
rate = tryRate;
this.fastLeqProcessing = new LeqProcessingThread(this,
AcousticIndicators.TIMEPERIOD_FAST, true,
hannWindowFast ? FFTSignalProcessing.WINDOW_TYPE.TUKEY :
FFTSignalProcessing.WINDOW_TYPE.RECTANGULAR, PROP_MOVING_SPECTRUM, true);
this.slowLeqProcessing = new LeqProcessingThread(this,
AcousticIndicators.TIMEPERIOD_SLOW, true,
hannWindowOneSecond ? FFTSignalProcessing.WINDOW_TYPE.TUKEY :
FFTSignalProcessing.WINDOW_TYPE.RECTANGULAR,
PROP_DELAYED_STANDART_PROCESSING, false);
return;
}
}
}
}
throw new IllegalStateException("This device is not compatible");
}
Example 13
| Source File: RecordAudioTester.java From PermissionAgent with Apache License 2.0 | 5 votes |
|
private static AudioRecord findAudioRecord() {
for (int rate : RATES) {
for (short format : new short[] {AudioFormat.ENCODING_PCM_8BIT, AudioFormat.ENCODING_PCM_16BIT}) {
for (short channel : new short[] {AudioFormat.CHANNEL_IN_MONO, AudioFormat.CHANNEL_IN_STEREO}) {
int buffer = AudioRecord.getMinBufferSize(rate, channel, format);
if (buffer != AudioRecord.ERROR_BAD_VALUE) {
AudioRecord recorder = new AudioRecord(MediaRecorder.AudioSource.MIC, rate, channel, format,
buffer);
if (recorder.getState() == AudioRecord.STATE_INITIALIZED) return recorder;
}
}
}
}
return null;
}
Example 14
| Source File: MainActivity.java From android-fskmodem with GNU General Public License v3.0 | 4 votes |
|
@Override
protected void onCreate(Bundle savedInstanceState) {
super.onCreate(savedInstanceState);
setContentView(R.layout.activity_main);
/// INIT FSK CONFIG
try {
mConfig = new FSKConfig(FSKConfig.SAMPLE_RATE_44100, FSKConfig.PCM_8BIT, FSKConfig.CHANNELS_MONO, FSKConfig.SOFT_MODEM_MODE_4, FSKConfig.THRESHOLD_20P);
} catch (IOException e1) {
e1.printStackTrace();
}
/// INIT FSK DECODER
mDecoder = new FSKDecoder(mConfig, new FSKDecoderCallback() {
@Override
public void decoded(byte[] newData) {
final String text = new String(newData);
runOnUiThread(new Runnable() {
public void run() {
TextView view = ((TextView) findViewById(R.id.result));
view.setText(view.getText()+text);
}
});
}
});
/// INIT FSK ENCODER
mEncoder = new FSKEncoder(mConfig, new FSKEncoderCallback() {
@Override
public void encoded(byte[] pcm8, short[] pcm16) {
if (mConfig.pcmFormat == FSKConfig.PCM_8BIT) {
//8bit buffer is populated, 16bit buffer is null
mAudioTrack.write(pcm8, 0, pcm8.length);
mDecoder.appendSignal(pcm8);
}
else if (mConfig.pcmFormat == FSKConfig.PCM_16BIT) {
//16bit buffer is populated, 8bit buffer is null
mAudioTrack.write(pcm16, 0, pcm16.length);
mDecoder.appendSignal(pcm16);
}
}
});
///
mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,
mConfig.sampleRate, AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_8BIT, 1024,
AudioTrack.MODE_STREAM);
mAudioTrack.play();
///
new Thread(mDataFeeder).start();
}
Example 15
| Source File: Record.java From SinVoiceDemo with Apache License 2.0 | 4 votes |
|
public void start() {
if (STATE_STOP == mState) {
mState = STATE_START;
switch (mChannel) {
case CHANNEL_1:
mChannelConfig = AudioFormat.CHANNEL_IN_MONO;
break;
case CHANNEL_2:
mChannelConfig = AudioFormat.CHANNEL_IN_STEREO;
break;
}
switch (mBits) {
case BITS_8:
