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/*
* 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 <inttypes.h>
#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
#include <memory>
#include <string>
#include <vector>
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
#include "api/function_view.h"
#include "common_audio/channel_buffer.h"
#include "common_audio/include/audio_util.h"
#include "common_audio/wav_file.h"
#include "modules/audio_processing/test/protobuf_utils.h"
#include "rtc_base/checks.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/system/arch.h"
// Generated at build-time by the protobuf compiler.
#include "modules/audio_processing/debug.pb.h"
ABSL_FLAG(std::string,
input_file,
"input",
"The name of the input stream file.");
ABSL_FLAG(std::string,
output_file,
"ref_out",
"The name of the reference output stream file.");
ABSL_FLAG(std::string,
reverse_file,
"reverse",
"The name of the reverse input stream file.");
ABSL_FLAG(std::string,
delay_file,
"delay.int32",
"The name of the delay file.");
ABSL_FLAG(std::string,
drift_file,
"drift.int32",
"The name of the drift file.");
ABSL_FLAG(std::string,
level_file,
"level.int32",
"The name of the applied input volume file.");
ABSL_FLAG(std::string,
keypress_file,
"keypress.bool",
"The name of the keypress file.");
ABSL_FLAG(std::string,
callorder_file,
"callorder",
"The name of the render/capture call order file.");
ABSL_FLAG(std::string,
settings_file,
"settings.txt",
"The name of the settings file.");
ABSL_FLAG(bool,
full,
false,
"Unpack the full set of files (normally not needed).");
ABSL_FLAG(bool, raw, false, "Write raw data instead of a WAV file.");
ABSL_FLAG(bool,
text,
false,
"Write non-audio files as text files instead of binary files.");
ABSL_FLAG(bool,
use_init_suffix,
false,
"Use init index instead of capture frame count as file name suffix.");
#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::Init;
using audioproc::ReverseStream;
using audioproc::Stream;
namespace {
class RawFile final {
public:
explicit RawFile(const std::string& filename)
: file_handle_(fopen(filename.c_str(), "wb")) {}
~RawFile() { fclose(file_handle_); }
RawFile(const RawFile&) = delete;
RawFile& operator=(const RawFile&) = delete;
void WriteSamples(const int16_t* samples, size_t num_samples) {
#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
#error "Need to convert samples to little-endian when writing to PCM file"
#endif
fwrite(samples, sizeof(*samples), num_samples, file_handle_);
}
void WriteSamples(const float* samples, size_t num_samples) {
fwrite(samples, sizeof(*samples), num_samples, file_handle_);
}
private:
FILE* file_handle_;
};
void WriteIntData(const int16_t* data,
size_t length,
WavWriter* wav_file,
RawFile* raw_file) {
if (wav_file) {
wav_file->WriteSamples(data, length);
}
if (raw_file) {
raw_file->WriteSamples(data, length);
}
}
void WriteFloatData(const float* const* data,
size_t samples_per_channel,
size_t num_channels,
WavWriter* wav_file,
RawFile* raw_file) {
size_t length = num_channels * samples_per_channel;
std::unique_ptr<float[]> buffer(new float[length]);
InterleavedView<float> view(buffer.get(), samples_per_channel, num_channels);
Interleave(data, samples_per_channel, num_channels, view);
if (raw_file) {
raw_file->WriteSamples(buffer.get(), length);
}
// TODO(aluebs): Use ScaleToInt16Range() from audio_util
for (size_t i = 0; i < length; ++i) {
buffer[i] = buffer[i] > 0
? buffer[i] * std::numeric_limits<int16_t>::max()
: -buffer[i] * std::numeric_limits<int16_t>::min();
}
if (wav_file) {
wav_file->WriteSamples(buffer.get(), length);
}
}
// Exits on failure; do not use in unit tests.
FILE* OpenFile(const std::string& filename, const char* mode) {
FILE* file = fopen(filename.c_str(), mode);
RTC_CHECK(file) << "Unable to open file " << filename;
return file;
}
void WriteData(const void* data,
size_t size,
FILE* file,
const std::string& filename) {
RTC_CHECK_EQ(fwrite(data, size, 1, file), 1)
<< "Error when writing to " << filename.c_str();
}
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 absl::GetFlag(FLAGS_full);
}
bool WritingRuntimeSettingFiles() {
return absl::GetFlag(FLAGS_full);
}
// Exports RuntimeSetting AEC dump events to Audacity-readable files.
