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

extensions/gvr/src/main/java/com/google/android/exoplayer2/ext/gvr/GvrAudioProcessor.java

@@ -139,6 +139,11 @@ public final class GvrAudioProcessor implements AudioProcessor {
   }
   }
 
 
   @Override
   @Override
+  public int getOutputSampleRateHz() {
+    return sampleRateHz;
+  }
+
+  @Override
   public void queueInput(ByteBuffer input) {
   public void queueInput(ByteBuffer input) {
     int position = input.position();
     int position = input.position();
     int readBytes = gvrAudioSurround.addInput(input, position, input.limit() - position);
     int readBytes = gvrAudioSurround.addInput(input, position, input.limit() - position);

library/core/src/main/java/com/google/android/exoplayer2/audio/AudioProcessor.java

@@ -20,7 +20,15 @@ import java.nio.ByteBuffer;
 import java.nio.ByteOrder;
 import java.nio.ByteOrder;
 
 
 /**
 /**
- * Interface for audio processors.
+ * Interface for audio processors, which take audio data as input and transform it, potentially
+ * modifying its channel count, encoding and/or sample rate.
+ * <p>
+ * Call {@link #configure(int, int, int)} to configure the processor to receive input audio, then
+ * call {@link #isActive()} to determine whether the processor is active.
+ * {@link #queueInput(ByteBuffer)}, {@link #queueEndOfStream()}, {@link #getOutput()},
+ * {@link #isEnded()}, {@link #getOutputChannelCount()}, {@link #getOutputEncoding()} and
+ * {@link #getOutputSampleRateHz()} may only be called if the processor is active. Call
+ * {@link #reset()} to reset the processor to its unconfigured state.
  */
  */
 public interface AudioProcessor {
 public interface AudioProcessor {
 
 
@@ -46,8 +54,9 @@ public interface AudioProcessor {
    * method, {@link #isActive()} returns whether the processor needs to handle buffers; if not, the
    * method, {@link #isActive()} returns whether the processor needs to handle buffers; if not, the
    * processor will not accept any buffers until it is reconfigured. Returns {@code true} if the
    * processor will not accept any buffers until it is reconfigured. Returns {@code true} if the
    * processor must be flushed, or if the value returned by {@link #isActive()} has changed as a
    * processor must be flushed, or if the value returned by {@link #isActive()} has changed as a
-   * result of the call. If it's active, {@link #getOutputChannelCount()} and
-   * {@link #getOutputEncoding()} return the processor's output format.
+   * result of the call. If it's active, {@link #getOutputSampleRateHz()},
+   * {@link #getOutputChannelCount()} and {@link #getOutputEncoding()} return the processor's output
+   * format.
    *
    *
    * @param sampleRateHz The sample rate of input audio in Hz.
    * @param sampleRateHz The sample rate of input audio in Hz.
    * @param channelCount The number of interleaved channels in input audio.
    * @param channelCount The number of interleaved channels in input audio.
@@ -65,17 +74,28 @@ public interface AudioProcessor {
   boolean isActive();
   boolean isActive();
 
 
   /**
   /**
-   * Returns the number of audio channels in the data output by the processor.
+   * Returns the number of audio channels in the data output by the processor. The value may change
+   * as a result of calling {@link #configure(int, int, int)} and is undefined if the instance is
+   * not active.
    */
    */
   int getOutputChannelCount();
   int getOutputChannelCount();
 
 
   /**
   /**
-   * Returns the audio encoding used in the data output by the processor.
+   * Returns the audio encoding used in the data output by the processor. The value may change as a
+   * result of calling {@link #configure(int, int, int)} and is undefined if the instance is not
+   * active.
    */
    */
   @C.Encoding
   @C.Encoding
   int getOutputEncoding();
   int getOutputEncoding();
 
