/* * Copyright (C) 2014 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. */ #ifndef ANDROID_AUDIO_RESAMPLER_PUBLIC_H #define ANDROID_AUDIO_RESAMPLER_PUBLIC_H #include #include namespace android { // AUDIO_RESAMPLER_DOWN_RATIO_MAX is the maximum ratio between the original // audio sample rate and the target rate when downsampling, // as permitted in the audio framework, e.g. AudioTrack and AudioFlinger. // In practice, it is not recommended to downsample more than 6:1 // for best audio quality, even though the audio framework permits a larger // downsampling ratio. // TODO: replace with an API #define AUDIO_RESAMPLER_DOWN_RATIO_MAX 256 // AUDIO_RESAMPLER_UP_RATIO_MAX is the maximum suggested ratio between the original // audio sample rate and the target rate when upsampling. It is loosely enforced by // the system. One issue with large upsampling ratios is the approximation by // an int32_t of the phase increments, making the resulting sample rate inexact. #define AUDIO_RESAMPLER_UP_RATIO_MAX 65536 // 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 // 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; } static inline bool isAudioPlaybackRateValid(const AudioPlaybackRate &playbackRate) { if (playbackRate.mFallbackMode == AUDIO_TIMESTRETCH_FALLBACK_FAIL && (playbackRate.mStretchMode == AUDIO_TIMESTRETCH_STRETCH_SPEECH || playbackRate.mStretchMode == AUDIO_TIMESTRETCH_STRETCH_DEFAULT)) { //test sonic specific constraints return playbackRate.mSpeed >= TIMESTRETCH_SONIC_SPEED_MIN && playbackRate.mSpeed <= TIMESTRETCH_SONIC_SPEED_MAX && playbackRate.mPitch >= AUDIO_TIMESTRETCH_PITCH_MIN && playbackRate.mPitch <= AUDIO_TIMESTRETCH_PITCH_MAX; } else { return playbackRate.mSpeed >= AUDIO_TIMESTRETCH_SPEED_MIN && playbackRate.mSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX && playbackRate.mPitch >= AUDIO_TIMESTRETCH_PITCH_MIN && playbackRate.mPitch <= AUDIO_TIMESTRETCH_PITCH_MAX; } } // 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. // If srcSampleRate and dstSampleRate are equal, then it returns destination frames, which // may not be true if the resampler is asynchronous. static inline size_t sourceFramesNeeded( uint32_t srcSampleRate, size_t dstFramesRequired, uint32_t dstSampleRate) { // +1 for rounding - always do this even if matched ratio (resampler may use phases not ratio) // +1 for additional sample needed for interpolation return srcSampleRate == dstSampleRate ? dstFramesRequired : size_t((uint64_t)dstFramesRequired * srcSampleRate / dstSampleRate + 1 + 1); } // An upper bound for the number of destination frames possible from srcFrames // after sample rate conversion. This may be used for buffer sizing. static inline size_t destinationFramesPossible(size_t srcFrames, uint32_t srcSampleRate, uint32_t dstSampleRate) { if (srcSampleRate == dstSampleRate) { return srcFrames; } uint64_t dstFrames = (uint64_t)srcFrames * dstSampleRate / srcSampleRate; 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 } // Identifies sample rates that we associate with music // and thus eligible for better resampling and fast capture. // This is somewhat less than 44100 to allow for pitch correction // involving resampling as well as asynchronous resampling. #define AUDIO_PROCESSING_MUSIC_RATE 40000 static inline bool isMusicRate(uint32_t sampleRate) { return sampleRate >= AUDIO_PROCESSING_MUSIC_RATE; } } // namespace android // --------------------------------------------------------------------------- #endif // ANDROID_AUDIO_RESAMPLER_PUBLIC_H