modules/audio_processing/test/unpack.cc - src/webrtc - Git at Google

 /*
  *  Copyright (c) 2011 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.
  */

 // Commandline tool to unpack audioproc debug files.
 //
 // The debug files are dumped as protobuf blobs. For analysis, it's necessary
 // to unpack the file into its component parts: audio and other data.

 #include <stdio.h>

 #include <memory>

 #include "gflags/gflags.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/rtc_base/format_macros.h"
 #include "webrtc/rtc_base/ignore_wundef.h"
 #include "webrtc/typedefs.h"

 RTC_PUSH_IGNORING_WUNDEF()
 #include "webrtc/modules/audio_processing/debug.pb.h"
 RTC_POP_IGNORING_WUNDEF()

 // TODO(andrew): unpack more of the data.
 DEFINE_string(input_file, "input", "The name of the input stream file.");
 DEFINE_string(output_file, "ref_out",
               "The name of the reference output stream file.");
 DEFINE_string(reverse_file, "reverse",
               "The name of the reverse input stream file.");
 DEFINE_string(delay_file, "delay.int32", "The name of the delay file.");
 DEFINE_string(drift_file, "drift.int32", "The name of the drift file.");
 DEFINE_string(level_file, "level.int32", "The name of the level file.");
 DEFINE_string(keypress_file, "keypress.bool", "The name of the keypress file.");
 DEFINE_string(settings_file, "settings.txt", "The name of the settings file.");
 DEFINE_bool(full, false,
             "Unpack the full set of files (normally not needed).");
 DEFINE_bool(raw, false, "Write raw data instead of a WAV file.");
 DEFINE_bool(text,
             false,
             "Write non-audio files as text files instead of binary files.");

 #define PRINT_CONFIG(field_name) \
   if (msg.has_##field_name()) { \
     fprintf(settings_file, "  " #field_name ": %d\n", msg.field_name()); \
   }

 namespace webrtc {

 using audioproc::Event;
 using audioproc::ReverseStream;
 using audioproc::Stream;
 using audioproc::Init;

 void WriteData(const void* data, size_t size, FILE* file,
                const std::string& filename) {
   if (fwrite(data, size, 1, file) != 1) {
     printf("Error when writing to %s\n", filename.c_str());
     exit(1);
   }
 }

 int do_main(int argc, char* argv[]) {
   std::string program_name = argv[0];
   std::string usage = "Commandline tool to unpack audioproc debug files.\n"
     "Example usage:\n" + program_name + " debug_dump.pb\n";
   google::SetUsageMessage(usage);
   google::ParseCommandLineFlags(&argc, &argv, true);

   if (argc < 2) {
     printf("%s", google::ProgramUsage());
     return 1;
   }

   FILE* debug_file = OpenFile(argv[1], "rb");

   Event event_msg;
   int frame_count = 0;
   size_t reverse_samples_per_channel = 0;
   size_t input_samples_per_channel = 0;
   size_t output_samples_per_channel = 0;
   size_t num_reverse_channels = 0;
   size_t num_input_channels = 0;
   size_t num_output_channels = 0;
   std::unique_ptr<WavWriter> reverse_wav_file;
   std::unique_ptr<WavWriter> input_wav_file;
   std::unique_ptr<WavWriter> output_wav_file;
   std::unique_ptr<RawFile> reverse_raw_file;
   std::unique_ptr<RawFile> input_raw_file;
   std::unique_ptr<RawFile> output_raw_file;

   FILE* settings_file = OpenFile(FLAGS_settings_file, "wb");

   while (ReadMessageFromFile(debug_file, &event_msg)) {
     if (event_msg.type() == Event::REVERSE_STREAM) {
       if (!event_msg.has_reverse_stream()) {
         printf("Corrupt input file: ReverseStream missing.\n");
         return 1;
       }

