diff --git a/freq_analyzer.c b/freq_analyzer.c
new file mode 100644
index 0000000..fdda9c5
--- /dev/null
+++ b/freq_analyzer.c
@@ -0,0 +1,321 @@
+/*
+ *  Code for frequency analysis
+ *
+ *  Copyright (C) 2015  Nest Labs. All rights reserved.
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License version 2 as
+ *  published by the Free Software Foundation.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+ *
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <stdint.h>
+#include <math.h>
+#include "fftw3.h"
+#include "wav_header.h"
+
+#define SAMPLE_BYTES			2
+#define MAX_NUM_CHANNELS		2
+
+#define RECTANGULAR			1
+#define HAMMING				2
+#define BLACKMAN_NUTTALL		3
+#define HANNING				4
+#define BLACKMAN_HARRIS			5
+#define WINDOW_TYPE			HANNING
+
+#define SQUARE(a)			( (a) * (a) )
+
+void apply_window(int16_t *input, double *windowed, int32_t num_samples);
+
+#if WINDOW_TYPE == RECTANGULAR
+void apply_window(int16_t *input, double *windowed, int32_t num_samples)
+{
+	int32_t sample_index;
+	for(sample_index = 0; sample_index < num_samples; sample_index++)
+		windowed[sample_index] = (double)input[sample_index]/32767.0;
+}
+#elif WINDOW_TYPE == HAMMING
+void apply_window(int16_t *input, double *windowed, int32_t num_samples)
+{
+	int32_t sample_index;
+	for(sample_index = 0; sample_index < num_samples; sample_index++) { 
+		windowed[sample_index] = 0.54 - 0.46 * cos(2 * M_PI * sample_index/(num_samples -1));
+		windowed[sample_index] *= (double)input[sample_index]/32767.0;
+	}
+}
+#elif WINDOW_TYPE == BLACKMAN_NUTTALL 
+void apply_window(int16_t *input, double *windowed, int32_t num_samples)
+{
+	int32_t sample_index;
+	for(sample_index = 0; sample_index < num_samples; sample_index++) { 
+		windowed[sample_index] = 0.3635819;
+		windowed[sample_index] -= 0.4891775 * cos(2 * M_PI * sample_index/(num_samples -1));
+		windowed[sample_index] += 0.1365995 * cos(4 * M_PI * sample_index/(num_samples -1));
+		windowed[sample_index] -= 0.0106411 * cos(6 * M_PI * sample_index/(num_samples -1));
+		windowed[sample_index] *= (double)input[sample_index]/32767.0;
+	}
+}
+#elif WINDOW_TYPE == HANNING
+void apply_window(int16_t *input, double *windowed, int32_t num_samples)
+{
+	int32_t sample_index;
+	for(sample_index = 0; sample_index < num_samples; sample_index++) { 
+		windowed[sample_index] = 0.5 - 0.5 * cos(2 * M_PI * sample_index/(num_samples -1));
+		windowed[sample_index] *= (double)input[sample_index]/32767.0;
+	}
+}
+#elif WINDOW_TYPE == BLACKMAN_HARRIS
+void apply_window(int16_t *input, double *windowed, int32_t num_samples)
+{
+	int32_t sample_index;
+	for(sample_index = 0; sample_index < num_samples; sample_index++) { 
+		windowed[sample_index] = 0.35875;
+		windowed[sample_index] -= 0.48829 * cos(2 * M_PI * sample_index/(num_samples -1));
+		windowed[sample_index] += 0.14128 * cos(4 * M_PI * sample_index/(num_samples -1));
+		windowed[sample_index] -= 0.01168 * cos(6 * M_PI * sample_index/(num_samples -1));
+		windowed[sample_index] *= (double)input[sample_index]/32767.0;
+	}
+}
+#endif
+
+typedef struct {
+	double *data_r;
+	double *mag;
+	fftw_complex *data_c;
+} scratch_t;
+
