blob: 490e99e495401ae4329ffea04db098ea7df074fd [file] [log] [blame]
/*
* Copyright (c) 2017 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_mixer/frame_combiner.h"
#include <numeric>
#include <sstream>
#include <string>
#include "audio/utility/audio_frame_operations.h"
#include "modules/audio_mixer/gain_change_calculator.h"
#include "modules/audio_mixer/sine_wave_generator.h"
#include "rtc_base/checks.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
std::string ProduceDebugText(int sample_rate_hz,
int number_of_channels,
int number_of_sources) {
std::ostringstream ss;
ss << "Sample rate: " << sample_rate_hz << " ,";
ss << "number of channels: " << number_of_channels << " ,";
ss << "number of sources: " << number_of_sources;
return ss.str();
}
std::string ProduceDebugText(int sample_rate_hz,
int number_of_channels,
int number_of_sources,
bool limiter_active,
float wave_frequency) {
std::ostringstream ss;
ss << "Sample rate: " << sample_rate_hz << " ,";
ss << "number of channels: " << number_of_channels << " ,";
ss << "number of sources: " << number_of_sources << " ,";
ss << "limiter active: " << (limiter_active ? "true" : "false") << " ,";
ss << "wave frequency: " << wave_frequency << " ,";
return ss.str();
}
AudioFrame frame1;
AudioFrame frame2;
AudioFrame audio_frame_for_mixing;
void SetUpFrames(int sample_rate_hz, int number_of_channels) {
for (auto* frame : {&frame1, &frame2}) {
frame->UpdateFrame(0, nullptr, rtc::CheckedDivExact(sample_rate_hz, 100),
sample_rate_hz, AudioFrame::kNormalSpeech,
AudioFrame::kVadActive, number_of_channels);
}
}
} // namespace
TEST(FrameCombiner, BasicApiCallsLimiter) {
FrameCombiner combiner(true);
for (const int rate : {8000, 16000, 32000, 48000}) {
for (const int number_of_channels : {1, 2}) {
const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
SetUpFrames(rate, number_of_channels);
for (const int number_of_frames : {0, 1, 2}) {
SCOPED_TRACE(
ProduceDebugText(rate, number_of_channels, number_of_frames));
const std::vector<AudioFrame*> frames_to_combine(
all_frames.begin(), all_frames.begin() + number_of_frames);
combiner.Combine(frames_to_combine, number_of_channels, rate,
frames_to_combine.size(), &audio_frame_for_mixing);
}
}
}
}
// No APM limiter means no AudioProcessing::NativeRate restriction
// on rate. The rate has to be divisible by 100 since we use
// 10 ms frames, though.
TEST(FrameCombiner, BasicApiCallsNoLimiter) {
FrameCombiner combiner(false);
for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
for (const int number_of_channels : {1, 2}) {
const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
SetUpFrames(rate, number_of_channels);
for (const int number_of_frames : {0, 1, 2}) {
SCOPED_TRACE(
ProduceDebugText(rate, number_of_channels, number_of_frames));
const std::vector<AudioFrame*> frames_to_combine(
all_frames.begin(), all_frames.begin() + number_of_frames);
combiner.Combine(frames_to_combine, number_of_channels, rate,
frames_to_combine.size(), &audio_frame_for_mixing);
}
}
}
}
TEST(FrameCombiner, CombiningZeroFramesShouldProduceSilence) {
FrameCombiner combiner(false);
for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
for (const int number_of_channels : {1, 2}) {
SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 0));
const std::vector<AudioFrame*> frames_to_combine;
combiner.Combine(frames_to_combine, number_of_channels, rate,
frames_to_combine.size(), &audio_frame_for_mixing);
const int16_t* audio_frame_for_mixing_data =
audio_frame_for_mixing.data();
const std::vector<int16_t> mixed_data(
audio_frame_for_mixing_data,
audio_frame_for_mixing_data + number_of_channels * rate / 100);
const std::vector<int16_t> expected(number_of_channels * rate / 100, 0);
EXPECT_EQ(mixed_data, expected);
}
}
}
TEST(FrameCombiner, CombiningOneFrameShouldNotChangeFrame) {
FrameCombiner combiner(false);
for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
for (const int number_of_channels : {1, 2}) {
SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 1));
SetUpFrames(rate, number_of_channels);
int16_t* frame1_data = frame1.mutable_data();
std::iota(frame1_data, frame1_data + number_of_channels * rate / 100, 0);
const std::vector<AudioFrame*> frames_to_combine = {&frame1};
combiner.Combine(frames_to_combine, number_of_channels, rate,
frames_to_combine.size(), &audio_frame_for_mixing);
const int16_t* audio_frame_for_mixing_data =
audio_frame_for_mixing.data();
const std::vector<int16_t> mixed_data(
audio_frame_for_mixing_data,
audio_frame_for_mixing_data + number_of_channels * rate / 100);
std::vector<int16_t> expected(number_of_channels * rate / 100);
std::iota(expected.begin(), expected.end(), 0);
EXPECT_EQ(mixed_data, expected);
}
}
}
// Send a sine wave through the FrameCombiner, and check that the
// difference between input and output varies smoothly. This is to
// catch issues like chromium:695993.
TEST(FrameCombiner, GainCurveIsSmoothForAlternatingNumberOfStreams) {
// Test doesn't work with rates requiring a band split, because it
// introduces a small delay measured in single samples, and this
// test cannot handle it.
//
// TODO(aleloi): Add more rates when APM limiter doesn't use band
// split.
for (const bool use_limiter : {true, false}) {
for (const int rate : {8000, 16000}) {
constexpr int number_of_channels = 2;
for (const float wave_frequency : {50, 400, 3200}) {
SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 1, use_limiter,
wave_frequency));
FrameCombiner combiner(use_limiter);
constexpr int16_t wave_amplitude = 30000;
SineWaveGenerator wave_generator(wave_frequency, wave_amplitude);
GainChangeCalculator change_calculator;
float cumulative_change = 0.f;
constexpr size_t iterations = 100;
for (size_t i = 0; i < iterations; ++i) {
SetUpFrames(rate, number_of_channels);
wave_generator.GenerateNextFrame(&frame1);
AudioFrameOperations::Mute(&frame2);
std::vector<AudioFrame*> frames_to_combine = {&frame1};
if (i % 2 == 0) {
frames_to_combine.push_back(&frame2);
}
const size_t number_of_samples =
frame1.samples_per_channel_ * number_of_channels;
// Ensures limiter is on if 'use_limiter'.
constexpr size_t number_of_streams = 2;
combiner.Combine(frames_to_combine, number_of_channels, rate,
number_of_streams, &audio_frame_for_mixing);
cumulative_change += change_calculator.CalculateGainChange(
rtc::ArrayView<const int16_t>(frame1.data(), number_of_samples),
rtc::ArrayView<const int16_t>(audio_frame_for_mixing.data(),
number_of_samples));
}
RTC_DCHECK_LT(cumulative_change, 10);
}
}
}
}
} // namespace webrtc