       const ReverseStream msg = event_msg.reverse_stream();
       if (msg.has_data()) {
         if (FLAGS_raw && !reverse_raw_file) {
           reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".pcm"));
         }
         // TODO(aluebs): Replace "num_reverse_channels *
         // reverse_samples_per_channel" with "msg.data().size() /
         // sizeof(int16_t)" and so on when this fix in audio_processing has made
         // it into stable: https://webrtc-codereview.appspot.com/15299004/
         WriteIntData(reinterpret_cast<const int16_t*>(msg.data().data()),
                      num_reverse_channels * reverse_samples_per_channel,
                      reverse_wav_file.get(),
                      reverse_raw_file.get());
       } else if (msg.channel_size() > 0) {
         if (FLAGS_raw && !reverse_raw_file) {
           reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".float"));
         }
         std::unique_ptr<const float* []> data(
             new const float* [num_reverse_channels]);
         for (size_t i = 0; i < num_reverse_channels; ++i) {
           data[i] = reinterpret_cast<const float*>(msg.channel(i).data());
         }
         WriteFloatData(data.get(),
                        reverse_samples_per_channel,
                        num_reverse_channels,
                        reverse_wav_file.get(),
                        reverse_raw_file.get());
       }
     } else if (event_msg.type() == Event::STREAM) {
       frame_count++;
       if (!event_msg.has_stream()) {
         printf("Corrupt input file: Stream missing.\n");
         return 1;
       }

       const Stream msg = event_msg.stream();
       if (msg.has_input_data()) {
         if (FLAGS_raw && !input_raw_file) {
           input_raw_file.reset(new RawFile(FLAGS_input_file + ".pcm"));
         }
         WriteIntData(reinterpret_cast<const int16_t*>(msg.input_data().data()),
                      num_input_channels * input_samples_per_channel,
                      input_wav_file.get(),
                      input_raw_file.get());
       } else if (msg.input_channel_size() > 0) {
         if (FLAGS_raw && !input_raw_file) {
           input_raw_file.reset(new RawFile(FLAGS_input_file + ".float"));
         }
         std::unique_ptr<const float* []> data(
             new const float* [num_input_channels]);
         for (size_t i = 0; i < num_input_channels; ++i) {
           data[i] = reinterpret_cast<const float*>(msg.input_channel(i).data());
         }
         WriteFloatData(data.get(),
                        input_samples_per_channel,
                        num_input_channels,
                        input_wav_file.get(),
                        input_raw_file.get());
       }

       if (msg.has_output_data()) {
         if (FLAGS_raw && !output_raw_file) {
           output_raw_file.reset(new RawFile(FLAGS_output_file + ".pcm"));
         }
         WriteIntData(reinterpret_cast<const int16_t*>(msg.output_data().data()),
                      num_output_channels * output_samples_per_channel,
                      output_wav_file.get(),
                      output_raw_file.get());
       } else if (msg.output_channel_size() > 0) {
         if (FLAGS_raw && !output_raw_file) {
           output_raw_file.reset(new RawFile(FLAGS_output_file + ".float"));
         }
         std::unique_ptr<const float* []> data(
             new const float* [num_output_channels]);
         for (size_t i = 0; i < num_output_channels; ++i) {
           data[i] =
               reinterpret_cast<const float*>(msg.output_channel(i).data());
         }
         WriteFloatData(data.get(),
                        output_samples_per_channel,
                        num_output_channels,
                        output_wav_file.get(),
                        output_raw_file.get());
       }

       if (FLAGS_full) {
         if (msg.has_delay()) {
           static FILE* delay_file = OpenFile(FLAGS_delay_file, "wb");
           int32_t delay = msg.delay();
           if (FLAGS_text) {
             fprintf(delay_file, "%d\n", delay);
           } else {
             WriteData(&delay, sizeof(delay), delay_file, FLAGS_delay_file);
           }
         }

         if (msg.has_drift()) {
           static FILE* drift_file = OpenFile(FLAGS_drift_file, "wb");
           int32_t drift = msg.drift();
           if (FLAGS_text) {
             fprintf(drift_file, "%d\n", drift);
           } else {
             WriteData(&drift, sizeof(drift), drift_file, FLAGS_drift_file);
           }
         }

         if (msg.has_level()) {
           static FILE* level_file = OpenFile(FLAGS_level_file, "wb");
           int32_t level = msg.level();
           if (FLAGS_text) {
             fprintf(level_file, "%d\n", level);
           } else {
             WriteData(&level, sizeof(level), level_file, FLAGS_level_file);
           }
         }

         if (msg.has_keypress()) {
           static FILE* keypress_file = OpenFile(FLAGS_keypress_file, "wb");
           bool keypress = msg.keypress();
           if (FLAGS_text) {
             fprintf(keypress_file, "%d\n", keypress);
           } else {
             WriteData(&keypress, sizeof(keypress), keypress_file,
                       FLAGS_keypress_file);
           }
         }
       }
     } else if (event_msg.type() == Event::CONFIG) {
       if (!event_msg.has_config()) {
         printf("Corrupt input file: Config missing.\n");
         return 1;
       }
       const audioproc::Config msg = event_msg.config();

       fprintf(settings_file, "APM re-config at frame: %d\n", frame_count);

