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Allow setting output sample rate in SonicAudioProcessor
This is not really useful with the DefaultAudioSink, but could be used in a custom AudioSink when mixing audio from sources that have different sample rates. ------------- Created by MOE: https://github.com/google/moe MOE_MIGRATED_REVID=172434482
This commit is contained in:
andrewlewis
Oliver Woodman
parent
e46a7600cf
commit
ac31dc7c7a
10 changed files with 292 additions and 90 deletions
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@ -138,6 +138,11 @@ public final class GvrAudioProcessor implements AudioProcessor {
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return C.ENCODING_PCM_16BIT;
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}
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@Override
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public int getOutputSampleRateHz() {
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return sampleRateHz;
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}
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@Override
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public void queueInput(ByteBuffer input) {
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int position = input.position();
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@ -20,7 +20,15 @@ import java.nio.ByteBuffer;
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import java.nio.ByteOrder;
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/**
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* Interface for audio processors.
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* Interface for audio processors, which take audio data as input and transform it, potentially
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* modifying its channel count, encoding and/or sample rate.
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* <p>
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* Call {@link #configure(int, int, int)} to configure the processor to receive input audio, then
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* call {@link #isActive()} to determine whether the processor is active.
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* {@link #queueInput(ByteBuffer)}, {@link #queueEndOfStream()}, {@link #getOutput()},
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* {@link #isEnded()}, {@link #getOutputChannelCount()}, {@link #getOutputEncoding()} and
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* {@link #getOutputSampleRateHz()} may only be called if the processor is active. Call
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* {@link #reset()} to reset the processor to its unconfigured state.
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*/
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public interface AudioProcessor {
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@ -46,8 +54,9 @@ public interface AudioProcessor {
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* method, {@link #isActive()} returns whether the processor needs to handle buffers; if not, the
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* processor will not accept any buffers until it is reconfigured. Returns {@code true} if the
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* processor must be flushed, or if the value returned by {@link #isActive()} has changed as a
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* result of the call. If it's active, {@link #getOutputChannelCount()} and
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* {@link #getOutputEncoding()} return the processor's output format.
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* result of the call. If it's active, {@link #getOutputSampleRateHz()},
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* {@link #getOutputChannelCount()} and {@link #getOutputEncoding()} return the processor's output
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* format.
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*
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* @param sampleRateHz The sample rate of input audio in Hz.
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* @param channelCount The number of interleaved channels in input audio.
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@ -65,16 +74,27 @@ public interface AudioProcessor {
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boolean isActive();
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/**
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* Returns the number of audio channels in the data output by the processor.
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* Returns the number of audio channels in the data output by the processor. The value may change
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* as a result of calling {@link #configure(int, int, int)} and is undefined if the instance is
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* not active.
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*/
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int getOutputChannelCount();
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/**
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* Returns the audio encoding used in the data output by the processor.
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* Returns the audio encoding used in the data output by the processor. The value may change as a
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* result of calling {@link #configure(int, int, int)} and is undefined if the instance is not
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* active.
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*/
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@C.Encoding
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int getOutputEncoding();
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/**
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* Returns the sample rate of audio output by the processor, in hertz. The value may change as a
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* result of calling {@link #configure(int, int, int)} and is undefined if the instance is not
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* active.
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*/
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int getOutputSampleRateHz();
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/**
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* Queues audio data between the position and limit of the input {@code buffer} for processing.
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* {@code buffer} must be a direct byte buffer with native byte order. Its contents are treated as
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@ -116,7 +136,7 @@ public interface AudioProcessor {
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void flush();
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/**
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* Resets the processor to its initial state.
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* Resets the processor to its unconfigured state.
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*/
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void reset();
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@ -186,12 +186,12 @@ public interface AudioSink {
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/**
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* Configures (or reconfigures) the sink.
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*
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* @param mimeType The MIME type of audio data provided in the input buffers.
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* @param channelCount The number of channels.
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* @param sampleRate The sample rate in Hz.
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* @param pcmEncoding For PCM formats, the encoding used. One of {@link C#ENCODING_PCM_16BIT},
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* {@link C#ENCODING_PCM_16BIT}, {@link C#ENCODING_PCM_24BIT} and
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* {@link C#ENCODING_PCM_32BIT}.