 
   /**
   /**
+   * Returns the sample rate of audio output by the processor, in hertz. The value may change as a
+   * result of calling {@link #configure(int, int, int)} and is undefined if the instance is not
+   * active.
+   */
+  int getOutputSampleRateHz();
+
+  /**
    * Queues audio data between the position and limit of the input {@code buffer} for processing.
    * Queues audio data between the position and limit of the input {@code buffer} for processing.
    * {@code buffer} must be a direct byte buffer with native byte order. Its contents are treated as
    * {@code buffer} must be a direct byte buffer with native byte order. Its contents are treated as
    * read-only. Its position will be advanced by the number of bytes consumed (which may be zero).
    * read-only. Its position will be advanced by the number of bytes consumed (which may be zero).
@@ -116,7 +136,7 @@ public interface AudioProcessor {
   void flush();
   void flush();
 
 
   /**
   /**
-   * Resets the processor to its initial state.
+   * Resets the processor to its unconfigured state.
    */
    */
   void reset();
   void reset();
 
 

library/core/src/main/java/com/google/android/exoplayer2/audio/AudioSink.java

@@ -186,12 +186,12 @@ public interface AudioSink {
   /**
   /**
    * Configures (or reconfigures) the sink.
    * Configures (or reconfigures) the sink.
    *
    *
-   * @param mimeType The MIME type of audio data provided in the input buffers.
-   * @param channelCount The number of channels.
-   * @param sampleRate The sample rate in Hz.
-   * @param pcmEncoding For PCM formats, the encoding used. One of {@link C#ENCODING_PCM_16BIT},
-   *     {@link C#ENCODING_PCM_16BIT}, {@link C#ENCODING_PCM_24BIT} and
-   *     {@link C#ENCODING_PCM_32BIT}.
+   * @param inputMimeType The MIME type of audio data provided in the input buffers.
+   * @param inputChannelCount The number of channels.
+   * @param inputSampleRate The sample rate in Hz.
+   * @param inputPcmEncoding For PCM formats, the encoding used. One of
+   *     {@link C#ENCODING_PCM_16BIT}, {@link C#ENCODING_PCM_16BIT}, {@link C#ENCODING_PCM_24BIT}
+   *     and {@link C#ENCODING_PCM_32BIT}.
    * @param specifiedBufferSize A specific size for the playback buffer in bytes, or 0 to infer a
    * @param specifiedBufferSize A specific size for the playback buffer in bytes, or 0 to infer a
    *     suitable buffer size.
    *     suitable buffer size.
    * @param outputChannels A mapping from input to output channels that is applied to this sink's
    * @param outputChannels A mapping from input to output channels that is applied to this sink's
@@ -206,9 +206,9 @@ public interface AudioSink {
    *     {@link #configure(String, int, int, int, int, int[], int, int)}.
    *     {@link #configure(String, int, int, int, int, int[], int, int)}.
    * @throws ConfigurationException If an error occurs configuring the sink.
    * @throws ConfigurationException If an error occurs configuring the sink.
    */
    */
-  void configure(String mimeType, int channelCount, int sampleRate, @C.PcmEncoding int pcmEncoding,
-      int specifiedBufferSize, @Nullable int[] outputChannels, int trimStartSamples,
-      int trimEndSamples) throws ConfigurationException;
+  void configure(String inputMimeType, int inputChannelCount, int inputSampleRate,
+      @C.PcmEncoding int inputPcmEncoding, int specifiedBufferSize, @Nullable int[] outputChannels,
+      int trimStartSamples, int trimEndSamples) throws ConfigurationException;
 
 
   /**
   /**
    * Starts or resumes consuming audio if initialized.
    * Starts or resumes consuming audio if initialized.

library/core/src/main/java/com/google/android/exoplayer2/audio/ChannelMappingAudioProcessor.java

