Files
tubestation/dom/media/gtest/TestAudioInputProcessing.cpp
Andreas Pehrson 880a44a58f Bug 1913932 - Add an event and a generation id to handle multiple concurrent platform processing params. r=padenot
As described, this patch adds two things:
A new event where an audio processing track can get notified that new
processing params are about to get applied to the source. This is to handle use
cases involving concurrent processing params, i.e. when two tracks want to apply
different processing params to the same device. A second such track would
otherwise cause the intersection of both tracks' params to get applied before
the first track gets notified async through the result event, possibly causing a
glitch -- if platform params are relaxed before compensated by the track. And,

A generation id of the processing params getting applied to a particular device.
Since the event informing a processing track of the result of applying platform
processing params is async from the new event informing the processing track of
the platform processing params about to get applied, described above, there can
in theory be any number of requests to apply processing params in flight at any
given time. The generation id is needed to distinguish two distinct requests to
apply identical processing params to a given device, where neither of the
requests have signaled their result yet. This allows the processing track to
avoid relaxing the software processing params prematurely, which could cause a
glitch similar to that described for the new event above.

Differential Revision: https://phabricator.services.mozilla.com/D219691
2024-08-22 09:13:03 +00:00

712 lines
26 KiB
C++

/* -*- 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,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#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();
}
TEST(TestAudioInputProcessing, DisabledPlatformProcessing)
{
const TrackRate rate = 44100;
const uint32_t channels = 1;
auto graph = MakeRefPtr<NiceMock<MockGraph>>(rate);
graph->Init(channels);
auto aip = MakeRefPtr<AudioInputProcessing>(channels);
MediaEnginePrefs settings;
settings.mUsePlatformProcessing = false;
settings.mAecOn = true;
aip->ApplySettings(graph, nullptr, settings);
aip->Start(graph);
EXPECT_EQ(aip->RequestedInputProcessingParams(graph),
CUBEB_INPUT_PROCESSING_PARAM_NONE);
aip->Stop(graph);
graph->Destroy();
}
TEST(TestAudioInputProcessing, EnabledPlatformProcessing)
{
const TrackRate rate = 44100;
const uint32_t channels = 1;
auto graph = MakeRefPtr<NiceMock<MockGraph>>(rate);
graph->Init(channels);
auto aip = MakeRefPtr<AudioInputProcessing>(channels);
MediaEnginePrefs settings;
settings.mUsePlatformProcessing = true;
settings.mAecOn = true;
aip->ApplySettings(graph, nullptr, settings);
aip->Start(graph);
EXPECT_EQ(aip->RequestedInputProcessingParams(graph),
CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION);
aip->Stop(graph);
graph->Destroy();
}
namespace webrtc {
bool operator==(const AudioProcessing::Config& aLhs,
const AudioProcessing::Config& aRhs) {
return aLhs.echo_canceller.enabled == aRhs.echo_canceller.enabled &&
(aLhs.gain_controller1.enabled == aRhs.gain_controller1.enabled ||
aLhs.gain_controller2.enabled == aRhs.gain_controller2.enabled) &&
aLhs.noise_suppression.enabled == aRhs.noise_suppression.enabled;
}
static std::ostream& operator<<(
std::ostream& aStream, const webrtc::AudioProcessing::Config& aConfig) {
aStream << "webrtc::AudioProcessing::Config[";
bool hadPrior = false;
if (aConfig.echo_canceller.enabled) {
aStream << "AEC";
hadPrior = true;
}
if (aConfig.gain_controller1.enabled || aConfig.gain_controller2.enabled) {
if (hadPrior) {
aStream << ", ";
}
aStream << "AGC";
}
if (aConfig.noise_suppression.enabled) {
if (hadPrior) {
aStream << ", ";
}
aStream << "NS";
}
aStream << "]";
return aStream;
}
} // namespace webrtc
TEST(TestAudioInputProcessing, PlatformProcessing)
{
const TrackRate rate = 44100;
const uint32_t channels = 1;
auto graph = MakeRefPtr<NiceMock<MockGraph>>(rate);
graph->Init(channels);
auto aip = MakeRefPtr<AudioInputProcessing>(channels);
MediaEnginePrefs settings;
settings.mUsePlatformProcessing = true;
settings.mAecOn = true;
aip->ApplySettings(graph, nullptr, settings);
aip->Start(graph);
webrtc::AudioProcessing::Config echoOnlyConfig;
echoOnlyConfig.echo_canceller.enabled = true;
webrtc::AudioProcessing::Config noiseOnlyConfig;
noiseOnlyConfig.noise_suppression.enabled = true;
webrtc::AudioProcessing::Config echoNoiseConfig = echoOnlyConfig;
echoNoiseConfig.noise_suppression.enabled = true;
// Config is applied, and platform processing requested.