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* @param inputMimeType The MIME type of audio data provided in the input buffers.
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* @param inputChannelCount The number of channels.
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* @param inputSampleRate The sample rate in Hz.
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* @param inputPcmEncoding For PCM formats, the encoding used. One of
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* {@link C#ENCODING_PCM_16BIT}, {@link C#ENCODING_PCM_16BIT}, {@link C#ENCODING_PCM_24BIT}
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* and {@link C#ENCODING_PCM_32BIT}.
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* @param specifiedBufferSize A specific size for the playback buffer in bytes, or 0 to infer a
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* suitable buffer size.
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* @param outputChannels A mapping from input to output channels that is applied to this sink's
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@ -206,9 +206,9 @@ public interface AudioSink {
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* {@link #configure(String, int, int, int, int, int[], int, int)}.
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* @throws ConfigurationException If an error occurs configuring the sink.
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*/
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void configure(String mimeType, int channelCount, int sampleRate, @C.PcmEncoding int pcmEncoding,
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int specifiedBufferSize, @Nullable int[] outputChannels, int trimStartSamples,
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int trimEndSamples) throws ConfigurationException;
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void configure(String inputMimeType, int inputChannelCount, int inputSampleRate,
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@C.PcmEncoding int inputPcmEncoding, int specifiedBufferSize, @Nullable int[] outputChannels,
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int trimStartSamples, int trimEndSamples) throws ConfigurationException;
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/**
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* Starts or resumes consuming audio if initialized.
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@ -103,6 +103,11 @@ import java.util.Arrays;
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return C.ENCODING_PCM_16BIT;
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}
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@Override
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public int getOutputSampleRateHz() {
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return sampleRateHz;
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}
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@Override
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public void queueInput(ByteBuffer inputBuffer) {
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int position = inputBuffer.position();
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@ -148,13 +148,6 @@ public final class DefaultAudioSink implements AudioSink {
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private static final int MIN_PLAYHEAD_OFFSET_SAMPLE_INTERVAL_US = 30000;
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private static final int MIN_TIMESTAMP_SAMPLE_INTERVAL_US = 500000;
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/**
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* The minimum number of output bytes from {@link #sonicAudioProcessor} at which the speedup is
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* calculated using the input/output byte counts from the processor, rather than using the
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* current playback parameters speed.
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*/
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private static final int SONIC_MIN_BYTES_FOR_SPEEDUP = 1024;
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/**
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* Whether to enable a workaround for an issue where an audio effect does not keep its session
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* active across releasing/initializing a new audio track, on platform builds where
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@ -189,6 +182,7 @@ public final class DefaultAudioSink implements AudioSink {
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*/
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private AudioTrack keepSessionIdAudioTrack;
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private AudioTrack audioTrack;
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private int inputSampleRate;
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private int sampleRate;
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private int channelConfig;
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private @C.Encoding int encoding;
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@ -337,14 +331,18 @@ public final class DefaultAudioSink implements AudioSink {
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}
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@Override
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public void configure(String mimeType, int channelCount, int sampleRate,
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@C.PcmEncoding int pcmEncoding, int specifiedBufferSize, @Nullable int[] outputChannels,
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public void configure(String inputMimeType, int inputChannelCount, int inputSampleRate,
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@C.PcmEncoding int inputPcmEncoding, int specifiedBufferSize, @Nullable int[] outputChannels,
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int trimStartSamples, int trimEndSamples) throws ConfigurationException {
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boolean passthrough = !MimeTypes.AUDIO_RAW.equals(mimeType);
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@C.Encoding int encoding = passthrough ? getEncodingForMimeType(mimeType) : pcmEncoding;
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this.inputSampleRate = inputSampleRate;
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int channelCount = inputChannelCount;
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int sampleRate = inputSampleRate;
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@C.Encoding int encoding;
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boolean passthrough = !MimeTypes.AUDIO_RAW.equals(inputMimeType);
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boolean flush = false;
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if (!passthrough) {
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pcmFrameSize = Util.getPcmFrameSize(pcmEncoding, channelCount);
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encoding = inputPcmEncoding;
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pcmFrameSize = Util.getPcmFrameSize(inputPcmEncoding, channelCount);
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trimmingAudioProcessor.setTrimSampleCount(trimStartSamples, trimEndSamples);
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channelMappingAudioProcessor.setChannelMap(outputChannels);
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for (AudioProcessor audioProcessor : availableAudioProcessors) {
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@ -355,12 +353,15 @@ public final class DefaultAudioSink implements AudioSink {
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}
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if (audioProcessor.isActive()) {
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channelCount = audioProcessor.getOutputChannelCount();
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sampleRate = audioProcessor.getOutputSampleRateHz();
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encoding = audioProcessor.getOutputEncoding();
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}
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}
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if (flush) {
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resetAudioProcessors();
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}
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} else {
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encoding = getEncodingForMimeType(inputMimeType);
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}
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int channelConfig;
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@ -598,8 +599,8 @@ public final class DefaultAudioSink implements AudioSink {
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startMediaTimeState = START_IN_SYNC;
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} else {
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// Sanity check that presentationTimeUs is consistent with the expected value.