mAudioEncoding = AudioFormat.ENCODING_PCM_8BIT;
break;
case BITS_16:
mAudioEncoding = AudioFormat.ENCODING_PCM_16BIT;
break;
}
int minBufferSize = AudioRecord.getMinBufferSize(mFrequence, mChannelConfig, mAudioEncoding);
LogHelper.d(TAG, "minBufferSize:" + minBufferSize);
AudioRecord record = new AudioRecord(MediaRecorder.AudioSource.MIC, mFrequence, mChannelConfig, mAudioEncoding, mBufferSize);
record.startRecording();
LogHelper.d(TAG, "record start");
if (null != mCallback) {
if (null != mListener) {
mListener.onStartRecord();
}
while (STATE_START == mState) {
BufferData data = mCallback.getRecordBuffer();
if (null != data) {
if (null != data.byteData) {
int bufferReadResult = record.read(data.byteData, 0, mBufferSize);
data.setFilledSize(bufferReadResult);
mCallback.freeRecordBuffer(data);
} else {
// end of input
LogHelper.d(TAG, "get end input data, so stop");
break;
}
} else {
LogHelper.d(TAG, "get null data");
break;
}
}
if (null != mListener) {
mListener.onStopRecord();
}
}
record.stop();
record.release();
LogHelper.d(TAG, "record stop");
}
}
Example 16
| Source File: RNAudioRecordModule.java From react-native-audio-record with MIT License | 4 votes |
|
private void addWavHeader(FileOutputStream out, long totalAudioLen, long totalDataLen)
throws Exception {
long sampleRate = sampleRateInHz;
int channels = channelConfig == AudioFormat.CHANNEL_IN_MONO ? 1 : 2;
int bitsPerSample = audioFormat == AudioFormat.ENCODING_PCM_8BIT ? 8 : 16;
long byteRate = sampleRate * channels * bitsPerSample / 8;
int blockAlign = channels * bitsPerSample / 8;
byte[] header = new byte[44];
header[0] = 'R'; // RIFF chunk
header[1] = 'I';
header[2] = 'F';
header[3] = 'F';
header[4] = (byte) (totalDataLen & 0xff); // how big is the rest of this file
header[5] = (byte) ((totalDataLen >> 8) & 0xff);
header[6] = (byte) ((totalDataLen >> 16) & 0xff);
header[7] = (byte) ((totalDataLen >> 24) & 0xff);
header[8] = 'W'; // WAVE chunk
header[9] = 'A';
header[10] = 'V';
header[11] = 'E';
header[12] = 'f'; // 'fmt ' chunk
header[13] = 'm';
header[14] = 't';
header[15] = ' ';
header[16] = 16; // 4 bytes: size of 'fmt ' chunk
header[17] = 0;
header[18] = 0;
header[19] = 0;
header[20] = 1; // format = 1 for PCM
header[21] = 0;
header[22] = (byte) channels; // mono or stereo
header[23] = 0;
header[24] = (byte) (sampleRate & 0xff); // samples per second
header[25] = (byte) ((sampleRate >> 8) & 0xff);
header[26] = (byte) ((sampleRate >> 16) & 0xff);
header[27] = (byte) ((sampleRate >> 24) & 0xff);
header[28] = (byte) (byteRate & 0xff); // bytes per second
header[29] = (byte) ((byteRate >> 8) & 0xff);
header[30] = (byte) ((byteRate >> 16) & 0xff);
header[31] = (byte) ((byteRate >> 24) & 0xff);
header[32] = (byte) blockAlign; // bytes in one sample, for all channels
header[33] = 0;
header[34] = (byte) bitsPerSample; // bits in a sample
header[35] = 0;
header[36] = 'd'; // beginning of the data chunk
header[37] = 'a';
header[38] = 't';
header[39] = 'a';
header[40] = (byte) (totalAudioLen & 0xff); // how big is this data chunk
header[41] = (byte) ((totalAudioLen >> 8) & 0xff);
header[42] = (byte) ((totalAudioLen >> 16) & 0xff);
header[43] = (byte) ((totalAudioLen >> 24) & 0xff);
