modules/audio_processing/transient/transient_suppression_test.cc - src - Git at Google

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */

 #include "modules/audio_processing/transient/transient_suppressor.h"

 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>

 #include <memory>
 #include <string>

 #include "common_audio/include/audio_util.h"
 #include "modules/audio_processing/agc/agc.h"
 #include "rtc_base/flags.h"
 #include "test/gtest.h"
 #include "test/testsupport/fileutils.h"

 WEBRTC_DEFINE_string(in_file_name, "", "PCM file that contains the signal.");
 WEBRTC_DEFINE_string(detection_file_name,
                      "",
                      "PCM file that contains the detection signal.");
 WEBRTC_DEFINE_string(reference_file_name,
                      "",
                      "PCM file that contains the reference signal.");

 WEBRTC_DEFINE_int(chunk_size_ms,
                   10,
                   "Time between each chunk of samples in milliseconds.");

 WEBRTC_DEFINE_int(sample_rate_hz,
                   16000,
                   "Sampling frequency of the signal in Hertz.");
 WEBRTC_DEFINE_int(detection_rate_hz,
                   0,
                   "Sampling frequency of the detection signal in Hertz.");

 WEBRTC_DEFINE_int(num_channels, 1, "Number of channels.");

 WEBRTC_DEFINE_bool(help, false, "Print this message.");

 namespace webrtc {

 const char kUsage[] =
     "\nDetects and suppresses transients from file.\n\n"
     "This application loads the signal from the in_file_name with a specific\n"
     "num_channels and sample_rate_hz, the detection signal from the\n"
     "detection_file_name with a specific detection_rate_hz, and the reference\n"
     "signal from the reference_file_name with sample_rate_hz, divides them\n"
     "into chunk_size_ms blocks, computes its voice value and depending on the\n"
     "voice_threshold does the respective restoration. You can always get the\n"
     "all-voiced or all-unvoiced cases by setting the voice_threshold to 0 or\n"
     "1 respectively.\n\n";

 // Read next buffers from the test files (signed 16-bit host-endian PCM
 // format). audio_buffer has int16 samples, detection_buffer has float samples
 // with range [-32768,32767], and reference_buffer has float samples with range
 // [-1,1]. Return true iff all the buffers were filled completely.
 bool ReadBuffers(FILE* in_file,
                  size_t audio_buffer_size,
                  int num_channels,
                  int16_t* audio_buffer,
                  FILE* detection_file,
                  size_t detection_buffer_size,
                  float* detection_buffer,
                  FILE* reference_file,
                  float* reference_buffer) {
   std::unique_ptr<int16_t[]> tmpbuf;
   int16_t* read_ptr = audio_buffer;
   if (num_channels > 1) {
     tmpbuf.reset(new int16_t[num_channels * audio_buffer_size]);
     read_ptr = tmpbuf.get();
   }
   if (fread(read_ptr, sizeof(*read_ptr), num_channels * audio_buffer_size,
             in_file) != num_channels * audio_buffer_size) {
     return false;
   }
   // De-interleave.
   if (num_channels > 1) {
     for (int i = 0; i < num_channels; ++i) {
       for (size_t j = 0; j < audio_buffer_size; ++j) {
         audio_buffer[i * audio_buffer_size + j] =
             read_ptr[i + j * num_channels];
       }
     }
   }
   if (detection_file) {
     std::unique_ptr<int16_t[]> ibuf(new int16_t[detection_buffer_size]);
     if (fread(ibuf.get(), sizeof(ibuf[0]), detection_buffer_size,
               detection_file) != detection_buffer_size)
       return false;
     for (size_t i = 0; i < detection_buffer_size; ++i)
       detection_buffer[i] = ibuf[i];
   }
   if (reference_file) {
     std::unique_ptr<int16_t[]> ibuf(new int16_t[audio_buffer_size]);
     if (fread(ibuf.get(), sizeof(ibuf[0]), audio_buffer_size, reference_file) !=
         audio_buffer_size)
       return false;
     S16ToFloat(ibuf.get(), audio_buffer_size, reference_buffer);
   }
   return true;
 }

 // Write a number of samples to an open signed 16-bit host-endian PCM file.
 static void WritePCM(FILE* f,
                      size_t num_samples,
                      int num_channels,
                      const float* buffer) {
   std::unique_ptr<int16_t[]> ibuf(new int16_t[num_channels * num_samples]);
   // Interleave.
   for (int i = 0; i < num_channels; ++i) {
     for (size_t j = 0; j < num_samples; ++j) {
       ibuf[i + j * num_channels] = FloatS16ToS16(buffer[i * num_samples + j]);
     }
   }
   fwrite(ibuf.get(), sizeof(ibuf[0]), num_channels * num_samples, f);
 }