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long expectedPresentationTimeUs = startMediaTimeUs
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+ framesToDurationUs(getSubmittedFrames());
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long expectedPresentationTimeUs =
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startMediaTimeUs + inputFramesToDurationUs(getSubmittedFrames());
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if (startMediaTimeState == START_IN_SYNC
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&& Math.abs(expectedPresentationTimeUs - presentationTimeUs) > 200000) {
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Log.e(TAG, "Discontinuity detected [expected " + expectedPresentationTimeUs + ", got "
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@ -997,15 +998,11 @@ public final class DefaultAudioSink implements AudioSink {
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return positionUs + playbackParametersOffsetUs - playbackParametersPositionUs;
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}
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if (playbackParametersCheckpoints.isEmpty()
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&& sonicAudioProcessor.getOutputByteCount() >= SONIC_MIN_BYTES_FOR_SPEEDUP) {
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if (playbackParametersCheckpoints.isEmpty()) {
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return playbackParametersOffsetUs
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+ Util.scaleLargeTimestamp(positionUs - playbackParametersPositionUs,
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sonicAudioProcessor.getInputByteCount(), sonicAudioProcessor.getOutputByteCount());
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+ sonicAudioProcessor.scaleDurationForSpeedup(positionUs - playbackParametersPositionUs);
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}
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// We are playing drained data at a previous playback speed, or don't have enough bytes to
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// calculate an accurate speedup, so fall back to multiplying by the speed.
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// We are playing data at a previous playback speed, so fall back to multiplying by the speed.
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return playbackParametersOffsetUs
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+ (long) ((double) playbackParameters.speed * (positionUs - playbackParametersPositionUs));
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}
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@ -1098,6 +1095,10 @@ public final class DefaultAudioSink implements AudioSink {
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return audioTrack != null;
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}
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private long inputFramesToDurationUs(long frameCount) {
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return (frameCount * C.MICROS_PER_SECOND) / inputSampleRate;
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}
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private long framesToDurationUs(long frameCount) {
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return (frameCount * C.MICROS_PER_SECOND) / sampleRate;
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}
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@ -79,6 +79,11 @@ import java.nio.ByteOrder;
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return C.ENCODING_PCM_16BIT;
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}
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@Override
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public int getOutputSampleRateHz() {
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return sampleRateHz;
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}
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@Override
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public void queueInput(ByteBuffer inputBuffer) {
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// Prepare the output buffer.
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@ -32,8 +32,11 @@ import java.util.Arrays;
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private static final int MAXIMUM_PITCH = 400;
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private static final int AMDF_FREQUENCY = 4000;
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private final int sampleRate;
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private final int inputSampleRateHz;
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private final int numChannels;
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private final float speed;
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private final float pitch;
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private final float rate;
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private final int minPeriod;
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private final int maxPeriod;
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private final int maxRequired;
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@ -47,8 +50,6 @@ import java.util.Arrays;
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private short[] pitchBuffer;
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private int oldRatePosition;
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private int newRatePosition;
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private float speed;
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private float pitch;
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private int numInputSamples;
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private int numOutputSamples;
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private int numPitchSamples;
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@ -61,14 +62,18 @@ import java.util.Arrays;
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/**
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* Creates a new Sonic audio stream processor.
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*
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* @param sampleRate The sample rate of input audio.
