Added NEON optimization for DDC.
Buffer size can now automatically adjust to sampling rate changes between csdr processes.
This commit is contained in:
ha7ilm
parent
6ee80ea7ce
commit
9cec3782e2
5 changed files with 582 additions and 109 deletions
10
Makefile
10
Makefile
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@ -32,7 +32,7 @@ LIBSOURCES = fft_fftw.c libcsdr_wrapper.c
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#SOURCES = csdr.c $(LIBSOURCES)
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cpufeature = $(if $(findstring $(1),$(shell cat /proc/cpuinfo)),$(2))
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PARAMS_SSE = $(call cpufeature,sse,-msse) $(call cpufeature,sse2,-msse2) $(call cpufeature,sse3,-msse3) $(call cpufeature,sse4,-msse4) $(call cpufeature,sse4_1,-msse4.1) $(call cpufeature,sse4_2,-msse4.2) -mfpmath=sse
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PARAMS_NEON = -mfloat-abi=hard -march=armv7-a -mtune=cortex-a8 -mfpu=neon -mvectorize-with-neon-quad -funsafe-math-optimizations -Wformat=0
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PARAMS_NEON = -mfloat-abi=hard -march=armv7-a -mtune=cortex-a8 -mfpu=neon -mvectorize-with-neon-quad -funsafe-math-optimizations -Wformat=0 -DNEON_OPTS
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#tnx Jan Szumiec for the Raspberry Pi support
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PARAMS_RASPI = -mfloat-abi=hard -mcpu=arm1176jzf-s -mfpu=vfp -funsafe-math-optimizations -Wformat=0
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PARAMS_ARM = $(if $(call cpufeature,BCM2708,dummy-text),$(PARAMS_RASPI),$(PARAMS_NEON))
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@ -47,12 +47,12 @@ all: clean-vect
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@echo NOTE: you may have to manually edit Makefile to optimize for your CPU \(especially if you compile on ARM, please edit PARAMS_NEON\).
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@echo Auto-detected optimization parameters: $(PARAMS_SIMD)
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@echo
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c99 $(PARAMS_LOOPVECT) $(PARAMS_SIMD) $(LIBSOURCES) $(PARAMS_LIBS) $(PARAMS_MISC) -fpic -shared -o libcsdr.so
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gcc -std=gnu99 $(PARAMS_LOOPVECT) $(PARAMS_SIMD) $(LIBSOURCES) $(PARAMS_LIBS) $(PARAMS_MISC) -fpic -shared -o libcsdr.so
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-./parsevect dumpvect*.vect
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c99 $(PARAMS_LOOPVECT) $(PARAMS_SIMD) csdr.c $(PARAMS_LIBS) -L. -lcsdr $(PARAMS_MISC) -o csdr
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gcc -std=gnu99 $(PARAMS_LOOPVECT) $(PARAMS_SIMD) csdr.c $(PARAMS_LIBS) -L. -lcsdr $(PARAMS_MISC) -o csdr
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arm-cross: clean-vect
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#note: this doesn't work since having added FFTW
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arm-linux-gnueabihf-gcc -std=c99 -O3 -fshort-double -ffast-math -dumpbase dumpvect-arm -fdump-tree-vect-details -mfloat-abi=softfp -march=armv7-a -mtune=cortex-a9 -mfpu=neon -mvectorize-with-neon-quad -Wno-unused-result -Wformat=0 $(SOURCES) -lm -o ./csdr
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arm-linux-gnueabihf-gcc -std=gnu99 -O3 -fshort-double -ffast-math -dumpbase dumpvect-arm -fdump-tree-vect-details -mfloat-abi=softfp -march=armv7-a -mtune=cortex-a9 -mfpu=neon -mvectorize-with-neon-quad -Wno-unused-result -Wformat=0 $(SOURCES) -lm -o ./csdr
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clean-vect:
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rm -f dumpvect*.vect
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clean: clean-vect
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@ -65,6 +65,8 @@ install:
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uninstall:
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rm /usr/lib/libcsdr.so /usr/bin/csdr /usr/bin/csdr-fm
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ldconfig
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disasm:
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objdump -S libcsdr.so > libcsdr.disasm
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emcc-clean:
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-rm sdr.js/sdr.js
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-rm sdr.js/sdrjs-compiled.js
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102
libcsdr.c
102
libcsdr.c
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@ -263,6 +263,75 @@ float shift_table_cc(complexf* input, complexf* output, int input_size, float ra
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return phase;
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}
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#ifdef NEON_OPTS
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#pragma message "We have a faster fir_decimate_cc now."
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//max help: http://community.arm.com/groups/android-community/blog/2015/03/27/arm-neon-programming-quick-reference
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int fir_decimate_cc(complexf *input, complexf *output, int input_size, int decimation, float *taps, int taps_length)
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{
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//Theory: http://www.dspguru.com/dsp/faqs/multirate/decimation
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//It uses real taps. It returns the number of output samples actually written.