// This class is not RAII compliant.
class RuntimeSettingWriter {
public:
RuntimeSettingWriter(
std::string name,
rtc::FunctionView<bool(const Event)> is_exporter_for,
rtc::FunctionView<std::string(const Event)> get_timeline_label)
: setting_name_(std::move(name)),
is_exporter_for_(is_exporter_for),
get_timeline_label_(get_timeline_label) {}
~RuntimeSettingWriter() { Flush(); }
bool IsExporterFor(const Event& event) const {
return is_exporter_for_(event);
}
// Writes to file the payload of `event` using `frame_count` to calculate
// timestamp.
void WriteEvent(const Event& event, int frame_count) {
RTC_DCHECK(is_exporter_for_(event));
if (file_ == nullptr) {
rtc::StringBuilder file_name;
file_name << setting_name_ << frame_offset_ << ".txt";
file_ = OpenFile(file_name.str(), "wb");
}
// Time in the current WAV file, in seconds.
double time = (frame_count - frame_offset_) / 100.0;
std::string label = get_timeline_label_(event);
// In Audacity, all annotations are encoded as intervals.
fprintf(file_, "%.6f\t%.6f\t%s \n", time, time, label.c_str());
}
// Handles an AEC dump initialization event, occurring at frame
// `frame_offset`.
void HandleInitEvent(int frame_offset) {
Flush();
frame_offset_ = frame_offset;
}
private:
void Flush() {
if (file_ != nullptr) {
fclose(file_);
file_ = nullptr;
}
}
FILE* file_ = nullptr;
int frame_offset_ = 0;
const std::string setting_name_;
const rtc::FunctionView<bool(Event)> is_exporter_for_;
const rtc::FunctionView<std::string(Event)> get_timeline_label_;
};
// Returns RuntimeSetting exporters for runtime setting types defined in
// debug.proto.
std::vector<RuntimeSettingWriter> RuntimeSettingWriters() {
return {
RuntimeSettingWriter(
"CapturePreGain",
[](const Event& event) -> bool {
return event.runtime_setting().has_capture_pre_gain();
},
[](const Event& event) -> std::string {
return std::to_string(event.runtime_setting().capture_pre_gain());
}),
RuntimeSettingWriter(
"CustomRenderProcessingRuntimeSetting",
[](const Event& event) -> bool {
return event.runtime_setting()
.has_custom_render_processing_setting();
},
[](const Event& event) -> std::string {
return std::to_string(
event.runtime_setting().custom_render_processing_setting());
}),
RuntimeSettingWriter(
"CaptureFixedPostGain",
[](const Event& event) -> bool {
return event.runtime_setting().has_capture_fixed_post_gain();
},
[](const Event& event) -> std::string {
return std::to_string(
event.runtime_setting().capture_fixed_post_gain());
}),
RuntimeSettingWriter(
"PlayoutVolumeChange",
[](const Event& event) -> bool {
return event.runtime_setting().has_playout_volume_change();
},
[](const Event& event) -> std::string {
return std::to_string(
event.runtime_setting().playout_volume_change());
})};
}
std::string GetWavFileIndex(int init_index, int frame_count) {
rtc::StringBuilder suffix;
if (absl::GetFlag(FLAGS_use_init_suffix)) {
suffix << "_" << init_index;
} else {
suffix << frame_count;
}
return suffix.str();
}
} // namespace
int do_main(int argc, char* argv[]) {
std::vector<char*> args = absl::ParseCommandLine(argc, argv);
std::string program_name = args[0];
std::string usage =
"Commandline tool to unpack audioproc debug files.\n"
"Example usage:\n" +
program_name + " debug_dump.pb\n";
if (args.size() < 2) {
printf("%s", usage.c_str());
return 1;
}
FILE* debug_file = OpenFile(args[1], "rb");
Event event_msg;
int frame_count = 0;
int init_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 << absl::GetFlag(FLAGS_callorder_file) << ".char";
FILE* callorder_char_file = WritingCallOrderFile()
? OpenFile(callorder_raw_name.str(), "wb")
: nullptr;
FILE* settings_file = OpenFile(absl::GetFlag(FLAGS_settings_file), "wb");
std::vector<RuntimeSettingWriter> runtime_setting_writers =
RuntimeSettingWriters();
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 (absl::GetFlag(FLAGS_raw) && !reverse_raw_file) {
reverse_raw_file.reset(
new RawFile(absl::GetFlag(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 (absl::GetFlag(FLAGS_raw) && !reverse_raw_file) {
reverse_raw_file.reset(
new RawFile(absl::GetFlag(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());
}
if (absl::GetFlag(FLAGS_full)) {
if (WritingCallOrderFile()) {
