diff options
Diffstat (limited to 'services')
| -rw-r--r-- | services/audioflinger/AudioFlinger.cpp | 4 | ||||
| -rw-r--r-- | services/audioflinger/AudioPolicyService.cpp | 14 | ||||
| -rw-r--r-- | services/audioflinger/AudioPolicyService.h | 23 | ||||
| -rw-r--r-- | services/audioflinger/AudioResamplerSinc.cpp | 316 | ||||
| -rw-r--r-- | services/audioflinger/Tracks.cpp | 2 | ||||
| -rw-r--r-- | services/camera/libcameraservice/device3/Camera3Device.cpp | 2 |
6 files changed, 197 insertions, 164 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 26dac95..c0c34f7 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -525,7 +525,7 @@ sp<IAudioTrack> AudioFlinger::createTrack( } // Look for sync events awaiting for a session to be used. - for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { + for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { if (lStatus == NO_ERROR) { @@ -831,7 +831,7 @@ status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) AutoMutex lock(mLock); mStreamTypes[stream].mute = muted; - for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) + for (size_t i = 0; i < mPlaybackThreads.size(); i++) mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); return NO_ERROR; diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp index a37272d..6ea5324 100644 --- a/services/audioflinger/AudioPolicyService.cpp +++ b/services/audioflinger/AudioPolicyService.cpp @@ -678,6 +678,10 @@ AudioPolicyService::AudioCommandThread::~AudioCommandThread() if (!mAudioCommands.isEmpty()) { release_wake_lock(mName.string()); } + for (size_t k=0; k < mAudioCommands.size(); k++) { + delete mAudioCommands[k]->mParam; + delete mAudioCommands[k]; + } mAudioCommands.clear(); delete mpToneGenerator; } @@ -849,7 +853,7 @@ void AudioPolicyService::AudioCommandThread::startToneCommand(ToneGenerator::ton ToneData *data = new ToneData(); data->mType = type; data->mStream = stream; - command->mParam = (void *)data; + command->mParam = data; Mutex::Autolock _l(mLock); insertCommand_l(command); ALOGV("AudioCommandThread() adding tone start type %d, stream %d", type, stream); @@ -950,7 +954,7 @@ void AudioPolicyService::AudioCommandThread::stopOutputCommand(audio_io_handle_t data->mIO = output; data->mStream = stream; data->mSession = session; - command->mParam = (void *)data; + command->mParam = data; Mutex::Autolock _l(mLock); insertCommand_l(command); ALOGV("AudioCommandThread() adding stop output %d", output); @@ -963,7 +967,7 @@ void AudioPolicyService::AudioCommandThread::releaseOutputCommand(audio_io_handl command->mCommand = RELEASE_OUTPUT; ReleaseOutputData *data = new ReleaseOutputData(); data->mIO = output; - command->mParam = (void *)data; + command->mParam = data; Mutex::Autolock _l(mLock); insertCommand_l(command); ALOGV("AudioCommandThread() adding release output %d", output); @@ -1052,6 +1056,10 @@ void AudioPolicyService::AudioCommandThread::insertCommand_l(AudioCommand *comma for (size_t k = i + 1; k < mAudioCommands.size(); k++) { if (mAudioCommands[k] == removedCommands[j]) { ALOGV("suppressing