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* @param inputSampleRateHz The sample rate of input audio, in hertz.
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* @param numChannels The number of channels in the input audio.
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* @param speed The speedup factor for output audio.
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* @param pitch The pitch factor for output audio.
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* @param outputSampleRateHz The sample rate for output audio, in hertz.
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*/
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public Sonic(int sampleRate, int numChannels) {
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this.sampleRate = sampleRate;
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public Sonic(int inputSampleRateHz, int numChannels, float speed, float pitch,
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int outputSampleRateHz) {
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this.inputSampleRateHz = inputSampleRateHz;
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this.numChannels = numChannels;
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minPeriod = sampleRate / MAXIMUM_PITCH;
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maxPeriod = sampleRate / MINIMUM_PITCH;
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minPeriod = inputSampleRateHz / MAXIMUM_PITCH;
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maxPeriod = inputSampleRateHz / MINIMUM_PITCH;
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maxRequired = 2 * maxPeriod;
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downSampleBuffer = new short[maxRequired];
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inputBufferSize = maxRequired;
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@ -80,36 +85,9 @@ import java.util.Arrays;
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oldRatePosition = 0;
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newRatePosition = 0;
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prevPeriod = 0;
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speed = 1.0f;
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pitch = 1.0f;
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}
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/**
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* Sets the output speed.
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*/
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public void setSpeed(float speed) {
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this.speed = speed;
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}
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/**
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* Gets the output speed.
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*/
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public float getSpeed() {
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return speed;
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}
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/**
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* Sets the output pitch.
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*/
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public void setPitch(float pitch) {
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this.pitch = pitch;
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}
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/**
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* Gets the output pitch.
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*/
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public float getPitch() {
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return pitch;
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this.rate = (float) inputSampleRateHz / outputSampleRateHz;
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}
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/**
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@ -148,8 +126,9 @@ import java.util.Arrays;
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public void queueEndOfStream() {
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int remainingSamples = numInputSamples;
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float s = speed / pitch;
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float r = rate * pitch;
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int expectedOutputSamples =
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numOutputSamples + (int) ((remainingSamples / s + numPitchSamples) / pitch + 0.5f);
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numOutputSamples + (int) ((remainingSamples / s + numPitchSamples) / r + 0.5f);
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// Add enough silence to flush both input and pitch buffers.
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enlargeInputBufferIfNeeded(remainingSamples + 2 * maxRequired);
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@ -292,7 +271,7 @@ import java.util.Arrays;
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// sampling.
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int period;
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int retPeriod;
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int skip = sampleRate > AMDF_FREQUENCY ? sampleRate / AMDF_FREQUENCY : 1;
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int skip = inputSampleRateHz > AMDF_FREQUENCY ? inputSampleRateHz / AMDF_FREQUENCY : 1;
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if (numChannels == 1 && skip == 1) {
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period = findPitchPeriodInRange(samples, position, minPeriod, maxPeriod);
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} else {
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@ -388,8 +367,8 @@ import java.util.Arrays;
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if (numOutputSamples == originalNumOutputSamples) {
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return;
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}
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int newSampleRate = (int) (sampleRate / rate);
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int oldSampleRate = sampleRate;
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int newSampleRate = (int) (inputSampleRateHz / rate);
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int oldSampleRate = inputSampleRateHz;
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// Set these values to help with the integer math.
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while (newSampleRate > (1 << 14) || oldSampleRate > (1 << 14)) {
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newSampleRate /= 2;
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@ -476,6 +455,7 @@ import java.util.Arrays;
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// Resample as many pitch periods as we have buffered on the input.
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int originalNumOutputSamples = numOutputSamples;
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float s = speed / pitch;
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float r = rate * pitch;
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if (s > 1.00001 || s < 0.99999) {
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changeSpeed(s);
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} else {
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@ -486,8 +466,8 @@ import java.util.Arrays;
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if (pitch != 1.0f) {
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adjustPitch(originalNumOutputSamples);
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}
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} else if (!USE_CHORD_PITCH && pitch != 1.0f) {
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adjustRate(pitch, originalNumOutputSamples);
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} else if (r != 1.0f) {
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adjustRate(r, originalNumOutputSamples);
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}
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}
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@ -24,7 +24,7 @@ import java.nio.ByteOrder;
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import java.nio.ShortBuffer;
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/**
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* An {@link AudioProcessor} that uses the Sonic library to modify the speed/pitch of audio.