EXPECT_EQ(aip->RequestedInputProcessingParams(graph),
CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION);
EXPECT_EQ(aip->AppliedConfig(graph), echoOnlyConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
// No other constraint requests present.
aip->NotifySetRequestedInputProcessingParams(
graph, 1, CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION);
EXPECT_EQ(aip->AppliedConfig(graph), echoOnlyConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
// Platform processing params successfully applied.
aip->NotifySetRequestedInputProcessingParamsResult(
graph, 1, CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION);
// Turns off the equivalent APM config.
EXPECT_EQ(aip->AppliedConfig(graph), webrtc::AudioProcessing::Config());
EXPECT_TRUE(aip->IsPassThrough(graph));
// Request for a response that comes back out-of-order later.
aip->NotifySetRequestedInputProcessingParams(
graph, 2, CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION);
// Simulate an error after a driver switch.
aip->NotifySetRequestedInputProcessingParams(
graph, 3, CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION);
// Requesting the same config that is already applied; does nothing.
EXPECT_EQ(aip->AppliedConfig(graph), webrtc::AudioProcessing::Config());
EXPECT_TRUE(aip->IsPassThrough(graph));
// Error notification.
aip->NotifySetRequestedInputProcessingParamsResult(graph, 3,
Err(CUBEB_ERROR));
// The APM config is turned back on, and platform processing is requested to
// be turned off.
EXPECT_EQ(aip->RequestedInputProcessingParams(graph),
CUBEB_INPUT_PROCESSING_PARAM_NONE);
EXPECT_EQ(aip->AppliedConfig(graph), echoOnlyConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
// The request for turning platform processing off.
aip->NotifySetRequestedInputProcessingParams(
graph, 4, CUBEB_INPUT_PROCESSING_PARAM_NONE);
EXPECT_EQ(aip->AppliedConfig(graph), echoOnlyConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
// Pretend there was a response for an old request.
aip->NotifySetRequestedInputProcessingParamsResult(
graph, 2, CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION);
// It does nothing since we are requesting NONE now.
EXPECT_EQ(aip->RequestedInputProcessingParams(graph),
CUBEB_INPUT_PROCESSING_PARAM_NONE);
EXPECT_EQ(aip->AppliedConfig(graph), echoOnlyConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
// Turn it off as requested.
aip->NotifySetRequestedInputProcessingParamsResult(
graph, 4, CUBEB_INPUT_PROCESSING_PARAM_NONE);
EXPECT_EQ(aip->RequestedInputProcessingParams(graph),
CUBEB_INPUT_PROCESSING_PARAM_NONE);
EXPECT_EQ(aip->AppliedConfig(graph), echoOnlyConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
// Test partial support for the requested params.
settings.mNoiseOn = true;
aip->ApplySettings(graph, nullptr, settings);
EXPECT_EQ(aip->RequestedInputProcessingParams(graph),
CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION |
CUBEB_INPUT_PROCESSING_PARAM_NOISE_SUPPRESSION);
EXPECT_EQ(aip->AppliedConfig(graph), echoNoiseConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
// The request doesn't change anything.
aip->NotifySetRequestedInputProcessingParams(
graph, 5,
CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION |
CUBEB_INPUT_PROCESSING_PARAM_NOISE_SUPPRESSION);
EXPECT_EQ(aip->AppliedConfig(graph), echoNoiseConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
// Only noise suppression was supported in the platform.
aip->NotifySetRequestedInputProcessingParamsResult(
graph, 5, CUBEB_INPUT_PROCESSING_PARAM_NOISE_SUPPRESSION);
// In the APM only echo cancellation is applied.