@@ -104,6 +104,11 @@ import java.util.Arrays;
   }
   }
 
 
   @Override
   @Override
+  public int getOutputSampleRateHz() {
+    return sampleRateHz;
+  }
+
+  @Override
   public void queueInput(ByteBuffer inputBuffer) {
   public void queueInput(ByteBuffer inputBuffer) {
     int position = inputBuffer.position();
     int position = inputBuffer.position();
     int limit = inputBuffer.limit();
     int limit = inputBuffer.limit();

library/core/src/main/java/com/google/android/exoplayer2/audio/DefaultAudioSink.java

@@ -149,13 +149,6 @@ public final class DefaultAudioSink implements AudioSink {
   private static final int MIN_TIMESTAMP_SAMPLE_INTERVAL_US = 500000;
   private static final int MIN_TIMESTAMP_SAMPLE_INTERVAL_US = 500000;
 
 
   /**
   /**
-   * The minimum number of output bytes from {@link #sonicAudioProcessor} at which the speedup is
-   * calculated using the input/output byte counts from the processor, rather than using the
-   * current playback parameters speed.
-   */
-  private static final int SONIC_MIN_BYTES_FOR_SPEEDUP = 1024;
-
-  /**
    * Whether to enable a workaround for an issue where an audio effect does not keep its session
    * Whether to enable a workaround for an issue where an audio effect does not keep its session
    * active across releasing/initializing a new audio track, on platform builds where
    * active across releasing/initializing a new audio track, on platform builds where
    * {@link Util#SDK_INT} < 21.
    * {@link Util#SDK_INT} < 21.
@@ -189,6 +182,7 @@ public final class DefaultAudioSink implements AudioSink {
    */
    */
   private AudioTrack keepSessionIdAudioTrack;
   private AudioTrack keepSessionIdAudioTrack;
   private AudioTrack audioTrack;
   private AudioTrack audioTrack;
+  private int inputSampleRate;
   private int sampleRate;
   private int sampleRate;
   private int channelConfig;
   private int channelConfig;
   private @C.Encoding int encoding;
   private @C.Encoding int encoding;
@@ -337,14 +331,18 @@ public final class DefaultAudioSink implements AudioSink {
   }
   }
 
 
   @Override
   @Override
-  public void configure(String mimeType, int channelCount, int sampleRate,
-      @C.PcmEncoding int pcmEncoding, int specifiedBufferSize, @Nullable int[] outputChannels,
+  public void configure(String inputMimeType, int inputChannelCount, int inputSampleRate,
+      @C.PcmEncoding int inputPcmEncoding, int specifiedBufferSize, @Nullable int[] outputChannels,
       int trimStartSamples, int trimEndSamples) throws ConfigurationException {
       int trimStartSamples, int trimEndSamples) throws ConfigurationException {
-    boolean passthrough = !MimeTypes.AUDIO_RAW.equals(mimeType);
-    @C.Encoding int encoding = passthrough ? getEncodingForMimeType(mimeType) : pcmEncoding;
+    this.inputSampleRate = inputSampleRate;
+    int channelCount = inputChannelCount;
+    int sampleRate = inputSampleRate;
+    @C.Encoding int encoding;
+    boolean passthrough = !MimeTypes.AUDIO_RAW.equals(inputMimeType);
     boolean flush = false;
     boolean flush = false;
     if (!passthrough) {
     if (!passthrough) {
-      pcmFrameSize = Util.getPcmFrameSize(pcmEncoding, channelCount);
+      encoding = inputPcmEncoding;
+      pcmFrameSize = Util.getPcmFrameSize(inputPcmEncoding, channelCount);
       trimmingAudioProcessor.setTrimSampleCount(trimStartSamples, trimEndSamples);
       trimmingAudioProcessor.setTrimSampleCount(trimStartSamples, trimEndSamples);
       channelMappingAudioProcessor.setChannelMap(outputChannels);
       channelMappingAudioProcessor.setChannelMap(outputChannels);
       for (AudioProcessor audioProcessor : availableAudioProcessors) {
       for (AudioProcessor audioProcessor : availableAudioProcessors) {
@@ -355,12 +353,15 @@ public final class DefaultAudioSink implements AudioSink {
         }
         }
         if (audioProcessor.isActive()) {
         if (audioProcessor.isActive()) {
           channelCount = audioProcessor.getOutputChannelCount();
           channelCount = audioProcessor.getOutputChannelCount();
+          sampleRate = audioProcessor.getOutputSampleRateHz();
           encoding = audioProcessor.getOutputEncoding();
           encoding = audioProcessor.getOutputEncoding();
         }
         }
       }
       }
       if (flush) {
       if (flush) {
         resetAudioProcessors();
         resetAudioProcessors();
       }
       }
+    } else {
+      encoding = getEncodingForMimeType(inputMimeType);
     }
     }
 