command: %d", mAudioCommands[k]->mCommand); + // for commands that are not filtered, + // command->mParam is deleted in threadLoop + delete mAudioCommands[k]->mParam; + delete mAudioCommands[k]; mAudioCommands.removeAt(k); break; } diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h index ae053a9..a38160f 100644 --- a/services/audioflinger/AudioPolicyService.h +++ b/services/audioflinger/AudioPolicyService.h @@ -197,6 +197,8 @@ private: void insertCommand_l(AudioCommand *command, int delayMs = 0); private: + class AudioCommandData; + // descriptor for requested tone playback event class AudioCommand { @@ -211,41 +213,48 @@ private: Condition mCond; // condition for status return status_t mStatus; // command status bool mWaitStatus; // true if caller is waiting for status - void *mParam; // command parameter (ToneData, VolumeData, ParametersData) + AudioCommandData *mParam; // command specific parameter data + }; + + class AudioCommandData { + public: + virtual ~AudioCommandData() {} + protected: + AudioCommandData() {} }; - class ToneData { + class ToneData : public AudioCommandData { public: ToneGenerator::tone_type mType; // tone type (START_TONE only) audio_stream_type_t mStream; // stream type (START_TONE only) }; - class VolumeData { + class VolumeData : public AudioCommandData { public: audio_stream_type_t mStream; float mVolume; audio_io_handle_t mIO; }; - class ParametersData { + class ParametersData : public AudioCommandData { public: audio_io_handle_t mIO; String8 mKeyValuePairs; }; - class VoiceVolumeData { + class VoiceVolumeData : public AudioCommandData { public: float mVolume; }; - class StopOutputData { + class StopOutputData : public AudioCommandData { public: audio_io_handle_t mIO; audio_stream_type_t mStream; int mSession; }; - class ReleaseOutputData { + class ReleaseOutputData : public AudioCommandData { public: audio_io_handle_t mIO; }; diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp index 207f26b..e50b192 100644 --- a/services/audioflinger/AudioResamplerSinc.cpp +++ b/services/audioflinger/AudioResamplerSinc.cpp @@ -17,6 +17,7 @@ #define LOG_TAG "AudioResamplerSinc" //#define LOG_NDEBUG 0 +#define __STDC_CONSTANT_MACROS #include <malloc.h> #include <string.h> #include <stdlib.h> @@ -37,12 +38,14 @@ #define USE_INLINE_ASSEMBLY (false) #endif -#if USE_INLINE_ASSEMBLY && defined(__ARM_NEON__) -#define USE_NEON (true) +#if defined(__aarch64__) || defined(__ARM_NEON__) +#include <arm_neon.h> +#define USE_NEON #else -#define USE_NEON (false) +#undef USE_NEON #endif +#define UNUSED(x) ((void)(x)) namespace android { // ---------------------------------------------------------------------------- @@ -634,8 +637,8 @@ void AudioResamplerSinc::read( } template<int CHANNELS> -void AudioResamplerSinc::filterCoefficient( - int32_t* out, uint32_t phase, const int16_t *samples, uint32_t vRL) +void AudioResamplerSinc::filterCoefficient(int32_t* out, uint32_t phase, + const int16_t *samples, uint32_t vRL) { // NOTE: be very careful