EXPECT_EQ(aip->AppliedConfig(graph), echoOnlyConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
// Test error for partial support.
aip->NotifySetRequestedInputProcessingParams(
graph, 6,
CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION |
CUBEB_INPUT_PROCESSING_PARAM_NOISE_SUPPRESSION);
EXPECT_EQ(aip->AppliedConfig(graph), echoOnlyConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
aip->NotifySetRequestedInputProcessingParamsResult(graph, 6,
Err(CUBEB_ERROR));
// The full config is applied in the APM, and NONE is requested.
EXPECT_EQ(aip->RequestedInputProcessingParams(graph),
CUBEB_INPUT_PROCESSING_PARAM_NONE);
EXPECT_EQ(aip->AppliedConfig(graph), echoNoiseConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
// Enable platform processing again.
aip->ApplySettings(graph, nullptr, settings);
EXPECT_EQ(aip->RequestedInputProcessingParams(graph),
CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION |
CUBEB_INPUT_PROCESSING_PARAM_NOISE_SUPPRESSION);
EXPECT_EQ(aip->AppliedConfig(graph), echoNoiseConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
// Request.
aip->NotifySetRequestedInputProcessingParams(
graph, 7,
CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION |
CUBEB_INPUT_PROCESSING_PARAM_NOISE_SUPPRESSION);
EXPECT_EQ(aip->AppliedConfig(graph), echoNoiseConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
// It succeeded.
aip->NotifySetRequestedInputProcessingParamsResult(
graph, 7,
static_cast<cubeb_input_processing_params>(
CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION |
CUBEB_INPUT_PROCESSING_PARAM_NOISE_SUPPRESSION));
// No config is applied in the APM, and the full set is requested.
EXPECT_EQ(aip->RequestedInputProcessingParams(graph),
CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION |
CUBEB_INPUT_PROCESSING_PARAM_NOISE_SUPPRESSION);
EXPECT_EQ(aip->AppliedConfig(graph), webrtc::AudioProcessing::Config());
EXPECT_TRUE(aip->IsPassThrough(graph));
// Simulate that another concurrent request was made, i.e. two tracks are
// using the same device with different processing params, where the
// intersection of processing params is NONE.
aip->NotifySetRequestedInputProcessingParams(
graph, 8, CUBEB_INPUT_PROCESSING_PARAM_NONE);
// The full config is applied in the APM.
EXPECT_EQ(aip->AppliedConfig(graph), echoNoiseConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
// The result succeeds, leading to no change since sw processing is already
// applied.
aip->NotifySetRequestedInputProcessingParamsResult(
graph, 8, CUBEB_INPUT_PROCESSING_PARAM_NONE);
EXPECT_EQ(aip->RequestedInputProcessingParams(graph),
CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION |
CUBEB_INPUT_PROCESSING_PARAM_NOISE_SUPPRESSION);
EXPECT_EQ(aip->AppliedConfig(graph), echoNoiseConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
// The other concurrent request goes away.
aip->NotifySetRequestedInputProcessingParams(
graph, 9,
CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION |
CUBEB_INPUT_PROCESSING_PARAM_NOISE_SUPPRESSION);
// The full config is still applied in the APM.
EXPECT_EQ(aip->AppliedConfig(graph), echoNoiseConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
// The result succeeds, leading to no change since sw processing is already
// applied.
aip->NotifySetRequestedInputProcessingParamsResult(
graph, 9,
static_cast<cubeb_input_processing_params>(
CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION |
CUBEB_INPUT_PROCESSING_PARAM_NOISE_SUPPRESSION));
EXPECT_EQ(aip->RequestedInputProcessingParams(graph),
CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION |
CUBEB_INPUT_PROCESSING_PARAM_NOISE_SUPPRESSION);
EXPECT_EQ(aip->AppliedConfig(graph), webrtc::AudioProcessing::Config());
EXPECT_TRUE(aip->IsPassThrough(graph));
// Changing input track resets the processing params generation. The applied
// config (AEC, NS) is adapted to the subset applied in the platform (AEC).
aip->Disconnect(graph);
aip->NotifySetRequestedInputProcessingParams(
graph, 1, CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION);
EXPECT_EQ(aip->AppliedConfig(graph), echoNoiseConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
aip->NotifySetRequestedInputProcessingParamsResult(
graph, 1, CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION);
EXPECT_EQ(aip->AppliedConfig(graph), noiseOnlyConfig);
EXPECT_FALSE(aip->IsPassThrough(graph));
aip->Stop(graph);
graph->Destroy();
}