       PRINT_CONFIG(aec_enabled);
       PRINT_CONFIG(aec_delay_agnostic_enabled);
       PRINT_CONFIG(aec_drift_compensation_enabled);
       PRINT_CONFIG(aec_extended_filter_enabled);
       PRINT_CONFIG(aec_suppression_level);
       PRINT_CONFIG(aecm_enabled);
       PRINT_CONFIG(aecm_comfort_noise_enabled);
       PRINT_CONFIG(aecm_routing_mode);
       PRINT_CONFIG(agc_enabled);
       PRINT_CONFIG(agc_mode);
       PRINT_CONFIG(agc_limiter_enabled);
       PRINT_CONFIG(noise_robust_agc_enabled);
       PRINT_CONFIG(hpf_enabled);
       PRINT_CONFIG(ns_enabled);
       PRINT_CONFIG(ns_level);
       PRINT_CONFIG(transient_suppression_enabled);
       PRINT_CONFIG(intelligibility_enhancer_enabled);
       if (msg.has_experiments_description()) {
         fprintf(settings_file, "  experiments_description: %s\n",
                 msg.experiments_description().c_str());
       }
     } else if (event_msg.type() == Event::INIT) {
       if (!event_msg.has_init()) {
         printf("Corrupt input file: Init missing.\n");
         return 1;
       }

       const Init msg = event_msg.init();
       // These should print out zeros if they're missing.
       fprintf(settings_file, "Init at frame: %d\n", frame_count);
       int input_sample_rate = msg.sample_rate();
       fprintf(settings_file, "  Input sample rate: %d\n", input_sample_rate);
       int output_sample_rate = msg.output_sample_rate();
       fprintf(settings_file, "  Output sample rate: %d\n", output_sample_rate);
       int reverse_sample_rate = msg.reverse_sample_rate();
       fprintf(settings_file,
               "  Reverse sample rate: %d\n",
               reverse_sample_rate);
       num_input_channels = msg.num_input_channels();
       fprintf(settings_file, "  Input channels: %" PRIuS "\n",
               num_input_channels);
       num_output_channels = msg.num_output_channels();
       fprintf(settings_file, "  Output channels: %" PRIuS "\n",
               num_output_channels);
       num_reverse_channels = msg.num_reverse_channels();
       fprintf(settings_file, "  Reverse channels: %" PRIuS "\n",
               num_reverse_channels);

       fprintf(settings_file, "\n");

       if (reverse_sample_rate == 0) {
         reverse_sample_rate = input_sample_rate;
       }
       if (output_sample_rate == 0) {
         output_sample_rate = input_sample_rate;
       }

       reverse_samples_per_channel =
           static_cast<size_t>(reverse_sample_rate / 100);
       input_samples_per_channel =
           static_cast<size_t>(input_sample_rate / 100);
       output_samples_per_channel =
           static_cast<size_t>(output_sample_rate / 100);

       if (!FLAGS_raw) {
         // The WAV files need to be reset every time, because they cant change
         // their sample rate or number of channels.
         std::stringstream reverse_name;
         reverse_name << FLAGS_reverse_file << frame_count << ".wav";
         reverse_wav_file.reset(new WavWriter(reverse_name.str(),
                                              reverse_sample_rate,
                                              num_reverse_channels));
         std::stringstream input_name;
         input_name << FLAGS_input_file << frame_count << ".wav";
         input_wav_file.reset(new WavWriter(input_name.str(),
                                            input_sample_rate,
                                            num_input_channels));
         std::stringstream output_name;
         output_name << FLAGS_output_file << frame_count << ".wav";
         output_wav_file.reset(new WavWriter(output_name.str(),
                                             output_sample_rate,
                                             num_output_channels));
       }
     }
   }

   return 0;
 }

 }  // namespace webrtc

 int main(int argc, char* argv[]) {
   return webrtc::do_main(argc, argv);
 }
	/*
	* Copyright (c) 2011 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.
	*/

	// Commandline tool to unpack audioproc debug files.
	//
	// The debug files are dumped as protobuf blobs. For analysis, it's necessary
	// to unpack the file into its component parts: audio and other data.