when modifying the code here. register // pressure is very high and a small change might cause the compiler @@ -662,160 +665,171 @@ void AudioResamplerSinc::filterCoefficient( size_t count = offset; - if (!USE_NEON) { - int32_t l = 0; - int32_t r = 0; - for (size_t i=0 ; i<count ; i++) { - interpolate<CHANNELS>(l, r, coefsP++, offset, lerpP, sP); - sP -= CHANNELS; - interpolate<CHANNELS>(l, r, coefsN++, offset, lerpN, sN); - sN += CHANNELS; - } - out[0] += 2 * mulRL(1, l, vRL); - out[1] += 2 * mulRL(0, r, vRL); - } else if (CHANNELS == 1) { +#ifndef USE_NEON + int32_t l = 0; + int32_t r = 0; + for (size_t i=0 ; i<count ; i++) { + interpolate<CHANNELS>(l, r, coefsP++, offset, lerpP, sP); + sP -= CHANNELS; + interpolate<CHANNELS>(l, r, coefsN++, offset, lerpN, sN); + sN += CHANNELS; + } + out[0] += 2 * mulRL(1, l, vRL); + out[1] += 2 * mulRL(0, r, vRL); +#else + UNUSED(vRL); + if (CHANNELS == 1) { int32_t const* coefsP1 = coefsP + offset; int32_t const* coefsN1 = coefsN + offset; sP -= CHANNELS*3; - asm ( - "vmov.32 d2[0], %[lerpP] \n" // load the positive phase - "vmov.32 d2[1], %[lerpN] \n" // load the negative phase - "veor q0, q0, q0 \n" // result, initialize to 0 - "vshl.s32 d2, d2, #16 \n" // convert to 32 bits - - "1: \n" - "vld1.16 { d4}, [%[sP]] \n" // load 4 16-bits stereo samples - "vld1.32 { q8}, [%[coefsP0]:128]! \n" // load 4 32-bits coefs - "vld1.32 { q9}, [%[coefsP1]:128]! \n" // load 4 32-bits coefs for interpolation - "vld1.16 { d6}, [%[sN]]! \n" // load 4 16-bits stereo samples - "vld1.32 {q10}, [%[coefsN0]:128]! \n" // load 4 32-bits coefs - "vld1.32 {q11}, [%[coefsN1]:128]! \n" // load 4 32-bits coefs for interpolation - - "vrev64.16 d4, d4 \n" // reverse 2 frames of the positive side - - "vsub.s32 q9, q9, q8 \n" // interpolate (step1) 1st set of coefs - "vsub.s32 q11, q11, q10 \n" // interpolate (step1) 2nd set of coets - "vshll.s16 q12, d4, #15 \n" // extend samples to 31 bits - - "vqrdmulh.s32 q9, q9, d2[0] \n" // interpolate (step2) 1st set of coefs - "vqrdmulh.s32 q11, q11, d2[1] \n" // interpolate (step3) 2nd set of coefs - "vshll.s16 q14, d6, #15 \n" // extend samples to 31 bits - - "vadd.s32 q8, q8, q9 \n" // interpolate (step3) 1st set - "vadd.s32 q10, q10, q11 \n" // interpolate (step4) 2nd set - "subs %[count], %[count], #4 \n" // update loop counter - - "vqrdmulh.s32 q12, q12, q8 \n" // multiply samples by interpolated coef - "vqrdmulh.s32 q14, q14, q10 \n" // multiply samples by interpolated coef - "sub %[sP], %[sP], #8 \n" // move pointer to next set of samples - - "vadd.s32 q0, q0, q12 \n" // accumulate result - "vadd.s32 q0, q0, q14 \n" // accumulate result - - "bne 1b \n" // loop - - "vld1.s32 {d2}, [%[vLR]] \n" // load volumes - "vld1.s32 {d3}, %[out] \n" // load the output - "vpadd.s32 d0, d0, d1 \n" // add all 4 partial sums - "vpadd.s32 d0, d0, d0 \n" // together - "vdup.i32 d0, d0[0] \n" // interleave L,R channels - "vqrdmulh.s32 d0, d0, d2 \n" // apply volume - "vadd.s32 d3, d3, d0 \n" // accumulate result - "vst1.s32 {d3}, %[out] \n" // store