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//It needs overlapping input based on its returned value:
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//number of processed input samples = returned value * decimation factor
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//The output buffer should be at least input_length / 3.
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// i: input index | ti: tap index | oi: output index
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int oi=0;
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for(int i=0; i<input_size; i+=decimation) //@fir_decimate_cc: outer loop
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{
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if(i+taps_length>input_size) break;
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register float acci=0;
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register float accq=0;
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register int ti=0;
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register float* pinput=(float*)&(input[i+ti]);
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register float* ptaps=taps;
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register float* ptaps_end=taps+taps_length;
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float quad_acciq [8];
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/*
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q0, q1: input signal I sample and Q sample
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q2: taps
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q4, q5: accumulator for I branch and Q branch (will be the output)
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*/
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//fprintf(stderr, "macska\n");
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asm volatile(
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//" vorr.f32 q4, #0\n\t" //null the accumulators
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//" vorr.f32 q5, #0\n\t"
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" vmov.f32 q4, #0.0\n\t" //another way to null the accumulators
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" vmov.f32 q5, #0.0\n\t"
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"for_fdccasm: vld2.32 {q0-q1}, [%[pinput]]!\n\t" //load q0 and q1 directly from the memory address stored in pinput, with interleaving (so that we get the I samples in q0 and the Q samples in q1), also increment the memory address in pinput (hence the "!" mark) //http://community.arm.com/groups/processors/blog/2010/03/17/coding-for-neon--part-1-load-and-stores
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" vld1.32 {q2}, [%[ptaps]]!\n\t"
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" vmla.f32 q4, q0, q2\n\t" //quad_acc_i += quad_input_i * quad_taps_1 //http://stackoverflow.com/questions/3240440/how-to-use-the-multiply-and-accumulate-intrinsics-in-arm-cortex-a8 //http://infocenter.arm.com/help/index.jsp?topic=/com.arm.doc.dui0489e/CIHEJBIE.html
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" vmla.f32 q5, q1, q2\n\t" //quad_acc_q += quad_input_q * quad_taps_1
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" cmp %[ptaps], %[ptaps_end]\n\t" //if(ptaps == ptaps_end)
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" bcc for_fdccasm\n\t" // then goto for_fdcasm
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" vst1.32 {q4}, [%[quad_acci]]\n\t" //if the loop is finished, store the two accumulators in memory
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" vst1.32 {q5}, [%[quad_accq]]\n\t"
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:
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[pinput]"+r"(pinput), [ptaps]"+r"(ptaps) //output operand list
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:
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[ptaps_end]"r"(ptaps_end), [quad_acci]"r"(quad_acciq), [quad_accq]"r"(quad_acciq+4) //input operand list
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:
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"memory", "q0", "q1", "q2", "q4", "q5", "cc" //clobber list
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);
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//original for loops for reference:
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//for(int ti=0; ti<taps_length; ti++) acci += (iof(input,i+ti)) * taps[ti]; //@fir_decimate_cc: i loop
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//for(int ti=0; ti<taps_length; ti++) accq += (qof(input,i+ti)) * taps[ti]; //@fir_decimate_cc: q loop
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//for(int n=0;n<8;n++) fprintf(stderr, "\n>> [%d] %g \n", n, quad_acciq[n]);
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iof(output,oi)=quad_acciq[0]+quad_acciq[1]+quad_acciq[2]+quad_acciq[3]; //we're still not ready, as we have to add up the contents of a quad accumulator register to get a single accumulated value
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qof(output,oi)=quad_acciq[4]+quad_acciq[5]+quad_acciq[6]+quad_acciq[7];
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oi++;
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}
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return oi;
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}
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#else
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int fir_decimate_cc(complexf *input, complexf *output, int input_size, int decimation, float *taps, int taps_length)
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{
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//Theory: http://www.dspguru.com/dsp/faqs/multirate/decimation
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@ -286,6 +355,34 @@ int fir_decimate_cc(complexf *input, complexf *output, int input_size, int decim
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return oi;
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}
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#endif
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/*
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int fir_decimate_cc(complexf *input, complexf *output, int input_size, int decimation, float *taps, int taps_length)
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{
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//Theory: http://www.dspguru.com/dsp/faqs/multirate/decimation
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//It uses real taps. It returns the number of output samples actually written.
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//It needs overlapping input based on its returned value:
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//number of processed input samples = returned value * decimation factor
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//The output buffer should be at least input_length / 3.
