Use AudioPlaybackRate to hold TimestretchBufferProvider parameters
Use this struct to handle the parameters for TimestretchBufferProvider all
across the system.
Add stretch mode and fallback mode to TimestretchBuffer Provider.
Change-Id: I19099924a7003c62e48bb6ead56c785cb129fba2
diff --git a/include/media/AudioResamplerPublic.h b/include/media/AudioResamplerPublic.h
index 07d946d..53b8c13 100644
--- a/include/media/AudioResamplerPublic.h
+++ b/include/media/AudioResamplerPublic.h
@@ -18,6 +18,9 @@
#define ANDROID_AUDIO_RESAMPLER_PUBLIC_H
#include <stdint.h>
+#include <math.h>
+
+namespace android {
// AUDIO_RESAMPLER_DOWN_RATIO_MAX is the maximum ratio between the original
// audio sample rate and the target rate when downsampling,
@@ -34,13 +37,76 @@
// an int32_t of the phase increments, making the resulting sample rate inexact.
#define AUDIO_RESAMPLER_UP_RATIO_MAX 65536
-#define AUDIO_TIMESTRETCH_SPEED_MIN 0.5f
-#define AUDIO_TIMESTRETCH_SPEED_MAX 2.0f
+// AUDIO_TIMESTRETCH_SPEED_MIN and AUDIO_TIMESTRETCH_SPEED_MAX define the min and max time stretch
+// speeds supported by the system. These are enforced by the system and values outside this range
+// will result in a runtime error.
+// Depending on the AudioPlaybackRate::mStretchMode, the effective limits might be narrower than
+// the ones specified here
+// AUDIO_TIMESTRETCH_SPEED_MIN_DELTA is the minimum absolute speed difference that might trigger a
+// parameter update
+#define AUDIO_TIMESTRETCH_SPEED_MIN 0.01f
+#define AUDIO_TIMESTRETCH_SPEED_MAX 20.0f
#define AUDIO_TIMESTRETCH_SPEED_NORMAL 1.0f
+#define AUDIO_TIMESTRETCH_SPEED_MIN_DELTA 0.0001f
-#define AUDIO_TIMESTRETCH_PITCH_MIN 0.5f
-#define AUDIO_TIMESTRETCH_PITCH_MAX 2.0f
+// AUDIO_TIMESTRETCH_PITCH_MIN and AUDIO_TIMESTRETCH_PITCH_MAX define the min and max time stretch
+// pitch shifting supported by the system. These are not enforced by the system and values
+// outside this range might result in a pitch different than the one requested.
+// Depending on the AudioPlaybackRate::mStretchMode, the effective limits might be narrower than
+// the ones specified here.
+// AUDIO_TIMESTRETCH_PITCH_MIN_DELTA is the minimum absolute pitch difference that might trigger a
+// parameter update
+#define AUDIO_TIMESTRETCH_PITCH_MIN 0.25f
+#define AUDIO_TIMESTRETCH_PITCH_MAX 4.0f
#define AUDIO_TIMESTRETCH_PITCH_NORMAL 1.0f
+#define AUDIO_TIMESTRETCH_PITCH_MIN_DELTA 0.0001f
+
+
+//Determines the current algorithm used for stretching
+enum AudioTimestretchStretchMode : int32_t {
+ AUDIO_TIMESTRETCH_STRETCH_DEFAULT = 0,
+ AUDIO_TIMESTRETCH_STRETCH_SPEECH = 1,
+ //TODO: add more stretch modes/algorithms
+};
+
+//Limits for AUDIO_TIMESTRETCH_STRETCH_SPEECH mode
+#define TIMESTRETCH_SONIC_SPEED_MIN 0.1f
+#define TIMESTRETCH_SONIC_SPEED_MAX 6.0f
+
+//Determines behavior of Timestretch if current algorithm can't perform
+//with current parameters.
+// FALLBACK_CUT_REPEAT: (internal only) for speed <1.0 will truncate frames
+// for speed > 1.0 will repeat frames
+// FALLBACK_MUTE: will set all processed frames to zero
+// FALLBACK_FAIL: will stop program execution and log a fatal error
+enum AudioTimestretchFallbackMode : int32_t {
+ AUDIO_TIMESTRETCH_FALLBACK_CUT_REPEAT = -1,
+ AUDIO_TIMESTRETCH_FALLBACK_DEFAULT = 0,
+ AUDIO_TIMESTRETCH_FALLBACK_MUTE = 1,
+ AUDIO_TIMESTRETCH_FALLBACK_FAIL = 2,
+};
+
+struct AudioPlaybackRate {
+ float mSpeed;
+ float mPitch;
+ enum AudioTimestretchStretchMode mStretchMode;
+ enum AudioTimestretchFallbackMode mFallbackMode;
+};
+
+static const AudioPlaybackRate AUDIO_PLAYBACK_RATE_DEFAULT = {
+ AUDIO_TIMESTRETCH_SPEED_NORMAL,
+ AUDIO_TIMESTRETCH_PITCH_NORMAL,
+ AUDIO_TIMESTRETCH_STRETCH_DEFAULT,
+ AUDIO_TIMESTRETCH_FALLBACK_DEFAULT
+};
+
+static inline bool isAudioPlaybackRateEqual(const AudioPlaybackRate &pr1,
+ const AudioPlaybackRate &pr2) {
+ return fabs(pr1.mSpeed - pr2.mSpeed) < AUDIO_TIMESTRETCH_SPEED_MIN_DELTA &&
+ fabs(pr1.mPitch - pr2.mPitch) < AUDIO_TIMESTRETCH_PITCH_MIN_DELTA &&
+ pr2.mStretchMode == pr2.mStretchMode &&
+ pr2.mFallbackMode == pr2.mFallbackMode;
+}
// TODO: Consider putting these inlines into a class scope
@@ -77,4 +143,8 @@
return required * (double)speed + 1 + 1; // accounting for rounding dependencies
}
+} // namespace android
+
+// ---------------------------------------------------------------------------
+
#endif // ANDROID_AUDIO_RESAMPLER_PUBLIC_H
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index a06197f..2d34c02 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -21,6 +21,7 @@
#include <media/AudioSystem.h>
#include <media/AudioTimestamp.h>
#include <media/IAudioTrack.h>
+#include <media/AudioResamplerPublic.h>
#include <utils/threads.h>
namespace android {
@@ -369,10 +370,10 @@
* 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);
+ status_t setPlaybackRate(const AudioPlaybackRate &playbackRate);
/* Return current playback rate */
- void getPlaybackRate(float *speed, float *pitch) const;
+ const AudioPlaybackRate& getPlaybackRate() const;
/* Enables looping and sets the start and end points of looping.
* Only supported for static buffer mode.
@@ -748,8 +749,7 @@
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.
+ AudioPlaybackRate mPlaybackRate;
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