+typedef struct {
+	double max_freq;
+	double max;
+} output_t;
+
+void execute_fft(int16_t * input_buffer, int32_t nfft, int32_t offset,
+		 int32_t num_input_samples, int32_t sample_rate, scratch_t *scratch,
+		 output_t *fft_output);
+int check_boundary(output_t *fft_output, double input_freq, double freq_dev,
+					 double input_vol, double vol_dev);
+
+void execute_fft(int16_t * input_buffer,
+		 int32_t nfft,
+		 int32_t offset,
+		 int32_t num_input_samples,
+		 int32_t sample_rate,
+		 scratch_t *scratch,
+		 output_t *fft_output)
+{
+	fftw_plan p;
+	int32_t max_freq_index = 1, sample_index, block_index;
+	double *data_r = scratch->data_r;
+	double *mag = scratch->mag;
+	double max = -32768.0; /* Initialize to a very low dB */
+	fftw_complex *data_c = scratch->data_c;
+
+	/* This array will contain the average over multiple blocks*/
+	for(sample_index = 0; sample_index < nfft/2+1; sample_index++) {
+		mag[sample_index] = 0;
+	}
+	/* Setup the plan to compute FFT */
+	p = fftw_plan_dft_r2c_1d(nfft, data_r, data_c, FFTW_ESTIMATE);
+	/* Execute FFT */
+	for(block_index = 0; block_index * offset + nfft < num_input_samples; block_index++) {
+		/* Windowing */
+		apply_window(input_buffer + block_index * offset, data_r, nfft);
+		/* FFT */
+		fftw_execute(p);
+		/* Accumulate for averaging. nfft is the normalization factor to
+		 * account for the difference in forward and reverse DFT
+		 */
+		for(sample_index = 0; sample_index < nfft/2+1; sample_index++) {
+			mag[sample_index] += SQUARE(data_c[sample_index][0]/nfft) + SQUARE(data_c[sample_index][1]/nfft);
+		}
+	}
+	/* Find the tone with the highest frequency, ignore the DC bin */
+	for(sample_index = 1; sample_index < nfft/2+1; sample_index++) {
+		mag[sample_index] = 10 * log10(mag[sample_index]/block_index);
+		if (mag[sample_index] > max) {
+			max = mag[sample_index];
+			max_freq_index = sample_index;
+		}
+	}
+	fft_output->max = max;
+	fft_output->max_freq = (float)max_freq_index/nfft * sample_rate;
+	printf("Number of blocks for FFT %d\n", block_index);
+#if 0
+	/* Print the final output */
+	for(sample_index = 0; sample_index < nfft/2+1; sample_index++) {
+		mag[sample_index] = 10 * log10(mag[sample_index]/block_index);
+		printf("%d: %lf(dB)\n", sample_index, mag[sample_index]);
+	}
+#endif
+	fftw_destroy_plan(p);
+}
+
+int check_boundary(output_t *fft_output, double input_freq, double freq_dev,
+					 double input_vol, double vol_dev)
+{
+	double lower_limit, upper_limit;
+
+	/* check if freq is within deviation */
+	lower_limit = input_freq * (1 - freq_dev/100);
+	upper_limit = input_freq * (1 + freq_dev/100);
+	if ( (fft_output->max_freq < lower_limit) || (fft_output->max_freq > upper_limit) )
+		return -1;
+
+	/* Check if volume is within deviation */
+	lower_limit = input_vol * (1 - vol_dev/100);
+	lower_limit = 20 * log10(lower_limit);
+	upper_limit = input_vol * (1 + vol_dev/100);
+	upper_limit = 20 * log10(upper_limit);
+	if ( (fft_output->max < lower_limit) || (fft_output->max > upper_limit) )
+		return -1;
+
+	return 0;
+}
+
+int main(int argc, char **argv)
+{
+	FILE *fin = NULL;
+	int16_t *input_buffer_i = NULL; /* contains interleaved data */