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* An {@link AudioProcessor} that uses the Sonic library to modify audio speed/pitch/sample rate.
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*/
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public final class SonicAudioProcessor implements AudioProcessor {
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@ -44,18 +44,30 @@ public final class SonicAudioProcessor implements AudioProcessor {
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* The minimum allowed pitch in {@link #setPitch(float)}.
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*/
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public static final float MINIMUM_PITCH = 0.1f;
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/**
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* Indicates that the output sample rate should be the same as the input.
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*/
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public static final int SAMPLE_RATE_NO_CHANGE = -1;
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/**
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* The threshold below which the difference between two pitch/speed factors is negligible.
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*/
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private static final float CLOSE_THRESHOLD = 0.01f;
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/**
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* The minimum number of output bytes at which the speedup is calculated using the input/output
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* byte counts, rather than using the current playback parameters speed.
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*/
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private static final int MIN_BYTES_FOR_SPEEDUP_CALCULATION = 1024;
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private int pendingOutputSampleRateHz;
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private int channelCount;
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private int sampleRateHz;
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private Sonic sonic;
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private float speed;
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private float pitch;
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private int outputSampleRateHz;
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private ByteBuffer buffer;
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private ShortBuffer shortBuffer;
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@ -72,9 +84,11 @@ public final class SonicAudioProcessor implements AudioProcessor {
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pitch = 1f;
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channelCount = Format.NO_VALUE;
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sampleRateHz = Format.NO_VALUE;
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outputSampleRateHz = Format.NO_VALUE;
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buffer = EMPTY_BUFFER;
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||||
shortBuffer = buffer.asShortBuffer();
|
||||
outputBuffer = EMPTY_BUFFER;
|
||||
pendingOutputSampleRateHz = SAMPLE_RATE_NO_CHANGE;
|
||||
}
|
||||
|
||||
/**
|
||||
|
|
@ -100,17 +114,34 @@ public final class SonicAudioProcessor implements AudioProcessor {
|
|||
}
|
||||
|
||||
/**
|
||||
* Returns the number of bytes of input queued since the last call to {@link #flush()}.
|
||||
* Sets the sample rate for output audio, in hertz. Pass {@link #SAMPLE_RATE_NO_CHANGE} to output
|
||||
* audio at the same sample rate as the input. After calling this method, call
|
||||
* {@link #configure(int, int, int)} to start using the new sample rate.
|
||||
*
|
||||
* @param sampleRateHz The sample rate for output audio, in hertz.
|
||||
* @see #configure(int, int, int)
|
||||
*/
|
||||
public long getInputByteCount() {
|
||||
return inputBytes;
|
||||
public void setOutputSampleRateHz(int sampleRateHz) {
|
||||
pendingOutputSampleRateHz = sampleRateHz;
|
||||
}
|
||||
|
||||
/**
|
||||
* Returns the number of bytes of output dequeued since the last call to {@link #flush()}.
|
||||
* Returns the specified duration scaled to take into account the speedup factor of this instance,
|
||||
* in the same units as {@code duration}.
|
||||
*
|
||||
* @param duration The duration to scale taking into account speedup.
|
||||
* @return The specified duration scaled to take into account speedup, in the same units as
|
||||
* {@code duration}.