 // This application tests the transient suppression by providing a processed
 // PCM file, which has to be listened to in order to evaluate the
 // performance.
 // It gets an audio file, and its voice gain information, and the suppressor
 // process it giving the output file "suppressed_keystrokes.pcm".
 void void_main() {
   // TODO(aluebs): Remove all FileWrappers.
   // Prepare the input file.
   FILE* in_file = fopen(FLAG_in_file_name, "rb");
   ASSERT_TRUE(in_file != NULL);

   // Prepare the detection file.
   FILE* detection_file = NULL;
   if (strlen(FLAG_detection_file_name) > 0) {
     detection_file = fopen(FLAG_detection_file_name, "rb");
   }

   // Prepare the reference file.
   FILE* reference_file = NULL;
   if (strlen(FLAG_reference_file_name) > 0) {
     reference_file = fopen(FLAG_reference_file_name, "rb");
   }

   // Prepare the output file.
   std::string out_file_name = test::OutputPath() + "suppressed_keystrokes.pcm";
   FILE* out_file = fopen(out_file_name.c_str(), "wb");
   ASSERT_TRUE(out_file != NULL);

   int detection_rate_hz = FLAG_detection_rate_hz;
   if (detection_rate_hz == 0) {
     detection_rate_hz = FLAG_sample_rate_hz;
   }

   Agc agc;

   TransientSuppressor suppressor;
   suppressor.Initialize(FLAG_sample_rate_hz, detection_rate_hz,
                         FLAG_num_channels);

   const size_t audio_buffer_size =
       FLAG_chunk_size_ms * FLAG_sample_rate_hz / 1000;
   const size_t detection_buffer_size =
       FLAG_chunk_size_ms * detection_rate_hz / 1000;

   // int16 and float variants of the same data.
   std::unique_ptr<int16_t[]> audio_buffer_i(
       new int16_t[FLAG_num_channels * audio_buffer_size]);
   std::unique_ptr<float[]> audio_buffer_f(
       new float[FLAG_num_channels * audio_buffer_size]);

   std::unique_ptr<float[]> detection_buffer, reference_buffer;

   if (detection_file)
     detection_buffer.reset(new float[detection_buffer_size]);
   if (reference_file)
     reference_buffer.reset(new float[audio_buffer_size]);

   while (ReadBuffers(in_file, audio_buffer_size, FLAG_num_channels,
                      audio_buffer_i.get(), detection_file,
                      detection_buffer_size, detection_buffer.get(),
                      reference_file, reference_buffer.get())) {
     agc.Process(audio_buffer_i.get(), static_cast<int>(audio_buffer_size),
                 FLAG_sample_rate_hz);

     for (size_t i = 0; i < FLAG_num_channels * audio_buffer_size; ++i) {
       audio_buffer_f[i] = audio_buffer_i[i];
     }

     ASSERT_EQ(0, suppressor.Suppress(audio_buffer_f.get(), audio_buffer_size,
                                      FLAG_num_channels, detection_buffer.get(),
                                      detection_buffer_size,
                                      reference_buffer.get(), audio_buffer_size,
                                      agc.voice_probability(), true))
         << "The transient suppressor could not suppress the frame";

     // Write result to out file.
     WritePCM(out_file, audio_buffer_size, FLAG_num_channels,
              audio_buffer_f.get());
   }

   fclose(in_file);
   if (detection_file) {
     fclose(detection_file);
   }
   if (reference_file) {
     fclose(reference_file);
   }
   fclose(out_file);
 }

 }  // namespace webrtc

 int main(int argc, char* argv[]) {
   if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
       argc != 1) {
     printf("%s", webrtc::kUsage);
     if (FLAG_help) {
       rtc::FlagList::Print(nullptr, false);
       return 0;
     }
     return 1;
   }
   RTC_CHECK_GT(FLAG_chunk_size_ms, 0);
   RTC_CHECK_GT(FLAG_sample_rate_hz, 0);
   RTC_CHECK_GT(FLAG_num_channels, 0);

   webrtc::void_main();
   return 0;
 }
	/*
	* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
	*
	* Use of this source code is governed by a BSD-style license
	* that can be found in the LICENSE file in the root of the source
	* tree. An additional intellectual property rights grant can be found
	* in the file PATENTS. All contributing project authors may
	* be found in the AUTHORS file in the root of the source tree.
	*/