 
     int channelConfig;
     int channelConfig;
@@ -598,8 +599,8 @@ public final class DefaultAudioSink implements AudioSink {
         startMediaTimeState = START_IN_SYNC;
         startMediaTimeState = START_IN_SYNC;
       } else {
       } else {
         // Sanity check that presentationTimeUs is consistent with the expected value.
         // Sanity check that presentationTimeUs is consistent with the expected value.
-        long expectedPresentationTimeUs = startMediaTimeUs
-            + framesToDurationUs(getSubmittedFrames());
+        long expectedPresentationTimeUs =
+            startMediaTimeUs + inputFramesToDurationUs(getSubmittedFrames());
         if (startMediaTimeState == START_IN_SYNC
         if (startMediaTimeState == START_IN_SYNC
             && Math.abs(expectedPresentationTimeUs - presentationTimeUs) > 200000) {
             && Math.abs(expectedPresentationTimeUs - presentationTimeUs) > 200000) {
           Log.e(TAG, "Discontinuity detected [expected " + expectedPresentationTimeUs + ", got "
           Log.e(TAG, "Discontinuity detected [expected " + expectedPresentationTimeUs + ", got "
@@ -997,15 +998,11 @@ public final class DefaultAudioSink implements AudioSink {
       return positionUs + playbackParametersOffsetUs - playbackParametersPositionUs;
       return positionUs + playbackParametersOffsetUs - playbackParametersPositionUs;
     }
     }
 
 
-    if (playbackParametersCheckpoints.isEmpty()
-        && sonicAudioProcessor.getOutputByteCount() >= SONIC_MIN_BYTES_FOR_SPEEDUP) {
+    if (playbackParametersCheckpoints.isEmpty()) {
       return playbackParametersOffsetUs
       return playbackParametersOffsetUs
-          + Util.scaleLargeTimestamp(positionUs - playbackParametersPositionUs,
-          sonicAudioProcessor.getInputByteCount(), sonicAudioProcessor.getOutputByteCount());
+          + sonicAudioProcessor.scaleDurationForSpeedup(positionUs - playbackParametersPositionUs);
     }
     }
-
-    // We are playing drained data at a previous playback speed, or don't have enough bytes to
-    // calculate an accurate speedup, so fall back to multiplying by the speed.
+    // We are playing data at a previous playback speed, so fall back to multiplying by the speed.
     return playbackParametersOffsetUs
     return playbackParametersOffsetUs
         + (long) ((double) playbackParameters.speed * (positionUs - playbackParametersPositionUs));
         + (long) ((double) playbackParameters.speed * (positionUs - playbackParametersPositionUs));
   }
   }
@@ -1098,6 +1095,10 @@ public final class DefaultAudioSink implements AudioSink {
     return audioTrack != null;
     return audioTrack != null;
   }
   }
 
 
+  private long inputFramesToDurationUs(long frameCount) {
+    return (frameCount * C.MICROS_PER_SECOND) / inputSampleRate;
+  }
+
   private long framesToDurationUs(long frameCount) {
   private long framesToDurationUs(long frameCount) {
     return (frameCount * C.MICROS_PER_SECOND) / sampleRate;
     return (frameCount * C.MICROS_PER_SECOND) / sampleRate;
   }
   }

library/core/src/main/java/com/google/android/exoplayer2/audio/ResamplingAudioProcessor.java