|
||||
*/
|
||||
public long getOutputByteCount() {
|
||||
return outputBytes;
|
||||
public long scaleDurationForSpeedup(long duration) {
|
||||
if (outputBytes >= MIN_BYTES_FOR_SPEEDUP_CALCULATION) {
|
||||
return outputSampleRateHz == sampleRateHz
|
||||
? Util.scaleLargeTimestamp(duration, inputBytes, outputBytes)
|
||||
: Util.scaleLargeTimestamp(duration, inputBytes * outputSampleRateHz,
|
||||
outputBytes * sampleRateHz);
|
||||
} else {
|
||||
return (long) ((double) speed * duration);
|
||||
}
|
||||
}
|
||||
|
||||
@Override
|
||||
|
|
@ -119,17 +150,22 @@ public final class SonicAudioProcessor implements AudioProcessor {
|
|||
if (encoding != C.ENCODING_PCM_16BIT) {
|
||||
throw new UnhandledFormatException(sampleRateHz, channelCount, encoding);
|
||||
}
|
||||
if (this.sampleRateHz == sampleRateHz && this.channelCount == channelCount) {
|
||||
int outputSampleRateHz = pendingOutputSampleRateHz == SAMPLE_RATE_NO_CHANGE
|
||||
? sampleRateHz : pendingOutputSampleRateHz;
|
||||
if (this.sampleRateHz == sampleRateHz && this.channelCount == channelCount
|
||||
&& this.outputSampleRateHz == outputSampleRateHz) {
|
||||
return false;
|
||||
}
|
||||
this.sampleRateHz = sampleRateHz;
|
||||
this.channelCount = channelCount;
|
||||
this.outputSampleRateHz = outputSampleRateHz;
|
||||
return true;
|
||||
}
|
||||
|
||||
@Override
|
||||
public boolean isActive() {
|
||||
return Math.abs(speed - 1f) >= CLOSE_THRESHOLD || Math.abs(pitch - 1f) >= CLOSE_THRESHOLD;
|
||||
return Math.abs(speed - 1f) >= CLOSE_THRESHOLD || Math.abs(pitch - 1f) >= CLOSE_THRESHOLD
|
||||
|| outputSampleRateHz != sampleRateHz;
|
||||
}
|
||||
|
||||
@Override
|
||||
|
|
@ -142,6 +178,11 @@ public final class SonicAudioProcessor implements AudioProcessor {
|
|||
return C.ENCODING_PCM_16BIT;
|
||||
}
|
||||
|
||||
@Override
|
||||
public int getOutputSampleRateHz() {
|
||||
return outputSampleRateHz;
|
||||
}
|
||||
|
||||
@Override
|
||||
public void queueInput(ByteBuffer inputBuffer) {
|
||||
if (inputBuffer.hasRemaining()) {
|
||||
|
|
@ -187,9 +228,7 @@ public final class SonicAudioProcessor implements AudioProcessor {
|
|||
|
||||
@Override
|
||||
public void flush() {
|
||||
sonic = new Sonic(sampleRateHz, channelCount);
|
||||
sonic.setSpeed(speed);
|
||||
sonic.setPitch(pitch);
|
||||
sonic = new Sonic(sampleRateHz, channelCount, speed, pitch, outputSampleRateHz);
|
||||
outputBuffer = EMPTY_BUFFER;
|
||||
inputBytes = 0;
|
||||
outputBytes = 0;
|
||||
|
|
@ -204,9 +243,11 @@ public final class SonicAudioProcessor implements AudioProcessor {
|
|||
outputBuffer = EMPTY_BUFFER;
|
||||
channelCount = Format.NO_VALUE;
|
||||
sampleRateHz = Format.NO_VALUE;
|
||||
outputSampleRateHz = Format.NO_VALUE;
|
||||
inputBytes = 0;
|
||||
outputBytes = 0;
|
||||
inputEnded = false;
|
||||
pendingOutputSampleRateHz = SAMPLE_RATE_NO_CHANGE;
|
||||
}
|
||||
|
||||
}
|
||||
|
|
|
|||
|
|
@ -31,6 +31,7 @@ import java.nio.ByteOrder;
|
|||
private int trimStartSamples;
|
||||
private int trimEndSamples;
|
||||
private int channelCount;
|
||||
private int sampleRateHz;
|
||||
|
||||
private int pendingTrimStartBytes;
|
||||
private ByteBuffer buffer;
|
||||
|
|
@ -69,6 +70,7 @@ import java.nio.ByteOrder;
|
|||
throw new UnhandledFormatException(sampleRateHz, channelCount, encoding);
|
||||
}
|
||||
this.channelCount = channelCount;
|
||||
this.sampleRateHz = sampleRateHz;
|
||||
endBuffer = new byte[trimEndSamples * channelCount * 2];
|
||||
endBufferSize = 0;
|
||||
pendingTrimStartBytes = trimStartSamples * channelCount * 2;
|
||||
|
|
@ -92,6 +94,11 @@ import java.nio.ByteOrder;
|
|||
return C.ENCODING_PCM_16BIT;
|
||||
}
|
||||
|
||||
@Override
|
||||
public int getOutputSampleRateHz() {
|
||||
return sampleRateHz;
|
||||
}
|
||||
|
||||
@Override
|
||||
public void queueInput(ByteBuffer inputBuffer) {
|
||||
int position = inputBuffer.position();
|
||||
|
|
@ -174,6 +181,7 @@ import java.nio.ByteOrder;
|
|||
flush();
|
||||
buffer = EMPTY_BUFFER;
|
||||
channelCount = Format.NO_VALUE;
|
||||
sampleRateHz = Format.NO_VALUE;
|
||||
endBuffer = null;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -0,0 +1,137 @@
|
|||
/*
|
||||
* Copyright (C) 2017 The Android Open Source Project
|
||||
*
|
||||
* Licensed under the Apache License, Version 2.0 (the "License");
|
||||
* you may not use this file except in compliance with the License.