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// i: input index | ti: tap index | oi: output index
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int oi=0;
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for(int i=0; i<input_size; i+=decimation) //@fir_decimate_cc: outer loop
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{
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if(i+taps_length>input_size) break;
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float acci=0;
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int taps_halflength = taps_length/2;
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for(int ti=0; ti<taps_halflength; ti++) acci += (iof(input,i+ti)+iof(input,i+taps_length-ti)) * taps[ti]; //@fir_decimate_cc: i loop
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float accq=0;
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for(int ti=0; ti<taps_halflength; ti++) accq += (qof(input,i+ti)+qof(input,i+taps_length-ti)) * taps[ti]; //@fir_decimate_cc: q loop
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iof(output,oi)=acci+iof(input,i+taps_halflength)*taps[taps_halflength];
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qof(output,oi)=accq+qof(input,i+taps_halflength)*taps[taps_halflength];
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oi++;
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}
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return oi;
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}
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*/
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rational_resampler_ff_t rational_resampler_ff(float *input, float *output, int input_size, int interpolation, int decimation, float *taps, int taps_length, int last_taps_delay)
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{
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@ -524,7 +621,8 @@ float fastdcblock_ff(float* input, float* output, int input_size, float last_dc_
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return avg;
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}
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#define FASTAGC_MAX_GAIN (65e3)
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//#define FASTAGC_MAX_GAIN (65e3)
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#define FASTAGC_MAX_GAIN 50
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void fastagc_ff(fastagc_ff_t* input, float* output)
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{
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@ -553,6 +651,7 @@ void fastagc_ff(fastagc_ff_t* input, float* output)
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//we change the gain linearly on the apply_block from the last_gain to target_gain.
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float target_gain=input->reference/target_peak;
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if(target_gain>FASTAGC_MAX_GAIN) target_gain=FASTAGC_MAX_GAIN;
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//fprintf(stderr, "target_gain: %g\n",target_gain);
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for(int i=0;i<input->input_size;i++) //@fastagc_ff: apply gain
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{
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@ -572,7 +671,6 @@ void fastagc_ff(fastagc_ff_t* input, float* output)
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//fprintf(stderr,"target_gain=%g\n", target_gain);
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}
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/*
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______ __ __ _ _ _ _
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| ____| \/ | | | | | | | | |
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@ -189,12 +189,14 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
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{
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if(last_peak<input_abs)
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{
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attack_wait_counter=attack_wait_time;
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last_peak=input_abs;
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}
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if(attack_wait_counter>0)
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{
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attack_wait_counter--;
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//fprintf(stderr,"A");
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dgain=0;
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}
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else
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@ -203,6 +205,7 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
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dgain=error*attack_rate;
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//Before starting to increase the gain next time, we will be waiting until hang_time for sure.
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hang_counter=hang_time;
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}
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}
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else //DECREASE IN SIGNAL LEVEL
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@ -222,7 +225,7 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
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}
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//output[i]=gain*input[i]; //Here we do the actual scaling of the samples.
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//Here we do the actual scaling of the samples, but we run an IIR filter on the gain values:
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output[i]=(gain+last_gain-gain_filter_alpha*last_gain)*input[i]; //dc-pass-filter: freqz([1 -1],[1 -0.99]) y[i]=x[i]+y[i-1]-alpha*x[i-1]
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output[i]=(gain=gain+last_gain-gain_filter_alpha*last_gain)*input[i]; //dc-pass-filter: freqz([1 -1],[1 -0.99]) y[i]=x[i]+y[i-1]-alpha*x[i-1]
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//output[i]=input[i]*(last_gain+gain_filter_alpha*(gain-last_gain)); //LPF
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last_gain=gain;
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38
sdr.js/sdrjs-test.html
Normal file
38
sdr.js/sdrjs-test.html
Normal file
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@ -0,0 +1,38 @@
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<!doctype html>
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<!--
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This software is part of libcsdr, a set of simple DSP routines for
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Software Defined Radio.
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Copyright (c) 2014, Andras Retzler <randras@sdr.hu>
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All rights reserved.
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Redistribution and use in source and binary forms, with or without
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modification, are permitted provided that the following conditions are met:
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* Redistributions of source code must retain the above copyright
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notice, this list of conditions and the following disclaimer.
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* Redistributions in binary form must reproduce the above copyright
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notice, this list of conditions and the following disclaimer in the
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documentation and/or other materials provided with the distribution.
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* Neither the name of the copyright holder nor the
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names of its contributors may be used to endorse or promote products
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derived from this software without specific prior written permission.
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THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND
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ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
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WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
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DISCLAIMED. IN NO EVENT SHALL ANDRAS RETZLER BE LIABLE FOR ANY
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DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
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(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
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LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
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ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
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(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
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SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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-->
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<html lang="en-us">
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<head>
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<script type="text/javascript" src="sdr.js"></script>
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</head>
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<body style="font-family: monospace">
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<script type="text/javascript">test_firdes_lowpass_f_new();</script>
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</body>
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</html>
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