out.write(header, 0, 44);
}
Example 17
| Source File: RNAudioRecordModule.java From react-native-audio-record with MIT License | 4 votes |
|
@ReactMethod
public void init(ReadableMap options) {
sampleRateInHz = 44100;
if (options.hasKey("sampleRate")) {
sampleRateInHz = options.getInt("sampleRate");
}
channelConfig = AudioFormat.CHANNEL_IN_MONO;
if (options.hasKey("channels")) {
if (options.getInt("channels") == 2) {
channelConfig = AudioFormat.CHANNEL_IN_STEREO;
}
}
audioFormat = AudioFormat.ENCODING_PCM_16BIT;
if (options.hasKey("bitsPerSample")) {
if (options.getInt("bitsPerSample") == 8) {
audioFormat = AudioFormat.ENCODING_PCM_8BIT;
}
}
audioSource = AudioSource.VOICE_RECOGNITION;
if (options.hasKey("audioSource")) {
audioSource = options.getInt("audioSource");
}
String documentDirectoryPath = getReactApplicationContext().getFilesDir().getAbsolutePath();
outFile = documentDirectoryPath + "/" + "audio.wav";
tmpFile = documentDirectoryPath + "/" + "temp.pcm";
if (options.hasKey("wavFile")) {
String fileName = options.getString("wavFile");
outFile = documentDirectoryPath + "/" + fileName;
}
isRecording = false;
eventEmitter = reactContext.getJSModule(DeviceEventManagerModule.RCTDeviceEventEmitter.class);
bufferSize = AudioRecord.getMinBufferSize(sampleRateInHz, channelConfig, audioFormat);
int recordingBufferSize = bufferSize * 3;
recorder = new AudioRecord(audioSource, sampleRateInHz, channelConfig, audioFormat, recordingBufferSize);
}
Example 18
| Source File: PlaybackSynthesisCallback.java From android_9.0.0_r45 with Apache License 2.0 | 4 votes |
|
@Override
public int start(int sampleRateInHz, int audioFormat, int channelCount) {
if (DBG) Log.d(TAG, "start(" + sampleRateInHz + "," + audioFormat + "," + channelCount
+ ")");
if (audioFormat != AudioFormat.ENCODING_PCM_8BIT &&
audioFormat != AudioFormat.ENCODING_PCM_16BIT &&
audioFormat != AudioFormat.ENCODING_PCM_FLOAT) {
Log.w(TAG, "Audio format encoding " + audioFormat + " not supported. Please use one " +
"of AudioFormat.ENCODING_PCM_8BIT, AudioFormat.ENCODING_PCM_16BIT or " +
"AudioFormat.ENCODING_PCM_FLOAT");
}
mDispatcher.dispatchOnBeginSynthesis(sampleRateInHz, audioFormat, channelCount);
int channelConfig = BlockingAudioTrack.getChannelConfig(channelCount);
synchronized (mStateLock) {
if (channelConfig == 0) {
Log.e(TAG, "Unsupported number of channels :" + channelCount);
mStatusCode = TextToSpeech.ERROR_OUTPUT;
return TextToSpeech.ERROR;
}
if (mStatusCode == TextToSpeech.STOPPED) {
if (DBG) Log.d(TAG, "stop() called before start(), returning.");
return errorCodeOnStop();
}
if (mStatusCode != TextToSpeech.SUCCESS) {
if (DBG) Log.d(TAG, "Error was raised");
return TextToSpeech.ERROR;
}
if (mItem != null) {
Log.e(TAG, "Start called twice");
return TextToSpeech.ERROR;
}
SynthesisPlaybackQueueItem item = new SynthesisPlaybackQueueItem(
mAudioParams, sampleRateInHz, audioFormat, channelCount,
mDispatcher, mCallerIdentity, mLogger);
mAudioTrackHandler.enqueue(item);
mItem = item;
}
return TextToSpeech.SUCCESS;
}
Example 19
| Source File: AudioStreamManagerTest.java From sdl_java_suite with BSD 3-Clause "New" or "Revised" License | 4 votes |
|