	#include "modules/audio_processing/transient/transient_suppressor.h"

	#include <stdio.h>
	#include <stdlib.h>
	#include <string.h>

	#include <memory>
	#include <string>

	#include "common_audio/include/audio_util.h"
	#include "modules/audio_processing/agc/agc.h"
	#include "rtc_base/flags.h"
	#include "test/gtest.h"
	#include "test/testsupport/fileutils.h"

	WEBRTC_DEFINE_string(in_file_name, "", "PCM file that contains the signal.");
	WEBRTC_DEFINE_string(detection_file_name,
	"",
	"PCM file that contains the detection signal.");
	WEBRTC_DEFINE_string(reference_file_name,
	"",
	"PCM file that contains the reference signal.");

	WEBRTC_DEFINE_int(chunk_size_ms,
	10,
	"Time between each chunk of samples in milliseconds.");

	WEBRTC_DEFINE_int(sample_rate_hz,
	16000,
	"Sampling frequency of the signal in Hertz.");
	WEBRTC_DEFINE_int(detection_rate_hz,
	0,
	"Sampling frequency of the detection signal in Hertz.");

	WEBRTC_DEFINE_int(num_channels, 1, "Number of channels.");

	WEBRTC_DEFINE_bool(help, false, "Print this message.");

	namespace webrtc {

	const char kUsage[] =
	"\nDetects and suppresses transients from file.\n\n"
	"This application loads the signal from the in_file_name with a specific\n"
	"num_channels and sample_rate_hz, the detection signal from the\n"
	"detection_file_name with a specific detection_rate_hz, and the reference\n"
	"signal from the reference_file_name with sample_rate_hz, divides them\n"
	"into chunk_size_ms blocks, computes its voice value and depending on the\n"
	"voice_threshold does the respective restoration. You can always get the\n"
	"all-voiced or all-unvoiced cases by setting the voice_threshold to 0 or\n"
	"1 respectively.\n\n";

	// Read next buffers from the test files (signed 16-bit host-endian PCM
	// format). audio_buffer has int16 samples, detection_buffer has float samples
	// with range [-32768,32767], and reference_buffer has float samples with range
	// [-1,1]. Return true iff all the buffers were filled completely.
	bool ReadBuffers(FILE* in_file,
	size_t audio_buffer_size,
	int num_channels,
	int16_t* audio_buffer,
	FILE* detection_file,
	size_t detection_buffer_size,
	float* detection_buffer,
	FILE* reference_file,
	float* reference_buffer) {
	std::unique_ptr<int16_t[]> tmpbuf;
	int16_t* read_ptr = audio_buffer;
	if (num_channels > 1) {
	tmpbuf.reset(new int16_t[num_channels * audio_buffer_size]);
	read_ptr = tmpbuf.get();
	}
	if (fread(read_ptr, sizeof(read_ptr), num_channels audio_buffer_size,
	in_file) != num_channels * audio_buffer_size) {
	return false;
	}
	// De-interleave.
	if (num_channels > 1) {
	for (int i = 0; i < num_channels; ++i) {
	for (size_t j = 0; j < audio_buffer_size; ++j) {
	audio_buffer[i * audio_buffer_size + j] =
	read_ptr[i + j * num_channels];
	}
	}
	}
	if (detection_file) {
	std::unique_ptr<int16_t[]> ibuf(new int16_t[detection_buffer_size]);
	if (fread(ibuf.get(), sizeof(ibuf[0]), detection_buffer_size,
	detection_file) != detection_buffer_size)
	return false;
	for (size_t i = 0; i < detection_buffer_size; ++i)
	detection_buffer[i] = ibuf[i];
	}
	if (reference_file) {
	std::unique_ptr<int16_t[]> ibuf(new int16_t[audio_buffer_size]);
	if (fread(ibuf.get(), sizeof(ibuf[0]), audio_buffer_size, reference_file) !=
	audio_buffer_size)
	return false;
	S16ToFloat(ibuf.get(), audio_buffer_size, reference_buffer);
	}
	return true;
	}