WriteCallOrderData(true /* render_call */, callorder_char_file,
absl::GetFlag(FLAGS_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 (absl::GetFlag(FLAGS_raw) && !input_raw_file) {
input_raw_file.reset(
new RawFile(absl::GetFlag(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 (absl::GetFlag(FLAGS_raw) && !input_raw_file) {
input_raw_file.reset(
new RawFile(absl::GetFlag(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 (absl::GetFlag(FLAGS_raw) && !output_raw_file) {
output_raw_file.reset(
new RawFile(absl::GetFlag(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 (absl::GetFlag(FLAGS_raw) && !output_raw_file) {
output_raw_file.reset(
new RawFile(absl::GetFlag(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 (absl::GetFlag(FLAGS_full)) {
if (WritingCallOrderFile()) {
WriteCallOrderData(false /* render_call */, callorder_char_file,
absl::GetFlag(FLAGS_callorder_file));
}
if (msg.has_delay()) {
static FILE* delay_file =
OpenFile(absl::GetFlag(FLAGS_delay_file), "wb");
int32_t delay = msg.delay();
if (absl::GetFlag(FLAGS_text)) {
fprintf(delay_file, "%d\n", delay);
} else {
WriteData(&delay, sizeof(delay), delay_file,
absl::GetFlag(FLAGS_delay_file));
}
}
if (msg.has_drift()) {
static FILE* drift_file =
OpenFile(absl::GetFlag(FLAGS_drift_file), "wb");
int32_t drift = msg.drift();
if (absl::GetFlag(FLAGS_text)) {
fprintf(drift_file, "%d\n", drift);
} else {
WriteData(&drift, sizeof(drift), drift_file,
absl::GetFlag(FLAGS_drift_file));
}
}
if (msg.has_applied_input_volume()) {
static FILE* level_file =
OpenFile(absl::GetFlag(FLAGS_level_file), "wb");
int32_t level = msg.applied_input_volume();
if (absl::GetFlag(FLAGS_text)) {
fprintf(level_file, "%d\n", level);
} else {
WriteData(&level, sizeof(level), level_file,
absl::GetFlag(FLAGS_level_file));
}
}
if (msg.has_keypress()) {
static FILE* keypress_file =
OpenFile(absl::GetFlag(FLAGS_keypress_file), "wb");
bool keypress = msg.keypress();
if (absl::GetFlag(FLAGS_text)) {
fprintf(keypress_file, "%d\n", keypress);
} else {
WriteData(&keypress, sizeof(keypress), keypress_file,
absl::GetFlag(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(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;
}
++init_count;
const Init msg = event_msg.init();
// These should print out zeros if they're missing.
fprintf(settings_file, "Init #%d at frame: %d\n", init_count,
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: %zu\n", num_input_channels);
num_output_channels = msg.num_output_channels();
fprintf(settings_file, " Output channels: %zu\n", num_output_channels);
num_reverse_channels = msg.num_reverse_channels();
fprintf(settings_file, " Reverse channels: %zu\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 (!absl::GetFlag(FLAGS_raw)) {
// The WAV files need to be reset every time, because they cant change
// their sample rate or number of channels.
std::string suffix = GetWavFileIndex(init_count, frame_count);
rtc::StringBuilder reverse_name;
reverse_name << absl::GetFlag(FLAGS_reverse_file) << suffix << ".wav";
reverse_wav_file.reset(new WavWriter(
reverse_name.str(), reverse_sample_rate, num_reverse_channels));
rtc::StringBuilder input_name;
input_name << absl::GetFlag(FLAGS_input_file) << suffix << ".wav";
input_wav_file.reset(new WavWriter(input_name.str(), input_sample_rate,
num_input_channels));
rtc::StringBuilder output_name;
output_name << absl::GetFlag(FLAGS_output_file) << suffix << ".wav";
output_wav_file.reset(new WavWriter(
output_name.str(), output_sample_rate, num_output_channels));
if (WritingCallOrderFile()) {
rtc::StringBuilder callorder_name;
callorder_name << absl::GetFlag(FLAGS_callorder_file) << suffix
<< ".char";
callorder_char_file = OpenFile(callorder_name.str(), "wb");
}
if (WritingRuntimeSettingFiles()) {
for (RuntimeSettingWriter& writer : runtime_setting_writers) {
writer.HandleInitEvent(frame_count);
}
}
}
} else if (event_msg.type() == Event::RUNTIME_SETTING) {
if (WritingRuntimeSettingFiles()) {
for (RuntimeSettingWriter& writer : runtime_setting_writers) {
if (writer.IsExporterFor(event_msg)) {
writer.WriteEvent(event_msg, frame_count);
}
}
}
}
}
return 0;
}
} // namespace webrtc
int main(int argc, char* argv[]) {
return webrtc::do_main(argc, argv);
}