| /* |
| * 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 <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, |
| // 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 && |
| pr1.mStretchMode == pr2.mStretchMode && |
| pr1.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 |