	#include <stdio.h>

	#include <memory>

	#include "gflags/gflags.h"
	#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
	#include "webrtc/modules/audio_processing/test/test_utils.h"
	#include "webrtc/rtc_base/format_macros.h"
	#include "webrtc/rtc_base/ignore_wundef.h"
	#include "webrtc/typedefs.h"

	RTC_PUSH_IGNORING_WUNDEF()
	#include "webrtc/modules/audio_processing/debug.pb.h"
	RTC_POP_IGNORING_WUNDEF()

	// TODO(andrew): unpack more of the data.
	DEFINE_string(input_file, "input", "The name of the input stream file.");
	DEFINE_string(output_file, "ref_out",
	"The name of the reference output stream file.");
	DEFINE_string(reverse_file, "reverse",
	"The name of the reverse input stream file.");
	DEFINE_string(delay_file, "delay.int32", "The name of the delay file.");
	DEFINE_string(drift_file, "drift.int32", "The name of the drift file.");
	DEFINE_string(level_file, "level.int32", "The name of the level file.");
	DEFINE_string(keypress_file, "keypress.bool", "The name of the keypress file.");
	DEFINE_string(settings_file, "settings.txt", "The name of the settings file.");
	DEFINE_bool(full, false,
	"Unpack the full set of files (normally not needed).");
	DEFINE_bool(raw, false, "Write raw data instead of a WAV file.");
	DEFINE_bool(text,
	false,
	"Write non-audio files as text files instead of binary files.");

	#define PRINT_CONFIG(field_name) \
	if (msg.has_##field_name()) { \
	fprintf(settings_file, " " #field_name ": %d\n", msg.field_name()); \
	}

	namespace webrtc {

	using audioproc::Event;
	using audioproc::ReverseStream;
	using audioproc::Stream;
	using audioproc::Init;

	void WriteData(const void* data, size_t size, FILE* file,
	const std::string& filename) {
	if (fwrite(data, size, 1, file) != 1) {
	printf("Error when writing to %s\n", filename.c_str());
	exit(1);
	}
	}

	int do_main(int argc, char* argv[]) {
	std::string program_name = argv[0];
	std::string usage = "Commandline tool to unpack audioproc debug files.\n"
	"Example usage:\n" + program_name + " debug_dump.pb\n";
	google::SetUsageMessage(usage);
	google::ParseCommandLineFlags(&argc, &argv, true);

	if (argc < 2) {
	printf("%s", google::ProgramUsage());
	return 1;
	}

	FILE* debug_file = OpenFile(argv[1], "rb");

	Event event_msg;
	int frame_count = 0;
	size_t reverse_samples_per_channel = 0;
	size_t input_samples_per_channel = 0;
	size_t output_samples_per_channel = 0;
	size_t num_reverse_channels = 0;
	size_t num_input_channels = 0;
	size_t num_output_channels = 0;
	std::unique_ptr<WavWriter> reverse_wav_file;
	std::unique_ptr<WavWriter> input_wav_file;
	std::unique_ptr<WavWriter> output_wav_file;
	std::unique_ptr<RawFile> reverse_raw_file;
	std::unique_ptr<RawFile> input_raw_file;
	std::unique_ptr<RawFile> output_raw_file;

	FILE* settings_file = OpenFile(FLAGS_settings_file, "wb");

	while (ReadMessageFromFile(debug_file, &event_msg)) {
	if (event_msg.type() == Event::REVERSE_STREAM) {
	if (!event_msg.has_reverse_stream()) {
	printf("Corrupt input file: ReverseStream missing.\n");
	return 1;
	}