+	int16_t *input_buffer_n = NULL; /* contains one channel of non-interleaved data */
+	wave_header_t hdr;
+	int ret = 0;
+	int32_t nfft, offset, channel_index, num_input_samples, sample_index;
+	scratch_t scratch = {NULL, NULL, NULL};
+	output_t fft_output;
+	double input_freq[MAX_NUM_CHANNELS], freq_dev[MAX_NUM_CHANNELS];
+	double input_vol[MAX_NUM_CHANNELS], vol_dev[MAX_NUM_CHANNELS];
+
+	/* Verify all required arguments are present */
+	if (argc < 8) {
+		printf("Usage: freq_analyzer <num FFT points> <overlap percent> <input file> <input freq> <%% freq deviation> <volume (dB)> <\%% volume deviation>\n");
+		printf("                                                       channel 2:    <input freq> <%% freq deviation> <volume (dB)> <\%% volume deviation>\n");
+		printf("                                                       ...........                                                                        \n");
+		ret = -1;
+		goto _done;
+	}
+	/* Number of FFT points */
+	nfft = atoi(argv[1]);
+	/* overlap */
+	offset = ( (100 - atoi(argv[2])) * nfft) / 100;
+	/* Read the samples from the input file into an array */
+	fin = fopen(argv[3], "rb");
+	if (!fin) {
+		printf("Unable to open input file %s\n", argv[3]);
+		ret = -1;
+		goto _done;
+	}
+	/* input frequency */
+	input_freq[0] = atof(argv[4]);
+	/* Deviation in input frequency */
+	freq_dev[0] = atof(argv[5]);
+	/* Input volume */
+	input_vol[0] = pow(10,atof(argv[6])/20);
+	/* Deviation from input volume */
+	vol_dev[0] = atof(argv[7]);
+	/* Read the header & input samples */
+	fread(&hdr, sizeof(wave_header_t), 1, fin);
+	if (hdr.num_channels > MAX_NUM_CHANNELS) {
+		printf("Only upto stereo supported\n");
+		ret = -1;
+		goto _done;
+	}
+	if ((hdr.num_channels == 2) && (argc < 12) ) {
+		printf("Not enough arguments\n");
+		ret = -1;
+		goto _done;
+	}
+	if (hdr.num_channels == 2) {
+		/* input frequency */
+		input_freq[1] = atof(argv[8]);
+		/* Deviation in input frequency */
+		freq_dev[1] = atof(argv[9]);
+		/* Input volume */
+		input_vol[1] = pow(10,atof(argv[10])/20);
+		/* Deviation from input volume */
+		vol_dev[1] = atof(argv[11]);
+	}
+	num_input_samples = hdr.subchunk2_size/(hdr.num_channels * SAMPLE_BYTES);
+	input_buffer_i = malloc(hdr.subchunk2_size);
+	if (NULL == input_buffer_i) {
+		printf("Unable to allocate space for input buffer\n");
+		ret = -1;
+		goto _done;
+	}
+	fread(input_buffer_i, 1, hdr.subchunk2_size, fin);
+	/* Make necessary memory allocations */
+	scratch.data_r = fftw_alloc_real(nfft);
+	if (NULL == scratch.data_r) {
+		printf("Unable to allocate space for input buffer\n");
+		ret = -1;
+		goto _done;
+	}
+	scratch.data_c = fftw_alloc_complex(nfft/2 + 1);
+	if (NULL == scratch.data_c) {
+		printf("Unable to allocate space for output buffer\n");
+		ret = -1;
+		goto _done;
+	}
+	scratch.mag = malloc(sizeof(double) * (nfft/2 + 1));
+	if (NULL == scratch.mag) {
+		printf("Unable to allocate space for output averaging buffer\n");
+		ret = -1;
+		goto _done;
+	}
+
+	/* FFT */
+	if (1 == hdr.num_channels) {
+		printf("Channel 0: ");
+		/* Mono */