	// Write a number of samples to an open signed 16-bit host-endian PCM file.
	static void WritePCM(FILE* f,
	size_t num_samples,
	int num_channels,
	const float* buffer) {
	std::unique_ptr<int16_t[]> ibuf(new int16_t[num_channels * num_samples]);
	// Interleave.
	for (int i = 0; i < num_channels; ++i) {
	for (size_t j = 0; j < num_samples; ++j) {
	ibuf[i + j * num_channels] = FloatS16ToS16(buffer[i * num_samples + j]);
	}
	}
	fwrite(ibuf.get(), sizeof(ibuf[0]), num_channels * num_samples, f);
	}

	// This application tests the transient suppression by providing a processed
	// PCM file, which has to be listened to in order to evaluate the
	// performance.
	// It gets an audio file, and its voice gain information, and the suppressor
	// process it giving the output file "suppressed_keystrokes.pcm".
	void void_main() {
	// TODO(aluebs): Remove all FileWrappers.
	// Prepare the input file.
	FILE* in_file = fopen(FLAG_in_file_name, "rb");
	ASSERT_TRUE(in_file != NULL);

	// Prepare the detection file.
	FILE* detection_file = NULL;
	if (strlen(FLAG_detection_file_name) > 0) {
	detection_file = fopen(FLAG_detection_file_name, "rb");
	}

	// Prepare the reference file.
	FILE* reference_file = NULL;
	if (strlen(FLAG_reference_file_name) > 0) {
	reference_file = fopen(FLAG_reference_file_name, "rb");
	}

	// Prepare the output file.
	std::string out_file_name = test::OutputPath() + "suppressed_keystrokes.pcm";
	FILE* out_file = fopen(out_file_name.c_str(), "wb");
	ASSERT_TRUE(out_file != NULL);

	int detection_rate_hz = FLAG_detection_rate_hz;
	if (detection_rate_hz == 0) {
	detection_rate_hz = FLAG_sample_rate_hz;
	}

	Agc agc;

	TransientSuppressor suppressor;
	suppressor.Initialize(FLAG_sample_rate_hz, detection_rate_hz,
	FLAG_num_channels);

	const size_t audio_buffer_size =
	FLAG_chunk_size_ms * FLAG_sample_rate_hz / 1000;
	const size_t detection_buffer_size =
	FLAG_chunk_size_ms * detection_rate_hz / 1000;

	// int16 and float variants of the same data.
	std::unique_ptr<int16_t[]> audio_buffer_i(
	new int16_t[FLAG_num_channels * audio_buffer_size]);
	std::unique_ptr<float[]> audio_buffer_f(
	new float[FLAG_num_channels * audio_buffer_size]);

	std::unique_ptr<float[]> detection_buffer, reference_buffer;

	if (detection_file)
	detection_buffer.reset(new float[detection_buffer_size]);
	if (reference_file)
	reference_buffer.reset(new float[audio_buffer_size]);

	while (ReadBuffers(in_file, audio_buffer_size, FLAG_num_channels,
	audio_buffer_i.get(), detection_file,
	detection_buffer_size, detection_buffer.get(),
	reference_file, reference_buffer.get())) {
	agc.Process(audio_buffer_i.get(), static_cast<int>(audio_buffer_size),
	FLAG_sample_rate_hz);

	for (size_t i = 0; i < FLAG_num_channels * audio_buffer_size; ++i) {
	audio_buffer_f[i] = audio_buffer_i[i];
	}

	ASSERT_EQ(0, suppressor.Suppress(audio_buffer_f.get(), audio_buffer_size,
	FLAG_num_channels, detection_buffer.get(),
	detection_buffer_size,
	reference_buffer.get(), audio_buffer_size,
	agc.voice_probability(), true))
	<< "The transient suppressor could not suppress the frame";

	// Write result to out file.
	WritePCM(out_file, audio_buffer_size, FLAG_num_channels,
	audio_buffer_f.get());
	}

	fclose(in_file);
	if (detection_file) {
	fclose(detection_file);
	}
	if (reference_file) {
	fclose(reference_file);
	}
	fclose(out_file);
	}

	} // namespace webrtc

	int main(int argc, char* argv[]) {
	if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) \|\| FLAG_help \|\|
	argc != 1) {
	printf("%s", webrtc::kUsage);
	if (FLAG_help) {
	rtc::FlagList::Print(nullptr, false);
	return 0;
	}
	return 1;
	}
	RTC_CHECK_GT(FLAG_chunk_size_ms, 0);
	RTC_CHECK_GT(FLAG_sample_rate_hz, 0);
	RTC_CHECK_GT(FLAG_num_channels, 0);

	webrtc::void_main();
	return 0;
	}