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/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
#include "gmock/gmock.h"
#include "gtest/gtest.h"
#include "AudioGenerator.h"
#include "MediaEngineWebRTCAudio.h"
#include "MediaTrackGraphImpl.h"
#include "PrincipalHandle.h"
#include "mozilla/Attributes.h"
#include "mozilla/NullPrincipal.h"
#include "mozilla/UniquePtr.h"
#include "mozilla/Unused.h"
#include "nsContentUtils.h"
#include "nsTArray.h"
using namespace mozilla;
using testing::NiceMock;
using testing::Return;
class MockGraph : public MediaTrackGraphImpl {
public:
explicit MockGraph(TrackRate aRate)
: MediaTrackGraphImpl(0, aRate, nullptr, AbstractThread::MainThread()) {
ON_CALL(*this, OnGraphThread).WillByDefault(Return(true));
}
void Init(uint32_t aChannels) {
MediaTrackGraphImpl::Init(OFFLINE_THREAD_DRIVER, DIRECT_DRIVER, aChannels);
MonitorAutoLock lock(mMonitor);
// We don't need a graph driver. Advance to
// LIFECYCLE_WAITING_FOR_TRACK_DESTRUCTION so that the driver never
// starts. Graph control messages run as in shutdown, synchronously.
// This permits the main thread part of track initialization through
// AudioProcessingTrack::Create().
mLifecycleState = LIFECYCLE_WAITING_FOR_TRACK_DESTRUCTION;
#ifdef DEBUG
mCanRunMessagesSynchronously = true;
#endif
// Remove this graph's driver since it holds a ref. We are still kept
// alive by the self-ref. Destroy() must be called to break that cycle if
// no tracks are created and destroyed.
mDriver = nullptr;
}
MOCK_CONST_METHOD0(OnGraphThread, bool());
protected:
~MockGraph() = default;
};
// AudioInputProcessing will put extra frames as pre-buffering data to avoid
// glitchs in non pass-through mode. The main goal of the test is to check how
// many frames left in the AudioInputProcessing's mSegment in various situations
// after input data has been processed.
TEST(TestAudioInputProcessing, Buffering)
{
const TrackRate rate = 8000; // So packet size is 80
const uint32_t channels = 1;
auto graph = MakeRefPtr<NiceMock<MockGraph>>(rate);
graph->Init(channels);
RefPtr track = AudioProcessingTrack::Create(graph);
auto aip = MakeRefPtr<AudioInputProcessing>(channels);
const size_t frames = 72;
AudioGenerator<AudioDataValue> generator(channels, rate);
GraphTime processedTime;
GraphTime nextTime;
AudioSegment output;
MediaEnginePrefs settings;
settings.mChannels = channels;
// pref "media.getusermedia.agc2_forced" defaults to true.
// mAgc would need to be set to something other than kAdaptiveAnalog
// for mobile, as asserted in AudioInputProcessing::ConfigForPrefs,
// if gain_controller1 were used.
settings.mAgc2Forced = true;
// Toggle pass-through mode without starting
{
EXPECT_EQ(aip->IsPassThrough(graph), true);
EXPECT_EQ(aip->NumBufferedFrames(graph), 0);
settings.mAgcOn = true;
aip->ApplySettings(graph, nullptr, settings);
EXPECT_EQ(aip->IsPassThrough(graph), false);
EXPECT_EQ(aip->NumBufferedFrames(graph), 0);
settings.mAgcOn = false;
aip->ApplySettings(graph, nullptr, settings);
EXPECT_EQ(aip->NumBufferedFrames(graph), 0);
}
{
// Need (nextTime - processedTime) = 128 - 0 = 128 frames this round.
// aip has not started and set to processing mode yet, so output will be
// filled with silence data directly.
processedTime = 0;
nextTime = MediaTrackGraphImpl::RoundUpToEndOfAudioBlock(frames);
AudioSegment input;
generator.Generate(input, nextTime - processedTime);
aip->Process(track, processedTime, nextTime, &input, &output);
EXPECT_EQ(input.GetDuration(), nextTime - processedTime);
EXPECT_EQ(output.GetDuration(), nextTime);
EXPECT_EQ(aip->NumBufferedFrames(graph), 0);
}
// Set aip to processing/non-pass-through mode
settings.mAgcOn = true;
aip->ApplySettings(graph, nullptr, settings);
{
// Need (nextTime - processedTime) = 256 - 128 = 128 frames this round.