+		execute_fft(input_buffer_i, nfft, offset, num_input_samples, hdr.sample_rate, &scratch, &fft_output);
+		/* Print the output data */
+		printf("%f (hz): %lf(dB)\n", fft_output.max_freq, fft_output.max);
+		if (check_boundary(&fft_output, input_freq[0], freq_dev[0], input_vol[0], vol_dev[0]) == 0)
+			printf("PASS\n");
+		else
+			printf("FAIL\n");
+	}
+	else {
+		/* Processing for more than 1 channel */
+		input_buffer_n = malloc(num_input_samples * SAMPLE_BYTES);
+		if (NULL == input_buffer_n) {
+			printf("Unable to allocate space for deinterleaving buffer\n");
+			ret = -1;
+			goto _done;
+		}
+		for(channel_index =  0; channel_index < hdr.num_channels; channel_index++) {
+			/* Deinterleave and fft for one channel at a time */
+			for(sample_index = 0; sample_index < num_input_samples; sample_index++)
+				input_buffer_n[sample_index] = input_buffer_i[hdr.num_channels * sample_index + channel_index];
+			/* FFT */
+			printf("Channel %d: ", channel_index);
+			execute_fft(input_buffer_n, nfft, offset, num_input_samples, hdr.sample_rate, &scratch, &fft_output);
+			/* Print the output data */
+			printf("%f (hz): %lf(dB)\n", fft_output.max_freq, fft_output.max);
+			if (check_boundary(&fft_output, input_freq[channel_index], freq_dev[channel_index], input_vol[channel_index], vol_dev[channel_index]) == 0)
+				printf("PASS\n");
+			else
+				printf("FAIL\n");
+		}
+	}
+_done:
+	if (fin) fclose(fin);
+	if (input_buffer_i) free(input_buffer_i);
+	if (input_buffer_n) free(input_buffer_n);
+	if (scratch.data_r) fftw_free(scratch.data_r);
+	if (scratch.data_c) fftw_free(scratch.data_c);
+	if (scratch.mag) free(scratch.mag);
+	return ret;
+}
diff --git a/wav_header.h b/wav_header.h
new file mode 100644
index 0000000..4ffda1b
--- /dev/null
+++ b/wav_header.h
@@ -0,0 +1,59 @@
+#ifndef _WAV_HEADER_H_
+#define _WAV_HEADER_H_
+/*
+ *  Format of WAVE (.wav) files.  This is a minimal definition and
+ *  ignores externsions as well as other RIFF variants.
+ *  Based upon ccrma.stanford.edu/courses/422/projects/WaveFormat/
+ *
+ *  Copyright (C) 2015  Nest Labs. All rights reserved.
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License version 2 as
+ *  published by the Free Software Foundation.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+ *
+ */
+
+#include <stdint.h>
+
+#define RIFF_ID			0x46464952	/* 'RIFF' */
+#define WAVE_ID			0x45564157	/* 'WAVE' */
+#define FMT_ID			0x20746d66	/* 'fmt ' */
+#define DATA_ID			0x61746164	/* 'data' */
+
+#define WAVE_FORMAT_PCM		1
+
+typedef struct wave_header_s {
+    uint32_t chunk_id;
+    uint32_t chunk_size;
+    uint32_t format;
+
+    /* "fmt " sub-chunk */
+    uint32_t subchunk1_id;
+    uint32_t subchunk1_size;
+    uint16_t audio_format;
+    uint16_t num_channels;
+    uint32_t sample_rate;
+    uint32_t byte_rate;
+    uint16_t block_align;
+    uint16_t bits_per_sample;
+
+    /* "data" sub-chunk */
+    uint32_t subchunk2_id;
+    uint32_t subchunk2_size;
+    int16_t  data[0];			/* variable size */
+} wave_header_t;
+
+#define WAVE_HDR_SIZE		sizeof(wave_header_t)
+
+#define PCM_CHUNK_SIZE		16	/* XXXehs: redo with offsetof? */
+
+#endif /* _WAV_HEADER_H_ */