blob: 142b49730a36170b6576174a5b731228fd657ba3 [file] [log] [blame]
/*
* 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 "modules/audio_processing/test/protobuf_utils.h"
#include "modules/audio_processing/test/test_utils.h"
#include "rtc_base/flags.h"
#include "rtc_base/format_macros.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/strings/string_builder.h"
RTC_PUSH_IGNORING_WUNDEF()
#include "modules/audio_processing/debug.pb.h"
RTC_POP_IGNORING_WUNDEF()
// TODO(andrew): unpack more of the data.
WEBRTC_DEFINE_string(input_file, "input", "The name of the input stream file.");
WEBRTC_DEFINE_string(output_file,
"ref_out",
"The name of the reference output stream file.");
WEBRTC_DEFINE_string(reverse_file,
"reverse",
"The name of the reverse input stream file.");
WEBRTC_DEFINE_string(delay_file, "delay.int32", "The name of the delay file.");
WEBRTC_DEFINE_string(drift_file, "drift.int32", "The name of the drift file.");
WEBRTC_DEFINE_string(level_file, "level.int32", "The name of the level file.");
WEBRTC_DEFINE_string(keypress_file,
"keypress.bool",
"The name of the keypress file.");
WEBRTC_DEFINE_string(callorder_file,
"callorder",
"The name of the render/capture call order file.");
WEBRTC_DEFINE_string(settings_file,
"settings.txt",
"The name of the settings file.");
WEBRTC_DEFINE_bool(full,
false,
"Unpack the full set of files (normally not needed).");
WEBRTC_DEFINE_bool(raw, false, "Write raw data instead of a WAV file.");
WEBRTC_DEFINE_bool(
text,
false,
"Write non-audio files as text files instead of binary files.");
WEBRTC_DEFINE_bool(help, false, "Print this message.");
#define PRINT_CONFIG(field_name) \
if (msg.has_##field_name()) { \
fprintf(settings_file, " " #field_name ": %d\n", msg.field_name()); \
}
#define PRINT_CONFIG_FLOAT(field_name) \
if (msg.has_##field_name()) { \
fprintf(settings_file, " " #field_name ": %f\n", msg.field_name()); \
}
namespace webrtc {
using audioproc::Event;
using audioproc::ReverseStream;
using audioproc::Stream;
using audioproc::Init;
namespace {
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);
}
}
void WriteCallOrderData(const bool render_call,
FILE* file,
const std::string& filename) {
const char call_type = render_call ? 'r' : 'c';
WriteData(&call_type, sizeof(call_type), file, filename.c_str());
}
bool WritingCallOrderFile() {
return FLAG_full;
}
} // namespace
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";
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
argc < 2) {
printf("%s", usage.c_str());
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
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;
rtc::StringBuilder callorder_raw_name;
callorder_raw_name << FLAG_callorder_file << ".char";
FILE* callorder_char_file = WritingCallOrderFile()
? OpenFile(callorder_raw_name.str(), "wb")
: nullptr;
FILE* settings_file = OpenFile(FLAG_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 (FLAG_raw && !reverse_raw_file) {
reverse_raw_file.reset(
new RawFile(std::string(FLAG_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 (FLAG_raw && !reverse_raw_file) {
reverse_raw_file.reset(
new RawFile(std::string(FLAG_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());
}
if (FLAG_full) {
if (WritingCallOrderFile()) {
WriteCallOrderData(true /* render_call */, callorder_char_file,
FLAG_callorder_file);
}
}
} 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 (FLAG_raw && !input_raw_file) {
input_raw_file.reset(
new RawFile(std::string(FLAG_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 (FLAG_raw && !input_raw_file) {
input_raw_file.reset(
new RawFile(std::string(FLAG_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 (FLAG_raw && !output_raw_file) {
output_raw_file.reset(
new RawFile(std::string(FLAG_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 (FLAG_raw && !output_raw_file) {
output_raw_file.reset(
new RawFile(std::string(FLAG_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 (FLAG_full) {
if (WritingCallOrderFile()) {
WriteCallOrderData(false /* render_call */, callorder_char_file,
FLAG_callorder_file);
}
if (msg.has_delay()) {
static FILE* delay_file = OpenFile(FLAG_delay_file, "wb");
int32_t delay = msg.delay();
if (FLAG_text) {
fprintf(delay_file, "%d\n", delay);
} else {
WriteData(&delay, sizeof(delay), delay_file, FLAG_delay_file);
}
}
if (msg.has_drift()) {
static FILE* drift_file = OpenFile(FLAG_drift_file, "wb");
int32_t drift = msg.drift();
if (FLAG_text) {
fprintf(drift_file, "%d\n", drift);
} else {
WriteData(&drift, sizeof(drift), drift_file, FLAG_drift_file);
}
}
if (msg.has_level()) {
static FILE* level_file = OpenFile(FLAG_level_file, "wb");
int32_t level = msg.level();
if (FLAG_text) {
fprintf(level_file, "%d\n", level);
} else {
WriteData(&level, sizeof(level), level_file, FLAG_level_file);
}
}
if (msg.has_keypress()) {
static FILE* keypress_file = OpenFile(FLAG_keypress_file, "wb");
bool keypress = msg.keypress();
if (FLAG_text) {
fprintf(keypress_file, "%d\n", keypress);
} else {
WriteData(&keypress, sizeof(keypress), keypress_file,
FLAG_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(pre_amplifier_enabled);
PRINT_CONFIG_FLOAT(pre_amplifier_fixed_gain_factor);
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);
if (msg.has_timestamp_ms()) {
const int64_t timestamp = msg.timestamp_ms();
fprintf(settings_file, " Timestamp in millisecond: %" PRId64 "\n",
timestamp);
}
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 (!FLAG_raw) {
// The WAV files need to be reset every time, because they cant change
// their sample rate or number of channels.
rtc::StringBuilder reverse_name;
reverse_name << FLAG_reverse_file << frame_count << ".wav";
reverse_wav_file.reset(new WavWriter(
reverse_name.str(), reverse_sample_rate, num_reverse_channels));
rtc::StringBuilder input_name;
input_name << FLAG_input_file << frame_count << ".wav";
input_wav_file.reset(new WavWriter(input_name.str(), input_sample_rate,
num_input_channels));
rtc::StringBuilder output_name;
output_name << FLAG_output_file << frame_count << ".wav";
output_wav_file.reset(new WavWriter(
output_name.str(), output_sample_rate, num_output_channels));
if (WritingCallOrderFile()) {
rtc::StringBuilder callorder_name;
callorder_name << FLAG_callorder_file << frame_count << ".char";
callorder_char_file = OpenFile(callorder_name.str(), "wb");
}
}
}
}
return 0;
}
} // namespace webrtc
int main(int argc, char* argv[]) {
return webrtc::do_main(argc, argv);
}