@@ -80,6 +80,11 @@ import java.nio.ByteOrder;
   }
   }
 
 
   @Override
   @Override
+  public int getOutputSampleRateHz() {
+    return sampleRateHz;
+  }
+
+  @Override
   public void queueInput(ByteBuffer inputBuffer) {
   public void queueInput(ByteBuffer inputBuffer) {
     // Prepare the output buffer.
     // Prepare the output buffer.
     int position = inputBuffer.position();
     int position = inputBuffer.position();

library/core/src/main/java/com/google/android/exoplayer2/audio/Sonic.java

@@ -32,8 +32,11 @@ import java.util.Arrays;
   private static final int MAXIMUM_PITCH = 400;
   private static final int MAXIMUM_PITCH = 400;
   private static final int AMDF_FREQUENCY = 4000;
   private static final int AMDF_FREQUENCY = 4000;
 
 
-  private final int sampleRate;
+  private final int inputSampleRateHz;
   private final int numChannels;
   private final int numChannels;
+  private final float speed;
+  private final float pitch;
+  private final float rate;
   private final int minPeriod;
   private final int minPeriod;
   private final int maxPeriod;
   private final int maxPeriod;
   private final int maxRequired;
   private final int maxRequired;
@@ -47,8 +50,6 @@ import java.util.Arrays;
   private short[] pitchBuffer;
   private short[] pitchBuffer;
   private int oldRatePosition;
   private int oldRatePosition;
   private int newRatePosition;
   private int newRatePosition;
-  private float speed;
-  private float pitch;
   private int numInputSamples;
   private int numInputSamples;
   private int numOutputSamples;
   private int numOutputSamples;
   private int numPitchSamples;
   private int numPitchSamples;
@@ -61,14 +62,18 @@ import java.util.Arrays;
   /**
   /**
    * Creates a new Sonic audio stream processor.
    * Creates a new Sonic audio stream processor.
    *
    *
-   * @param sampleRate The sample rate of input audio.
+   * @param inputSampleRateHz The sample rate of input audio, in hertz.
    * @param numChannels The number of channels in the input audio.
    * @param numChannels The number of channels in the input audio.
+   * @param speed The speedup factor for output audio.
+   * @param pitch The pitch factor for output audio.
+   * @param outputSampleRateHz The sample rate for output audio, in hertz.
    */
    */
-  public Sonic(int sampleRate, int numChannels) {
-    this.sampleRate = sampleRate;
+  public Sonic(int inputSampleRateHz, int numChannels, float speed, float pitch,
+      int outputSampleRateHz) {
+    this.inputSampleRateHz = inputSampleRateHz;
     this.numChannels = numChannels;
     this.numChannels = numChannels;
-    minPeriod = sampleRate / MAXIMUM_PITCH;
-    maxPeriod = sampleRate / MINIMUM_PITCH;
+    minPeriod = inputSampleRateHz / MAXIMUM_PITCH;
+    maxPeriod = inputSampleRateHz / MINIMUM_PITCH;
     maxRequired = 2 * maxPeriod;
     maxRequired = 2 * maxPeriod;
     downSampleBuffer = new short[maxRequired];
     downSampleBuffer = new short[maxRequired];
     inputBufferSize = maxRequired;
     inputBufferSize = maxRequired;
@@ -80,36 +85,9 @@ import java.util.Arrays;
     oldRatePosition = 0;
     oldRatePosition = 0;
     newRatePosition = 0;
     newRatePosition = 0;
     prevPeriod = 0;
     prevPeriod = 0;
-    speed = 1.0f;
-    pitch = 1.0f;
-  }
-
-  /**
-   * Sets the output speed.
-   */
-  public void setSpeed(float speed) {
     this.speed = speed;
     this.speed = speed;
-  }
-
-  /**
-   * Gets the output speed.
-   */
-  public float getSpeed() {
-    return speed;
-  }
-
-  /**
-   * Sets the output pitch.
-   */
-  public void setPitch(float pitch) {
     this.pitch = pitch;
     this.pitch = pitch;
-  }
-
-  /**
-   * Gets the output pitch.
-   */
-  public float getPitch() {
-    return pitch;
+    this.rate = (float) inputSampleRateHz / outputSampleRateHz;
   }
   }
 