public void testOutputFormatChanged() {
BaseAudioDecoder mockDecoder = mock(BaseAudioDecoder.class, Mockito.CALLS_REAL_METHODS);
try {
Field outputChannelCountField = BaseAudioDecoder.class.getDeclaredField("outputChannelCount");
Field outputSampleRateField = BaseAudioDecoder.class.getDeclaredField("outputSampleRate");
Field outputSampleTypeField = BaseAudioDecoder.class.getDeclaredField("outputSampleType");
outputChannelCountField.setAccessible(true);
outputSampleRateField.setAccessible(true);
outputSampleTypeField.setAccessible(true);
// channel count, sample rate, sample type
int key_channel_count = 0, key_sample_rate = 1, key_sample_type = 2, key_sample_type_result = 3;
int[][] tests = new int[][] {
{ 47, 42000, AudioFormat.ENCODING_PCM_8BIT, SampleType.UNSIGNED_8_BIT },
{ 2, 16000, AudioFormat.ENCODING_PCM_16BIT, SampleType.SIGNED_16_BIT },
{ 1, 22050, AudioFormat.ENCODING_PCM_FLOAT, SampleType.FLOAT },
{ 3, 48000, AudioFormat.ENCODING_INVALID, SampleType.SIGNED_16_BIT },
};
for (int[] test : tests) {
int channel_count = test[key_channel_count];
int sample_rate = test[key_sample_rate];
int sample_type = test[key_sample_type];
int sample_type_result = test[key_sample_type_result];
MediaFormat format = new MediaFormat();
format.setInteger(MediaFormat.KEY_CHANNEL_COUNT, channel_count);
format.setInteger(MediaFormat.KEY_SAMPLE_RATE, sample_rate);
format.setInteger(MediaFormat.KEY_PCM_ENCODING, sample_type);
// in case the phone version is old the method does not take sample type into account but
// always expected 16 bit. See https://developer.android.com/reference/android/media/MediaFormat.html#KEY_PCM_ENCODING
if (android.os.Build.VERSION.SDK_INT < android.os.Build.VERSION_CODES.N) {
sample_type_result = SampleType.SIGNED_16_BIT;
}
mockDecoder.onOutputFormatChanged(format);
int output_channel_count = outputChannelCountField.getInt(mockDecoder);
int output_sample_rate = outputSampleRateField.getInt(mockDecoder);
int output_sample_type = outputSampleTypeField.getInt(mockDecoder);
// changing from assertEquals to if and fail so travis gives better results
if (channel_count != output_channel_count) {
fail("AssertEqualsFailed: channel_count == output_channel_count (" + channel_count + " == " + output_channel_count + ")");
}
if (sample_rate != output_sample_rate) {
fail("AssertEqualsFailed: sample_rate == output_sample_rate (" + sample_rate + " == " + output_sample_rate + ")");
}
if (sample_type_result != output_sample_type) {
fail("Assert: sample_type_result == output_sample_type (" + sample_type_result + " == " + output_sample_type + ")");
}
}
} catch (Exception e) {
e.printStackTrace();
fail();
}
}
Example 20
| Source File: AudioQuality.java From DeviceConnect-Android with MIT License | 3 votes |
|
/**
* 音声のフォーマットを設定します.
*
* <p>
* デフォルトでは、 {@link AudioFormat#ENCODING_PCM_16BIT} が設定されています。
* </p>
*
* @param format {@link AudioFormat#ENCODING_PCM_16BIT} or {@link AudioFormat#ENCODING_PCM_8BIT}
*/
public void setFormat(int format) {
if (format != AudioFormat.ENCODING_PCM_16BIT && format != AudioFormat.ENCODING_PCM_8BIT) {
throw new IllegalArgumentException("Not supported a format. format=" + format);
}
mFormat = format;
}