result - - : [out] "=Uv" (out[0]), - [count] "+r" (count), - [coefsP0] "+r" (coefsP), - [coefsP1] "+r" (coefsP1), - [coefsN0] "+r" (coefsN), - [coefsN1] "+r" (coefsN1), - [sP] "+r" (sP), - [sN] "+r" (sN) - : [lerpP] "r" (lerpP), - [lerpN] "r" (lerpN), - [vLR] "r" (mVolumeSIMD) - : "cc", "memory", - "q0", "q1", "q2", "q3", - "q8", "q9", "q10", "q11", - "q12", "q14" - ); + + int32x4_t sum; + int32x2_t lerpPN; + lerpPN = vdup_n_s32(0); + lerpPN = vld1_lane_s32((int32_t *)&lerpP, lerpPN, 0); + lerpPN = vld1_lane_s32((int32_t *)&lerpN, lerpPN, 1); + lerpPN = vshl_n_s32(lerpPN, 16); + sum = vdupq_n_s32(0); + + int16x4_t sampleP, sampleN; + int32x4_t samplePExt, sampleNExt; + int32x4_t coefsPV0, coefsPV1, coefsNV0, coefsNV1; + + coefsP = (const int32_t*)__builtin_assume_aligned(coefsP, 16); + coefsN = (const int32_t*)__builtin_assume_aligned(coefsN, 16); + coefsP1 = (const int32_t*)__builtin_assume_aligned(coefsP1, 16); + coefsN1 = (const int32_t*)__builtin_assume_aligned(coefsN1, 16); + for (; count > 0; count -= 4) { + sampleP = vld1_s16(sP); + sampleN = vld1_s16(sN); + coefsPV0 = vld1q_s32(coefsP); + coefsNV0 = vld1q_s32(coefsN); + coefsPV1 = vld1q_s32(coefsP1); + coefsNV1 = vld1q_s32(coefsN1); + sP -= 4; + sN += 4; + coefsP += 4; + coefsN += 4; + coefsP1 += 4; + coefsN1 += 4; + + sampleP = vrev64_s16(sampleP); + + // interpolate (step1) + coefsPV1 = vsubq_s32(coefsPV1, coefsPV0); + coefsNV1 = vsubq_s32(coefsNV1, coefsNV0); + samplePExt = vshll_n_s16(sampleP, 15); + // interpolate (step2) + coefsPV1 = vqrdmulhq_lane_s32(coefsPV1, lerpPN, 0); + coefsNV1 = vqrdmulhq_lane_s32(coefsNV1, lerpPN, 1); + sampleNExt = vshll_n_s16(sampleN, 15); + // interpolate (step3) + coefsPV0 = vaddq_s32(coefsPV0, coefsPV1); + coefsNV0 = vaddq_s32(coefsNV0, coefsNV1); + + samplePExt = vqrdmulhq_s32(samplePExt, coefsPV0); + sampleNExt = vqrdmulhq_s32(sampleNExt, coefsNV0); + sum = vaddq_s32(sum, samplePExt); + sum = vaddq_s32(sum, sampleNExt); + } + int32x2_t volumesV, outV; + volumesV = vld1_s32(mVolumeSIMD); + outV = vld1_s32(out); + + //add all 4 partial sums + int32x2_t sumLow, sumHigh; + sumLow = vget_low_s32(sum); + sumHigh = vget_high_s32(sum); + sumLow = vpadd_s32(sumLow, sumHigh); + sumLow = vpadd_s32(sumLow, sumLow); + + sumLow = vqrdmulh_s32(sumLow, volumesV); + outV = vadd_s32(outV, sumLow); + vst1_s32(out, outV); } else if (CHANNELS == 2) { int32_t const* coefsP1 = coefsP + offset; int32_t const* coefsN1 = coefsN + offset; sP -= CHANNELS*3; - asm ( - "vmov.32 d2[0], %[lerpP] \n" // load the positive phase - "vmov.32 d2[1], %[lerpN] \n" // load the negative phase - "veor q0, q0, q0 \n" // result, initialize to 0 - "veor q4, q4, q4 \n" // result, initialize to 0 - "vshl.s32 d2, d2, #16 \n" // convert to 32 bits - - "1: \n" - "vld2.16 {d4,d5}, [%[sP]] \n" // load 4 16-bits stereo samples - "vld1.32 { q8}, [%[coefsP0]:128]! \n" // load 4 32-bits coefs - "vld1.32 { q9}, [%[coefsP1]:128]! \n" // load 4 32-bits coefs for interpolation - "vld2.16 {d6,d7}, [%[sN]]! \n" // load 4 16-bits stereo samples - "vld1.32 {q10}, [%[coefsN0]:128]! \n" // load 4 32-bits coefs - "vld1.32 {q11}, [%[coefsN1]:128]! \n" // load 4 32-bits coefs for interpolation - - "vrev64.16 d4, d4 \n" // reverse 2 frames of the positive side - "vrev64.16 d5, d5 \n" // reverse 2 frames of the positive side - - "vsub.s32 q9, q9, q8 \n" // interpolate (step1) 1st set of coefs - "vsub.s32 q11, q11, q10 \n" // interpolate (step1) 2nd set of coets - "vshll.s16 q12, d4, #15 \n" // extend samples to 31 bits - "vshll.s16 q13, d5, #15 \n" // extend samples to 31 bits - - "vqrdmulh.s32 q9, q9, d2[0] \n" // interpolate (step2) 1st set of coefs - "vqrdmulh.s32 q11, q11, d2[1] \n" // interpolate (step3) 2nd set of coefs - "vshll.s16 q14, d6, #15 \n" // extend samples to 31 bits - "vshll.s16 q15, d7, #15 \n" // extend samples to 31 bits - - "vadd.s32 q8, q8, q9 \n" // interpolate (step3) 1st set - "vadd.s32 q10, q10, q11 \n" // interpolate (step4) 2nd set - "subs %[count], %[count], #4 \n" // update loop counter - - "vqrdmulh.s32 q12, q12, q8 \n" // multiply samples by interpolated coef - "vqrdmulh.s32 q13, q13, q8 \n" // multiply samples by interpolated coef - "vqrdmulh.s32 q14, q14, q10 \n" // multiply samples by interpolated coef - "vqrdmulh.s32 q15, q15, q10 \n" // multiply samples by interpolated coef - "sub %[sP], %[sP], #16 \n" // move pointer to next set of samples - - "vadd.s32 q0, q0, q12 \n" // accumulate result - "vadd.s32 q4, q4, q13 \n" // accumulate result - "vadd.s32 q0, q0, q14 \n" // accumulate result - "vadd.s32 q4, q4, q15 \n" // accumulate result - - "bne 1b \n" // loop - - "vld1.s32 {d2}, [%[vLR]] \n" // load volumes - "vld1.s32 {d3}, %[out] \n" // load the output - "vpadd.s32 d0, d0, d1 \n" // add all 4 partial sums from q0 - "vpadd.s32 d8, d8, d9 \n" // add all 4 partial sums from q4 - "vpadd.s32 d0, d0, d0 \n" // together - "vpadd.s32 d8, d8, d8 \n" // together - "vtrn.s32 d0, d8 \n" // interlace L,R channels - "vqrdmulh.s32 d0, d0, d2 \n" // apply volume - "vadd.s32 d3, d3, d0 \n" // accumulate result - "vst1.s32 {d3}, %[out] \n" // store result - - : [out] "=Uv" (out[0]), - [count] "+r" (count), - [coefsP0] "+r" (coefsP), - [coefsP1] "+r" (coefsP1), - [coefsN0] "+r" (coefsN), - [coefsN1] "+r" (coefsN1), - [sP] "+r" (sP), - [sN] "+r" (sN) - : [lerpP] "r" (lerpP), - [lerpN] "r" (lerpN), - [vLR] "r" (mVolumeSIMD) - : "cc", "memory", - "q0", "q1", "q2", "q3", "q4", - "q8", "q9", "q10", "q11", - "q12", "q13", "q14", "q15" - ); + + int32x4_t sum0, sum1; + int32x2_t lerpPN; + + lerpPN = vdup_n_s32(0); + lerpPN = vld1_lane_s32((int32_t *)&lerpP, lerpPN, 0); + lerpPN = vld1_lane_s32((int32_t *)&lerpN, lerpPN, 1); + lerpPN = vshl_n_s32(lerpPN, 16); + sum0 = vdupq_n_s32(0); + sum1 = vdupq_n_s32(0); + + int16x4x2_t sampleP, sampleN; + int32x4x2_t samplePExt, sampleNExt; + int32x4_t coefsPV0, coefsPV1, coefsNV0, coefsNV1; + + coefsP = (const int32_t*)__builtin_assume_aligned(coefsP, 16); + coefsN = (const int32_t*)__builtin_assume_aligned(coefsN, 16); + coefsP1 = (const int32_t*)__builtin_assume_aligned(coefsP1, 