	const ReverseStream msg = event_msg.reverse_stream();
	if (msg.has_data()) {
	if (FLAGS_raw && !reverse_raw_file) {
	reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".pcm"));
	}
	// TODO(aluebs): Replace "num_reverse_channels *
	// reverse_samples_per_channel" with "msg.data().size() /
	// sizeof(int16_t)" and so on when this fix in audio_processing has made
	// it into stable: https://webrtc-codereview.appspot.com/15299004/
	WriteIntData(reinterpret_cast<const int16_t*>(msg.data().data()),
	num_reverse_channels * reverse_samples_per_channel,
	reverse_wav_file.get(),
	reverse_raw_file.get());
	} else if (msg.channel_size() > 0) {
	if (FLAGS_raw && !reverse_raw_file) {
	reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".float"));
	}
	std::unique_ptr<const float* []> data(
	new const float* [num_reverse_channels]);
	for (size_t i = 0; i < num_reverse_channels; ++i) {
	data[i] = reinterpret_cast<const float*>(msg.channel(i).data());
	}
	WriteFloatData(data.get(),
	reverse_samples_per_channel,
	num_reverse_channels,
	reverse_wav_file.get(),
	reverse_raw_file.get());
	}
	} else if (event_msg.type() == Event::STREAM) {
	frame_count++;
	if (!event_msg.has_stream()) {
	printf("Corrupt input file: Stream missing.\n");
	return 1;
	}

	const Stream msg = event_msg.stream();
	if (msg.has_input_data()) {
	if (FLAGS_raw && !input_raw_file) {
	input_raw_file.reset(new RawFile(FLAGS_input_file + ".pcm"));
	}
	WriteIntData(reinterpret_cast<const int16_t*>(msg.input_data().data()),
	num_input_channels * input_samples_per_channel,
	input_wav_file.get(),
	input_raw_file.get());
	} else if (msg.input_channel_size() > 0) {
	if (FLAGS_raw && !input_raw_file) {
	input_raw_file.reset(new RawFile(FLAGS_input_file + ".float"));
	}
	std::unique_ptr<const float* []> data(
	new const float* [num_input_channels]);
	for (size_t i = 0; i < num_input_channels; ++i) {
	data[i] = reinterpret_cast<const float*>(msg.input_channel(i).data());
	}
	WriteFloatData(data.get(),
	input_samples_per_channel,
	num_input_channels,
	input_wav_file.get(),
	input_raw_file.get());
	}

	if (msg.has_output_data()) {
	if (FLAGS_raw && !output_raw_file) {
	output_raw_file.reset(new RawFile(FLAGS_output_file + ".pcm"));
	}
	WriteIntData(reinterpret_cast<const int16_t*>(msg.output_data().data()),
	num_output_channels * output_samples_per_channel,
	output_wav_file.get(),
	output_raw_file.get());
	} else if (msg.output_channel_size() > 0) {
	if (FLAGS_raw && !output_raw_file) {
	output_raw_file.reset(new RawFile(FLAGS_output_file + ".float"));
	}
	std::unique_ptr<const float* []> data(
	new const float* [num_output_channels]);
	for (size_t i = 0; i < num_output_channels; ++i) {
	data[i] =
	reinterpret_cast<const float*>(msg.output_channel(i).data());
	}
	WriteFloatData(data.get(),
	output_samples_per_channel,
	num_output_channels,
	output_wav_file.get(),
	output_raw_file.get());
	}

	if (FLAGS_full) {
	if (msg.has_delay()) {
	static FILE* delay_file = OpenFile(FLAGS_delay_file, "wb");
	int32_t delay = msg.delay();
	if (FLAGS_text) {
	fprintf(delay_file, "%d\n", delay);
	} else {
	WriteData(&delay, sizeof(delay), delay_file, FLAGS_delay_file);
	}
	}

	if (msg.has_drift()) {
	static FILE* drift_file = OpenFile(FLAGS_drift_file, "wb");
	int32_t drift = msg.drift();
	if (FLAGS_text) {
	fprintf(drift_file, "%d\n", drift);
	} else {
	WriteData(&drift, sizeof(drift), drift_file, FLAGS_drift_file);
	}
	}

	if (msg.has_level()) {
	static FILE* level_file = OpenFile(FLAGS_level_file, "wb");
	int32_t level = msg.level();
	if (FLAGS_text) {
	fprintf(level_file, "%d\n", level);
	} else {
	WriteData(&level, sizeof(level), level_file, FLAGS_level_file);
	}
	}

	if (msg.has_keypress()) {
	static FILE* keypress_file = OpenFile(FLAGS_keypress_file, "wb");
	bool keypress = msg.keypress();
	if (FLAGS_text) {
	fprintf(keypress_file, "%d\n", keypress);
	} else {
	WriteData(&keypress, sizeof(keypress), keypress_file,
	FLAGS_keypress_file);
	}
	}
	}
	} else if (event_msg.type() == Event::CONFIG) {
	if (!event_msg.has_config()) {
	printf("Corrupt input file: Config missing.\n");
	return 1;
	}
	const audioproc::Config msg = event_msg.config();

	fprintf(settings_file, "APM re-config at frame: %d\n", frame_count);