// aip has not started yet, so output will be filled with silence data
// directly.
processedTime = nextTime;
nextTime = MediaTrackGraphImpl::RoundUpToEndOfAudioBlock(2 * frames);
AudioSegment input;
generator.Generate(input, nextTime - processedTime);
aip->Process(track, processedTime, nextTime, &input, &output);
EXPECT_EQ(input.GetDuration(), nextTime - processedTime);
EXPECT_EQ(output.GetDuration(), nextTime);
EXPECT_EQ(aip->NumBufferedFrames(graph), 0);
}
// aip has been set to processing mode and is started.
aip->Start(graph);
{
// Need (nextTime - processedTime) = 256 - 256 = 0 frames this round.
// Process() will return early on 0 frames of input.
// Pre-buffering is not triggered.
processedTime = nextTime;
nextTime = MediaTrackGraphImpl::RoundUpToEndOfAudioBlock(3 * frames);
AudioSegment input;
generator.Generate(input, nextTime - processedTime);
aip->Process(track, processedTime, nextTime, &input, &output);
EXPECT_EQ(input.GetDuration(), nextTime - processedTime);
EXPECT_EQ(output.GetDuration(), nextTime);
EXPECT_EQ(aip->NumBufferedFrames(graph), 0);
}
{
// Need (nextTime - processedTime) = 384 - 256 = 128 frames this round.
// On receipt of the these first frames, aip will insert 80 frames
// into its internal buffer as pre-buffering.
// Process() will take 128 frames from input, packetize and process
// these frames into floor(128/80) = 1 80-frame packet (48 frames left in
// packetizer), insert packets into aip's internal buffer, then move 128
// frames the internal buffer to output, leaving 80 + 80 - 128 = 32 frames
// in aip's internal buffer.
processedTime = nextTime;
nextTime = MediaTrackGraphImpl::RoundUpToEndOfAudioBlock(4 * frames);
AudioSegment input;
generator.Generate(input, nextTime - processedTime);
aip->Process(track, processedTime, nextTime, &input, &output);
EXPECT_EQ(input.GetDuration(), nextTime - processedTime);
EXPECT_EQ(output.GetDuration(), nextTime);
EXPECT_EQ(aip->NumBufferedFrames(graph), 32);
}
{
// Need (nextTime - processedTime) = 384 - 384 = 0 frames this round.
processedTime = nextTime;
nextTime = MediaTrackGraphImpl::RoundUpToEndOfAudioBlock(5 * frames);
AudioSegment input;
generator.Generate(input, nextTime - processedTime);
aip->Process(track, processedTime, nextTime, &input, &output);
EXPECT_EQ(input.GetDuration(), nextTime - processedTime);
EXPECT_EQ(output.GetDuration(), nextTime);
EXPECT_EQ(aip->NumBufferedFrames(graph), 32);
}
{
// Need (nextTime - processedTime) = 512 - 384 = 128 frames this round.
// The Process() aip will take 128 frames from input, packetize and process
// these frames into floor(128+48/80) = 2 80-frame packet (16 frames left in
// packetizer), insert packets into aip's internal buffer, then move 128
// frames the internal buffer to output, leaving 32 + 2*80 - 128 = 64 frames
// in aip's internal buffer.
processedTime = nextTime;
nextTime = MediaTrackGraphImpl::RoundUpToEndOfAudioBlock(6 * frames);
AudioSegment input;
generator.Generate(input, nextTime - processedTime);
aip->Process(track, processedTime, nextTime, &input, &output);
EXPECT_EQ(input.GetDuration(), nextTime - processedTime);
EXPECT_EQ(output.GetDuration(), nextTime);
EXPECT_EQ(aip->NumBufferedFrames(graph), 64);
}
// Set aip to pass-through mode
settings.mAgcOn = false;
aip->ApplySettings(graph, nullptr, settings);
{
// Need (nextTime - processedTime) = 512 - 512 = 0 frames this round.