16); + coefsN1 = (const int32_t*)__builtin_assume_aligned(coefsN1, 16); + for (; count > 0; count -= 4) { + sampleP = vld2_s16(sP); + sampleN = vld2_s16(sN); + coefsPV0 = vld1q_s32(coefsP); + coefsNV0 = vld1q_s32(coefsN); + coefsPV1 = vld1q_s32(coefsP1); + coefsNV1 = vld1q_s32(coefsN1); + sP -= 8; + sN += 8; + coefsP += 4; + coefsN += 4; + coefsP1 += 4; + coefsN1 += 4; + + sampleP.val[0] = vrev64_s16(sampleP.val[0]); + sampleP.val[1] = vrev64_s16(sampleP.val[1]); + + // interpolate (step1) + coefsPV1 = vsubq_s32(coefsPV1, coefsPV0); + coefsNV1 = vsubq_s32(coefsNV1, coefsNV0); + samplePExt.val[0] = vshll_n_s16(sampleP.val[0], 15); + samplePExt.val[1] = vshll_n_s16(sampleP.val[1], 15); + // interpolate (step2) + coefsPV1 = vqrdmulhq_lane_s32(coefsPV1, lerpPN, 0); + coefsNV1 = vqrdmulhq_lane_s32(coefsNV1, lerpPN, 1); + sampleNExt.val[0] = vshll_n_s16(sampleN.val[0], 15); + sampleNExt.val[1] = vshll_n_s16(sampleN.val[1], 15); + // interpolate (step3) + coefsPV0 = vaddq_s32(coefsPV0, coefsPV1); + coefsNV0 = vaddq_s32(coefsNV0, coefsNV1); + + samplePExt.val[0] = vqrdmulhq_s32(samplePExt.val[0], coefsPV0); + samplePExt.val[1] = vqrdmulhq_s32(samplePExt.val[1], coefsPV0); + sampleNExt.val[0] = vqrdmulhq_s32(sampleNExt.val[0], coefsNV0); + sampleNExt.val[1] = vqrdmulhq_s32(sampleNExt.val[1], coefsNV0); + sum0 = vaddq_s32(sum0, samplePExt.val[0]); + sum1 = vaddq_s32(sum1, samplePExt.val[1]); + sum0 = vaddq_s32(sum0, sampleNExt.val[0]); + sum1 = vaddq_s32(sum1, sampleNExt.val[1]); + } + int32x2_t volumesV, outV; + volumesV = vld1_s32(mVolumeSIMD); + outV = vld1_s32(out); + + //add all 4 partial sums + int32x2_t sumLow0, sumHigh0, sumLow1, sumHigh1; + sumLow0 = vget_low_s32(sum0); + sumHigh0 = vget_high_s32(sum0); + sumLow1 = vget_low_s32(sum1); + sumHigh1 = vget_high_s32(sum1); + sumLow0 = vpadd_s32(sumLow0, sumHigh0); + sumLow0 = vpadd_s32(sumLow0, sumLow0); + sumLow1 = vpadd_s32(sumLow1, sumHigh1); + sumLow1 = vpadd_s32(sumLow1, sumLow1); + + sumLow0 = vtrn_s32(sumLow0, sumLow1).val[0]; + sumLow0 = vqrdmulh_s32(sumLow0, volumesV); + outV = vadd_s32(outV, sumLow0); + vst1_s32(out, outV); } +#endif } template<int CHANNELS> diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp index fccc7b8..b59333f 100644 --- a/services/audioflinger/Tracks.cpp +++ b/services/audioflinger/Tracks.cpp @@ -895,7 +895,7 @@ bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWrit void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) { - for (int i = 0; i < (int)mSyncEvents.size(); i++) { + for (size_t i = 0; i < mSyncEvents.size(); i++) { if (mSyncEvents[i]->type() == type) { mSyncEvents[i]->trigger(); mSyncEvents.removeAt(i); diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp index 7e11a3b..1d4768c 100644 --- a/services/camera/libcameraservice/device3/Camera3Device.cpp +++ b/services/camera/libcameraservice/device3/Camera3Device.cpp @@ -654,6 +654,8 @@ status_t Camera3Device::createZslStream( } mInputStream = newStream; + mNeedConfig = true; + *id = mNextStreamId++; *zslStream = newStream; |