	PRINT_CONFIG(aec_enabled);
	PRINT_CONFIG(aec_delay_agnostic_enabled);
	PRINT_CONFIG(aec_drift_compensation_enabled);
	PRINT_CONFIG(aec_extended_filter_enabled);
	PRINT_CONFIG(aec_suppression_level);
	PRINT_CONFIG(aecm_enabled);
	PRINT_CONFIG(aecm_comfort_noise_enabled);
	PRINT_CONFIG(aecm_routing_mode);
	PRINT_CONFIG(agc_enabled);
	PRINT_CONFIG(agc_mode);
	PRINT_CONFIG(agc_limiter_enabled);
	PRINT_CONFIG(noise_robust_agc_enabled);
	PRINT_CONFIG(hpf_enabled);
	PRINT_CONFIG(ns_enabled);
	PRINT_CONFIG(ns_level);
	PRINT_CONFIG(transient_suppression_enabled);
	PRINT_CONFIG(intelligibility_enhancer_enabled);
	if (msg.has_experiments_description()) {
	fprintf(settings_file, " experiments_description: %s\n",
	msg.experiments_description().c_str());
	}
	} else if (event_msg.type() == Event::INIT) {
	if (!event_msg.has_init()) {
	printf("Corrupt input file: Init missing.\n");
	return 1;
	}

	const Init msg = event_msg.init();
	// These should print out zeros if they're missing.
	fprintf(settings_file, "Init at frame: %d\n", frame_count);
	int input_sample_rate = msg.sample_rate();
	fprintf(settings_file, " Input sample rate: %d\n", input_sample_rate);
	int output_sample_rate = msg.output_sample_rate();
	fprintf(settings_file, " Output sample rate: %d\n", output_sample_rate);
	int reverse_sample_rate = msg.reverse_sample_rate();
	fprintf(settings_file,
	" Reverse sample rate: %d\n",
	reverse_sample_rate);
	num_input_channels = msg.num_input_channels();
	fprintf(settings_file, " Input channels: %" PRIuS "\n",
	num_input_channels);
	num_output_channels = msg.num_output_channels();
	fprintf(settings_file, " Output channels: %" PRIuS "\n",
	num_output_channels);
	num_reverse_channels = msg.num_reverse_channels();
	fprintf(settings_file, " Reverse channels: %" PRIuS "\n",
	num_reverse_channels);

	fprintf(settings_file, "\n");

	if (reverse_sample_rate == 0) {
	reverse_sample_rate = input_sample_rate;
	}
	if (output_sample_rate == 0) {
	output_sample_rate = input_sample_rate;
	}

	reverse_samples_per_channel =
	static_cast<size_t>(reverse_sample_rate / 100);
	input_samples_per_channel =
	static_cast<size_t>(input_sample_rate / 100);
	output_samples_per_channel =
	static_cast<size_t>(output_sample_rate / 100);

	if (!FLAGS_raw) {
	// The WAV files need to be reset every time, because they cant change
	// their sample rate or number of channels.
	std::stringstream reverse_name;
	reverse_name << FLAGS_reverse_file << frame_count << ".wav";
	reverse_wav_file.reset(new WavWriter(reverse_name.str(),
	reverse_sample_rate,
	num_reverse_channels));
	std::stringstream input_name;
	input_name << FLAGS_input_file << frame_count << ".wav";
	input_wav_file.reset(new WavWriter(input_name.str(),
	input_sample_rate,
	num_input_channels));
	std::stringstream output_name;
	output_name << FLAGS_output_file << frame_count << ".wav";
	output_wav_file.reset(new WavWriter(output_name.str(),
	output_sample_rate,
	num_output_channels));
	}
	}
	}

	return 0;
	}

	} // namespace webrtc

	int main(int argc, char* argv[]) {
	return webrtc::do_main(argc, argv);
	}