 
   /**
   /**
@@ -148,8 +126,9 @@ import java.util.Arrays;
   public void queueEndOfStream() {
   public void queueEndOfStream() {
     int remainingSamples = numInputSamples;
     int remainingSamples = numInputSamples;
     float s = speed / pitch;
     float s = speed / pitch;
+    float r = rate * pitch;
     int expectedOutputSamples =
     int expectedOutputSamples =
-        numOutputSamples + (int) ((remainingSamples / s + numPitchSamples) / pitch + 0.5f);
+        numOutputSamples + (int) ((remainingSamples / s + numPitchSamples) / r + 0.5f);
 
 
     // Add enough silence to flush both input and pitch buffers.
     // Add enough silence to flush both input and pitch buffers.
     enlargeInputBufferIfNeeded(remainingSamples + 2 * maxRequired);
     enlargeInputBufferIfNeeded(remainingSamples + 2 * maxRequired);
@@ -292,7 +271,7 @@ import java.util.Arrays;
     // sampling.
     // sampling.
     int period;
     int period;
     int retPeriod;
     int retPeriod;
-    int skip = sampleRate > AMDF_FREQUENCY ? sampleRate / AMDF_FREQUENCY : 1;
+    int skip = inputSampleRateHz > AMDF_FREQUENCY ? inputSampleRateHz / AMDF_FREQUENCY : 1;
     if (numChannels == 1 && skip == 1) {
     if (numChannels == 1 && skip == 1) {
       period = findPitchPeriodInRange(samples, position, minPeriod, maxPeriod);
       period = findPitchPeriodInRange(samples, position, minPeriod, maxPeriod);
     } else {
     } else {
@@ -388,8 +367,8 @@ import java.util.Arrays;
     if (numOutputSamples == originalNumOutputSamples) {
     if (numOutputSamples == originalNumOutputSamples) {
       return;
       return;
     }
     }
-    int newSampleRate = (int) (sampleRate / rate);
-    int oldSampleRate = sampleRate;
+    int newSampleRate = (int) (inputSampleRateHz / rate);
+    int oldSampleRate = inputSampleRateHz;
     // Set these values to help with the integer math.
     // Set these values to help with the integer math.
     while (newSampleRate > (1 << 14) || oldSampleRate > (1 << 14)) {
     while (newSampleRate > (1 << 14) || oldSampleRate > (1 << 14)) {
       newSampleRate /= 2;
       newSampleRate /= 2;
@@ -476,6 +455,7 @@ import java.util.Arrays;
     // Resample as many pitch periods as we have buffered on the input.
     // Resample as many pitch periods as we have buffered on the input.
     int originalNumOutputSamples = numOutputSamples;
     int originalNumOutputSamples = numOutputSamples;
     float s = speed / pitch;
     float s = speed / pitch;
+    float r = rate * pitch;
     if (s > 1.00001 || s < 0.99999) {
     if (s > 1.00001 || s < 0.99999) {
       changeSpeed(s);
       changeSpeed(s);
     } else {
     } else {
@@ -486,8 +466,8 @@ import java.util.Arrays;
       if (pitch != 1.0f) {
       if (pitch != 1.0f) {
         adjustPitch(originalNumOutputSamples);
         adjustPitch(originalNumOutputSamples);
       }
       }
-    } else if (!USE_CHORD_PITCH && pitch != 1.0f) {
-      adjustRate(pitch, originalNumOutputSamples);
+    } else if (r != 1.0f) {
+      adjustRate(r, originalNumOutputSamples);
     }
     }
   }
   }
 
