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/*
* Copyright (c) 2018 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 "api/audio/audio_mixer.h"
#include <cstring>
#include <iostream>
#include <vector>
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
#include "common_audio/wav_file.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/audio_mixer/default_output_rate_calculator.h"
#include "rtc_base/strings/string_builder.h"
ABSL_FLAG(int,
sampling_rate,
16000,
"Rate at which to mix (all input streams must have this rate)");
ABSL_FLAG(bool,
stereo,
false,
"Enable stereo (interleaved). Inputs need not be as this parameter.");
ABSL_FLAG(bool, limiter, true, "Enable limiter.");
ABSL_FLAG(std::string,
output_file,
"mixed_file.wav",
"File in which to store the mixed result.");
ABSL_FLAG(std::string, input_file_1, "", "First input. Default none.");
ABSL_FLAG(std::string, input_file_2, "", "Second input. Default none.");
ABSL_FLAG(std::string, input_file_3, "", "Third input. Default none.");
ABSL_FLAG(std::string, input_file_4, "", "Fourth input. Default none.");
namespace webrtc {
namespace test {
class FilePlayingSource : public AudioMixer::Source {
public:
explicit FilePlayingSource(absl::string_view filename)
: wav_reader_(new WavReader(filename)),
sample_rate_hz_(wav_reader_->sample_rate()),
samples_per_channel_(sample_rate_hz_ / 100),
number_of_channels_(wav_reader_->num_channels()) {}
AudioFrameInfo GetAudioFrameWithInfo(int target_rate_hz,
AudioFrame* frame) override {
frame->samples_per_channel_ = samples_per_channel_;
frame->num_channels_ = number_of_channels_;
frame->sample_rate_hz_ = target_rate_hz;
RTC_CHECK_EQ(target_rate_hz, sample_rate_hz_);
const size_t num_to_read = number_of_channels_ * samples_per_channel_;
const size_t num_read =
wav_reader_->ReadSamples(num_to_read, frame->mutable_data());
file_has_ended_ = num_to_read != num_read;
if (file_has_ended_) {
frame->Mute();
}
return file_has_ended_ ? AudioFrameInfo::kMuted : AudioFrameInfo::kNormal;
}
int Ssrc() const override { return 0; }
int PreferredSampleRate() const override { return sample_rate_hz_; }
bool FileHasEnded() const { return file_has_ended_; }
std::string ToString() const {
StringBuilder ss;
ss << "{rate: " << sample_rate_hz_ << ", channels: " << number_of_channels_
<< ", samples_tot: " << wav_reader_->num_samples() << "}";
return ss.Release();
}
private:
std::unique_ptr<WavReader> wav_reader_;
int sample_rate_hz_;
int samples_per_channel_;
int number_of_channels_;
bool file_has_ended_ = false;
};
} // namespace test
} // namespace webrtc
namespace {
const std::vector<std::string> parse_input_files() {
std::vector<std::string> result;
for (auto& x :
{absl::GetFlag(FLAGS_input_file_1), absl::GetFlag(FLAGS_input_file_2),
absl::GetFlag(FLAGS_input_file_3), absl::GetFlag(FLAGS_input_file_4)}) {
if (!x.empty()) {
result.push_back(x);
}
}
return result;
}
} // namespace
int main(int argc, char* argv[]) {
absl::ParseCommandLine(argc, argv);
webrtc::scoped_refptr<webrtc::AudioMixerImpl> mixer(
webrtc::AudioMixerImpl::Create(
std::unique_ptr<webrtc::OutputRateCalculator>(
new webrtc::DefaultOutputRateCalculator()),
absl::GetFlag(FLAGS_limiter)));
const std::vector<std::string> input_files = parse_input_files();
std::vector<webrtc::test::FilePlayingSource> sources;
const int num_channels = absl::GetFlag(FLAGS_stereo) ? 2 : 1;
sources.reserve(input_files.size());
for (const auto& input_file : input_files) {
sources.emplace_back(input_file);
}
for (auto& source : sources) {
auto error = mixer->AddSource(&source);
RTC_CHECK(error);
}
if (sources.empty()) {
std::cout << "Need at least one source!\n";
return 1;
}
const size_t sample_rate = sources[0].PreferredSampleRate();
for (const auto& source : sources) {
RTC_CHECK_EQ(sample_rate, source.PreferredSampleRate());
}
// Print stats.
std::cout << "Limiting is: " << (absl::GetFlag(FLAGS_limiter) ? "on" : "off")
<< "\n"
"Channels: "
<< num_channels
<< "\n"
"Rate: "
<< sample_rate
<< "\n"
"Number of input streams: "
<< input_files.size() << "\n";
for (const auto& source : sources) {
std::cout << "\t" << source.ToString() << "\n";
}
std::cout << "Now mixing\n...\n";
webrtc::WavWriter wav_writer(absl::GetFlag(FLAGS_output_file), sample_rate,
num_channels);
webrtc::AudioFrame frame;
bool all_streams_finished = false;
while (!all_streams_finished) {
mixer->Mix(num_channels, &frame);
RTC_CHECK_EQ(sample_rate / 100, frame.samples_per_channel_);
RTC_CHECK_EQ(sample_rate, frame.sample_rate_hz_);
RTC_CHECK_EQ(num_channels, frame.num_channels_);
wav_writer.WriteSamples(frame.data(),
num_channels * frame.samples_per_channel_);
all_streams_finished =
std::all_of(sources.begin(), sources.end(),
[](const webrtc::test::FilePlayingSource& source) {
return source.FileHasEnded();
});
}
std::cout << "Done!\n" << std::endl;
}
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