// No buffering in pass-through mode
processedTime = nextTime;
nextTime = MediaTrackGraphImpl::RoundUpToEndOfAudioBlock(7 * frames);
AudioSegment input;
generator.Generate(input, nextTime - processedTime);
aip->Process(track, processedTime, nextTime, &input, &output);
EXPECT_EQ(input.GetDuration(), nextTime - processedTime);
EXPECT_EQ(output.GetDuration(), processedTime);
EXPECT_EQ(aip->NumBufferedFrames(graph), 0);
}
aip->Stop(graph);
track->Destroy();
}
TEST(TestAudioInputProcessing, ProcessDataWithDifferentPrincipals)
{
const TrackRate rate = 48000; // so # of output frames from packetizer is 480
const uint32_t channels = 2;
auto graph = MakeRefPtr<NiceMock<MockGraph>>(rate);
graph->Init(channels);
RefPtr track = AudioProcessingTrack::Create(graph);
auto aip = MakeRefPtr<AudioInputProcessing>(channels);
AudioGenerator<AudioDataValue> generator(channels, rate);
RefPtr<nsIPrincipal> dummy_principal =
NullPrincipal::CreateWithoutOriginAttributes();
const PrincipalHandle principal1 = MakePrincipalHandle(dummy_principal.get());
const PrincipalHandle principal2 =
MakePrincipalHandle(nsContentUtils::GetSystemPrincipal());
// Total 4800 frames. It's easier to test with frames of multiples of 480.
nsTArray<std::pair<TrackTime, PrincipalHandle>> framesWithPrincipal = {
{100, principal1},
{200, PRINCIPAL_HANDLE_NONE},
{300, principal2},
{400, principal1},
{440, PRINCIPAL_HANDLE_NONE},
// 3 packet-size above.
{480, principal1},
{480, principal2},
{480, PRINCIPAL_HANDLE_NONE},
// 3 packet-size above.
{500, principal2},
{490, principal1},
{600, principal1},
{330, principal1}
// 4 packet-size above.
};
// Generate 4800 frames of data with different principals.
AudioSegment input;
{
for (const auto& [duration, principal] : framesWithPrincipal) {
AudioSegment data;
generator.Generate(data, duration);
for (AudioSegment::ChunkIterator it(data); !it.IsEnded(); it.Next()) {
it->mPrincipalHandle = principal;
}
input.AppendFrom(&data);
}
}
auto verifyPrincipals = [&](const AudioSegment& data) {
TrackTime start = 0;
for (const auto& [duration, principal] : framesWithPrincipal) {
const TrackTime end = start + duration;
AudioSegment slice;
slice.AppendSlice(data, start, end);
start = end;
for (AudioSegment::ChunkIterator it(slice); !it.IsEnded(); it.Next()) {
EXPECT_EQ(it->mPrincipalHandle, principal);
}
}
};
// Check the principals in audio-processing mode.
MediaEnginePrefs settings;
settings.mChannels = channels;
settings.mAgcOn = true;
settings.mAgc2Forced = true;
aip->ApplySettings(graph, nullptr, settings);
EXPECT_EQ(aip->IsPassThrough(graph), false);
aip->Start(graph);
{
AudioSegment output;
{
AudioSegment data;
aip->Process(track, 0, 4800, &input, &data);
EXPECT_EQ(input.GetDuration(), 4800);
EXPECT_EQ(data.GetDuration(), 4800);
// Extract another 480 frames to account for delay from pre-buffering.