 

library/core/src/main/java/com/google/android/exoplayer2/audio/SonicAudioProcessor.java

@@ -24,7 +24,7 @@ import java.nio.ByteOrder;
 import java.nio.ShortBuffer;
 import java.nio.ShortBuffer;
 
 
 /**
 /**
- * An {@link AudioProcessor} that uses the Sonic library to modify the speed/pitch of audio.
+ * An {@link AudioProcessor} that uses the Sonic library to modify audio speed/pitch/sample rate.
  */
  */
 public final class SonicAudioProcessor implements AudioProcessor {
 public final class SonicAudioProcessor implements AudioProcessor {
 
 
@@ -44,18 +44,30 @@ public final class SonicAudioProcessor implements AudioProcessor {
    * The minimum allowed pitch in {@link #setPitch(float)}.
    * The minimum allowed pitch in {@link #setPitch(float)}.
    */
    */
   public static final float MINIMUM_PITCH = 0.1f;
   public static final float MINIMUM_PITCH = 0.1f;
+  /**
+   * Indicates that the output sample rate should be the same as the input.
+   */
+  public static final int SAMPLE_RATE_NO_CHANGE = -1;
 
 
   /**
   /**
    * The threshold below which the difference between two pitch/speed factors is negligible.
    * The threshold below which the difference between two pitch/speed factors is negligible.
    */
    */
   private static final float CLOSE_THRESHOLD = 0.01f;
   private static final float CLOSE_THRESHOLD = 0.01f;
 
 
+  /**
+   * The minimum number of output bytes at which the speedup is calculated using the input/output
+   * byte counts, rather than using the current playback parameters speed.
+   */
+  private static final int MIN_BYTES_FOR_SPEEDUP_CALCULATION = 1024;
+
+  private int pendingOutputSampleRateHz;
   private int channelCount;
   private int channelCount;
   private int sampleRateHz;
   private int sampleRateHz;
 
 
   private Sonic sonic;
   private Sonic sonic;
   private float speed;
   private float speed;
   private float pitch;
   private float pitch;
+  private int outputSampleRateHz;
 
 
   private ByteBuffer buffer;
   private ByteBuffer buffer;
   private ShortBuffer shortBuffer;
   private ShortBuffer shortBuffer;
@@ -72,9 +84,11 @@ public final class SonicAudioProcessor implements AudioProcessor {
     pitch = 1f;
     pitch = 1f;
     channelCount = Format.NO_VALUE;
     channelCount = Format.NO_VALUE;
     sampleRateHz = Format.NO_VALUE;
     sampleRateHz = Format.NO_VALUE;
+    outputSampleRateHz = Format.NO_VALUE;
     buffer = EMPTY_BUFFER;
     buffer = EMPTY_BUFFER;
     shortBuffer = buffer.asShortBuffer();
     shortBuffer = buffer.asShortBuffer();
     outputBuffer = EMPTY_BUFFER;
     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
   @Override
@@ -119,17 +150,22 @@ public final class SonicAudioProcessor implements AudioProcessor {
     if (encoding != C.ENCODING_PCM_16BIT) {
     if (encoding != C.ENCODING_PCM_16BIT) {
       throw new UnhandledFormatException(sampleRateHz, channelCount, encoding);
       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;
       return false;
     }
     }
     this.sampleRateHz = sampleRateHz;
     this.sampleRateHz = sampleRateHz;
     this.channelCount = channelCount;
     this.channelCount = channelCount;
+    this.outputSampleRateHz = outputSampleRateHz;
     return true;
     return true;
   }
   }
 
 
   @Override
   @Override
   public boolean isActive() {
   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
   @Override
@@ -143,6 +179,11 @@ public final class SonicAudioProcessor implements AudioProcessor {
   }
   }
 