|
||||
* You may obtain a copy of the License at
|
||||
*
|
||||
* http://www.apache.org/licenses/LICENSE-2.0
|
||||
*
|
||||
* Unless required by applicable law or agreed to in writing, software
|
||||
* distributed under the License is distributed on an "AS IS" BASIS,
|
||||
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
|
||||
* See the License for the specific language governing permissions and
|
||||
* limitations under the License.
|
||||
*/
|
||||
package com.google.android.exoplayer2.audio;
|
||||
|
||||
import static com.google.common.truth.Truth.assertThat;
|
||||
import static org.junit.Assert.fail;
|
||||
|
||||
import com.google.android.exoplayer2.C;
|
||||
import org.junit.Before;
|
||||
import org.junit.Test;
|
||||
import org.junit.runner.RunWith;
|
||||
import org.robolectric.RobolectricTestRunner;
|
||||
import org.robolectric.annotation.Config;
|
||||
|
||||
/**
|
||||
* Unit test for {@link SonicAudioProcessor}.
|
||||
*/
|
||||
@RunWith(RobolectricTestRunner.class)
|
||||
@Config(sdk = Config.TARGET_SDK, manifest = Config.NONE)
|
||||
public final class SonicAudioProcessorTest {
|
||||
|
||||
private SonicAudioProcessor sonicAudioProcessor;
|
||||
|
||||
@Before
|
||||
public void setUp() {
|
||||
sonicAudioProcessor = new SonicAudioProcessor();
|
||||
}
|
||||
|
||||
@Test
|
||||
public void testReconfigureWithSameSampleRate() throws Exception {
|
||||
// When configured for resampling from 44.1 kHz to 48 kHz, the output sample rate is correct.
|
||||
sonicAudioProcessor.setOutputSampleRateHz(48000);
|
||||
sonicAudioProcessor.configure(44100, 2, C.ENCODING_PCM_16BIT);
|
||||
assertThat(sonicAudioProcessor.getOutputSampleRateHz()).isEqualTo(48000);
|
||||
assertThat(sonicAudioProcessor.isActive()).isTrue();
|
||||
// When reconfigured with 48 kHz input, there is no resampling.
|
||||
sonicAudioProcessor.configure(48000, 2, C.ENCODING_PCM_16BIT);
|
||||
assertThat(sonicAudioProcessor.getOutputSampleRateHz()).isEqualTo(48000);
|
||||
assertThat(sonicAudioProcessor.isActive()).isFalse();
|
||||
// When reconfigure with 44.1 kHz input, resampling is enabled again.
|
||||
sonicAudioProcessor.configure(44100, 2, C.ENCODING_PCM_16BIT);
|
||||
assertThat(sonicAudioProcessor.getOutputSampleRateHz()).isEqualTo(48000);
|
||||
assertThat(sonicAudioProcessor.isActive()).isTrue();
|
||||
}
|
||||
|
||||
@Test
|
||||
public void testNoSampleRateChange() throws Exception {
|
||||
// Configure for resampling 44.1 kHz to 48 kHz.
|
||||
sonicAudioProcessor.setOutputSampleRateHz(48000);
|
||||
sonicAudioProcessor.configure(44100, 2, C.ENCODING_PCM_16BIT);
|
||||
// Reconfigure to not modify the sample rate.
