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authorAndy Hung <hunga@google.com>2015-04-10 03:28:12 +0000
committerAndroid (Google) Code Review <android-gerrit@google.com>2015-04-10 03:28:12 +0000
commit7d014e50b3acd60e73ed7d7a74dc58485c7a413c (patch)
treed2d5a07889b721c5d775c9da8c9dbc0531d19173 /include/media
parente62e16371b6c2f33072b3345e1d1853673b9a3d8 (diff)
parent8edb8dc44b8a2f81bdb5db645b6b708548771a31 (diff)
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Merge "Add playback rate to AudioTrack"
Diffstat (limited to 'include/media')
-rw-r--r--include/media/AudioResamplerPublic.h11
-rw-r--r--include/media/AudioTrack.h20
2 files changed, 31 insertions, 0 deletions
diff --git a/include/media/AudioResamplerPublic.h b/include/media/AudioResamplerPublic.h
index 3c30148..07d946d 100644
--- a/include/media/AudioResamplerPublic.h
+++ b/include/media/AudioResamplerPublic.h
@@ -42,6 +42,8 @@
#define AUDIO_TIMESTRETCH_PITCH_MAX 2.0f
#define AUDIO_TIMESTRETCH_PITCH_NORMAL 1.0f
+// TODO: Consider putting these inlines into a class scope
+
// Returns the source frames needed to resample to destination frames. This is not a precise
// value and depends on the resampler (and possibly how it handles rounding internally).
// Nevertheless, this should be an upper bound on the requirements of the resampler.
@@ -66,4 +68,13 @@ static inline size_t destinationFramesPossible(size_t srcFrames, uint32_t srcSam
return dstFrames > 2 ? dstFrames - 2 : 0;
}
+static inline size_t sourceFramesNeededWithTimestretch(
+ uint32_t srcSampleRate, size_t dstFramesRequired, uint32_t dstSampleRate,
+ float speed) {
+ // required is the number of input frames the resampler needs
+ size_t required = sourceFramesNeeded(srcSampleRate, dstFramesRequired, dstSampleRate);
+ // to deliver this, the time stretcher requires:
+ return required * (double)speed + 1 + 1; // accounting for rounding dependencies
+}
+
#endif // ANDROID_AUDIO_RESAMPLER_PUBLIC_H
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index e7e0703..a06197f 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -359,6 +359,21 @@ public:
/* Return current source sample rate in Hz */
uint32_t getSampleRate() const;
+ /* Set source playback rate for timestretch
+ * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster
+ * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch
+ *
+ * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX
+ * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX
+ *
+ * Speed increases the playback rate of media, but does not alter pitch.
+ * Pitch increases the "tonal frequency" of media, but does not affect the playback rate.
+ */
+ status_t setPlaybackRate(float speed, float pitch);
+
+ /* Return current playback rate */
+ void getPlaybackRate(float *speed, float *pitch) const;
+
/* Enables looping and sets the start and end points of looping.
* Only supported for static buffer mode.
*
@@ -719,6 +734,9 @@ protected:
// increment mPosition by the delta of mServer, and return new value of mPosition
uint32_t updateAndGetPosition_l();
+ // check sample rate and speed is compatible with AudioTrack
+ bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const;
+
// Next 4 fields may be changed if IAudioTrack is re-created, but always != 0
sp<IAudioTrack> mAudioTrack;
sp<IMemory> mCblkMemory;
@@ -730,6 +748,8 @@ protected:
float mVolume[2];
float mSendLevel;
mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it
+ float mSpeed; // timestretch: 1.0f for normal speed.
+ float mPitch; // timestretch: 1.0f for normal pitch.
size_t mFrameCount; // corresponds to current IAudioTrack, value is
// reported back by AudioFlinger to the client
size_t mReqFrameCount; // frame count to request the first or next time