 
   @Override
   @Override
+  public int getOutputSampleRateHz() {
+    return outputSampleRateHz;
+  }
+
+  @Override
   public void queueInput(ByteBuffer inputBuffer) {
   public void queueInput(ByteBuffer inputBuffer) {
     if (inputBuffer.hasRemaining()) {
     if (inputBuffer.hasRemaining()) {
       ShortBuffer shortBuffer = inputBuffer.asShortBuffer();
       ShortBuffer shortBuffer = inputBuffer.asShortBuffer();
@@ -187,9 +228,7 @@ public final class SonicAudioProcessor implements AudioProcessor {
 
 
   @Override
   @Override
   public void flush() {
   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;
     outputBuffer = EMPTY_BUFFER;
     inputBytes = 0;
     inputBytes = 0;
     outputBytes = 0;
     outputBytes = 0;
@@ -204,9 +243,11 @@ public final class SonicAudioProcessor implements AudioProcessor {
     outputBuffer = EMPTY_BUFFER;
     outputBuffer = EMPTY_BUFFER;
     channelCount = Format.NO_VALUE;
     channelCount = Format.NO_VALUE;
     sampleRateHz = Format.NO_VALUE;
     sampleRateHz = Format.NO_VALUE;
+    outputSampleRateHz = Format.NO_VALUE;
     inputBytes = 0;
     inputBytes = 0;
     outputBytes = 0;
     outputBytes = 0;
     inputEnded = false;
     inputEnded = false;
+    pendingOutputSampleRateHz = SAMPLE_RATE_NO_CHANGE;
   }
   }
 
 
 }
 }

library/core/src/main/java/com/google/android/exoplayer2/audio/TrimmingAudioProcessor.java

@@ -31,6 +31,7 @@ import java.nio.ByteOrder;
   private int trimStartSamples;
   private int trimStartSamples;
   private int trimEndSamples;
   private int trimEndSamples;
   private int channelCount;
   private int channelCount;
+  private int sampleRateHz;
 
 
   private int pendingTrimStartBytes;
   private int pendingTrimStartBytes;
   private ByteBuffer buffer;
   private ByteBuffer buffer;
@@ -69,6 +70,7 @@ import java.nio.ByteOrder;
       throw new UnhandledFormatException(sampleRateHz, channelCount, encoding);
       throw new UnhandledFormatException(sampleRateHz, channelCount, encoding);
     }
     }
     this.channelCount = channelCount;
     this.channelCount = channelCount;
+    this.sampleRateHz = sampleRateHz;
     endBuffer = new byte[trimEndSamples * channelCount * 2];
     endBuffer = new byte[trimEndSamples * channelCount * 2];
     endBufferSize = 0;
     endBufferSize = 0;
     pendingTrimStartBytes = trimStartSamples * channelCount * 2;
     pendingTrimStartBytes = trimStartSamples * channelCount * 2;
@@ -93,6 +95,11 @@ import java.nio.ByteOrder;
   }
   }
 
 
   @Override
   @Override
+  public int getOutputSampleRateHz() {
+    return sampleRateHz;
+  }
+
+  @Override
   public void queueInput(ByteBuffer inputBuffer) {
   public void queueInput(ByteBuffer inputBuffer) {
     int position = inputBuffer.position();
     int position = inputBuffer.position();
     int limit = inputBuffer.limit();
     int limit = inputBuffer.limit();
@@ -174,6 +181,7 @@ import java.nio.ByteOrder;
     flush();
     flush();
     buffer = EMPTY_BUFFER;
     buffer = EMPTY_BUFFER;
     channelCount = Format.NO_VALUE;
     channelCount = Format.NO_VALUE;
+    sampleRateHz = Format.NO_VALUE;
     endBuffer = null;
     endBuffer = null;
   }
   }
 
 

library/core/src/test/java/com/google/android/exoplayer2/audio/SonicAudioProcessorTest.java

+/*
+ * 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.
+    }
+  }
+
+}