|
||||
sonicAudioProcessor.setOutputSampleRateHz(SonicAudioProcessor.SAMPLE_RATE_NO_CHANGE);
|
||||
sonicAudioProcessor.configure(22050, 2, C.ENCODING_PCM_16BIT);
|
||||
// The sample rate is unmodified, and the audio processor is not active.
|
||||
assertThat(sonicAudioProcessor.getOutputSampleRateHz()).isEqualTo(22050);
|
||||
assertThat(sonicAudioProcessor.isActive()).isFalse();
|
||||
}
|
||||
|
||||
@Test
|
||||
public void testBecomesActiveAfterConfigure() throws Exception {
|
||||
sonicAudioProcessor.configure(44100, 2, C.ENCODING_PCM_16BIT);
|
||||
// Set a new sample rate.
|
||||
sonicAudioProcessor.setOutputSampleRateHz(22050);
|
||||
// The new sample rate is not active yet.
|
||||
assertThat(sonicAudioProcessor.isActive()).isFalse();
|
||||
assertThat(sonicAudioProcessor.getOutputSampleRateHz()).isEqualTo(44100);
|
||||
}
|
||||
|
||||
@Test
|
||||
public void testSampleRateChangeBecomesActiveAfterConfigure() throws Exception {
|
||||
// Configure for resampling 44.1 kHz to 48 kHz.
|
||||
sonicAudioProcessor.setOutputSampleRateHz(48000);
|
||||
sonicAudioProcessor.configure(44100, 2, C.ENCODING_PCM_16BIT);
|
||||
// Set a new sample rate, which isn't active yet.
|
||||
sonicAudioProcessor.setOutputSampleRateHz(22050);
|
||||
assertThat(sonicAudioProcessor.getOutputSampleRateHz()).isEqualTo(48000);
|
||||
// The new sample rate takes effect on reconfiguration.
|
||||
sonicAudioProcessor.configure(44100, 2, C.ENCODING_PCM_16BIT);
|
||||
assertThat(sonicAudioProcessor.getOutputSampleRateHz()).isEqualTo(22050);
|
||||
}
|
||||
|
||||
@Test
|
||||
public void testIsActiveWithSpeedChange() throws Exception {
|
||||
sonicAudioProcessor.setSpeed(1.5f);
|
||||
sonicAudioProcessor.configure(44100, 2, C.ENCODING_PCM_16BIT);
|
||||
assertThat(sonicAudioProcessor.isActive()).isTrue();
|
||||
}
|
||||
|
||||
@Test
|
||||
public void testIsActiveWithPitchChange() throws Exception {
|
||||
sonicAudioProcessor.setPitch(1.5f);
|
||||
sonicAudioProcessor.configure(44100, 2, C.ENCODING_PCM_16BIT);
|
||||
assertThat(sonicAudioProcessor.isActive()).isTrue();
|
||||
}
|
||||
|
||||
@Test
|
||||
public void testIsNotActiveWithNoChange() throws Exception {
|
||||
sonicAudioProcessor.configure(44100, 2, C.ENCODING_PCM_16BIT);
|
||||
assertThat(sonicAudioProcessor.isActive()).isFalse();
|
||||
}
|
||||
|
||||
@Test
|
||||
public void testDoesNotSupportNon16BitInput() throws Exception {
|
||||
try {
|
||||
sonicAudioProcessor.configure(44100, 2, C.ENCODING_PCM_8BIT);
|
||||
fail();
|
||||
} catch (AudioProcessor.UnhandledFormatException e) {
|
||||
// Expected.
|
||||
}
|
||||
try {
|
||||
sonicAudioProcessor.configure(44100, 2, C.ENCODING_PCM_24BIT);
|
||||
fail();
|
||||
} catch (AudioProcessor.UnhandledFormatException e) {
|
||||
// Expected.
|
||||
}
|
||||
try {
|
||||
sonicAudioProcessor.configure(44100, 2, C.ENCODING_PCM_32BIT);
|
||||
fail();
|
||||
} catch (AudioProcessor.UnhandledFormatException e) {
|
||||
// Expected.
|
||||
}
|
||||
}
|
||||
|
||||
}
|
||||
Loading…
Reference in a new issue