EXPECT_EQ(aip->NumBufferedFrames(graph), 480);
AudioSegment dummy;
dummy.AppendNullData(480);
aip->Process(track, 0, 480, &dummy, &data);
EXPECT_EQ(dummy.GetDuration(), 480);
EXPECT_EQ(data.GetDuration(), 480 + 4800);
// Ignore the pre-buffering silence.
output.AppendSlice(data, 480, 480 + 4800);
}
verifyPrincipals(output);
}
// Check the principals in pass-through mode.
settings.mAgcOn = false;
aip->ApplySettings(graph, nullptr, settings);
EXPECT_EQ(aip->IsPassThrough(graph), true);
{
AudioSegment output;
aip->Process(track, 0, 4800, &input, &output);
EXPECT_EQ(input.GetDuration(), 4800);
EXPECT_EQ(output.GetDuration(), 4800);
verifyPrincipals(output);
}
aip->Stop(graph);
track->Destroy();
}
TEST(TestAudioInputProcessing, Downmixing)
{
const TrackRate rate = 44100;
const uint32_t channels = 4;
auto graph = MakeRefPtr<NiceMock<MockGraph>>(rate);
graph->Init(channels);
RefPtr track = AudioProcessingTrack::Create(graph);
auto aip = MakeRefPtr<AudioInputProcessing>(channels);
const size_t frames = 44100;
AudioGenerator<AudioDataValue> generator(channels, rate);
GraphTime processedTime;
GraphTime nextTime;
MediaEnginePrefs settings;
settings.mChannels = channels;
settings.mAgcOn = true;
settings.mAgc2Forced = true;
aip->ApplySettings(graph, nullptr, settings);
EXPECT_EQ(aip->IsPassThrough(graph), false);
aip->Start(graph);
processedTime = 0;
nextTime = MediaTrackGraphImpl::RoundUpToEndOfAudioBlock(frames);
{
AudioSegment input;
AudioSegment output;
generator.Generate(input, nextTime - processedTime);
// Intentionally reduce the amplitude of the generated sine wave so there's
// no chance the max amplitude reaches 1.0, but not enough so that 4
// channels summed together won't clip.
input.ApplyVolume(0.9);
// Process is going to see that it has 4 channels of input, and is going to
// downmix to mono, scaling the input by 1/4 in the process.
// We can't compare the input and output signal because the sine is going to
// be mangledui
aip->Process(track, processedTime, nextTime, &input, &output);
EXPECT_EQ(input.GetDuration(), nextTime - processedTime);
EXPECT_EQ(output.GetDuration(), nextTime);
EXPECT_EQ(output.MaxChannelCount(), 1u);
// Verify that it doesn't clip: the input signal has likely been mangled by
// the various processing passes, but at least it shouldn't clip. We know we
// always have floating point audio here, regardless of the sample-type used
// by Gecko.
for (AudioSegment::ChunkIterator iterOutput(output); !iterOutput.IsEnded();
iterOutput.Next()) {
const float* const output = iterOutput->ChannelData<float>()[0];
for (uint32_t i = 0; i < iterOutput->GetDuration(); i++) {
// Very conservative here, it's likely that the AGC lowers the volume a
// lot.
EXPECT_LE(std::abs(output[i]), 0.95);
}
}
}
// Now, repeat the test in pass-through mode, checking we get the unmodified
// 4 channels.
settings.mAgcOn = false;
aip->ApplySettings(graph, nullptr, settings);
EXPECT_EQ(aip->IsPassThrough(graph), true);
AudioSegment input, output;
processedTime = nextTime;
nextTime += MediaTrackGraphImpl::RoundUpToEndOfAudioBlock(frames);
generator.Generate(input, nextTime - processedTime);
aip->Process(track, processedTime, nextTime, &input, &output);
EXPECT_EQ(input.GetDuration(), nextTime - processedTime);
EXPECT_EQ(output.GetDuration(), nextTime - processedTime);
// This time, no downmix: 4 channels of input, 4 channels of output
EXPECT_EQ(output.MaxChannelCount(), 4u);
nsTArray<AudioDataValue> inputLinearized, outputLinearized;
input.WriteToInterleavedBuffer(inputLinearized, input.MaxChannelCount());
output.WriteToInterleavedBuffer(outputLinearized, output.MaxChannelCount());
// The data should be passed through, and exactly equal.
for (uint32_t i = 0; i < frames * channels; i++) {
EXPECT_EQ(inputLinearized[i], outputLinearized[i]);
}
aip->Stop(graph);
track->Destroy();
}