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webrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
Alessio Bazzica b46c4bf27b [ACM] iSAC audio codec removed 
Note: this CL has to leave behind one part of iSAC, which is its VAD
currently used by AGC1 in APM. The target visibility has been
restricted and the VAD will be removed together with AGC1 when the
time comes.

Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319

Bug: webrtc:14450
Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38652}
2022-11-16 16:42:55 +00:00

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/*
* Copyright (c) 2013 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_coding/acm2/acm_receiver.h"
#include <algorithm> // std::min
#include <memory>
#include "absl/types/optional.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "system_wrappers/include/clock.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
namespace acm2 {
class AcmReceiverTestOldApi : public AudioPacketizationCallback,
public ::testing::Test {
protected:
AcmReceiverTestOldApi()
: timestamp_(0),
packet_sent_(false),
last_packet_send_timestamp_(timestamp_),
last_frame_type_(AudioFrameType::kEmptyFrame) {
config_.decoder_factory = decoder_factory_;
}
~AcmReceiverTestOldApi() {}
void SetUp() override {
acm_.reset(AudioCodingModule::Create(config_));
receiver_.reset(new AcmReceiver(config_));
ASSERT_TRUE(receiver_.get() != NULL);
ASSERT_TRUE(acm_.get() != NULL);
acm_->InitializeReceiver();
acm_->RegisterTransportCallback(this);
rtp_header_.sequenceNumber = 0;
rtp_header_.timestamp = 0;
rtp_header_.markerBit = false;
rtp_header_.ssrc = 0x12345678; // Arbitrary.
rtp_header_.numCSRCs = 0;
rtp_header_.payloadType = 0;
}
void TearDown() override {}
AudioCodecInfo SetEncoder(int payload_type,
const SdpAudioFormat& format,
const std::map<int, int> cng_payload_types = {}) {
// Create the speech encoder.
absl::optional<AudioCodecInfo> info =
encoder_factory_->QueryAudioEncoder(format);
RTC_CHECK(info.has_value());
std::unique_ptr<AudioEncoder> enc =
encoder_factory_->MakeAudioEncoder(payload_type, format, absl::nullopt);
// If we have a compatible CN specification, stack a CNG on top.
auto it = cng_payload_types.find(info->sample_rate_hz);
if (it != cng_payload_types.end()) {
AudioEncoderCngConfig config;
config.speech_encoder = std::move(enc);
config.num_channels = 1;
config.payload_type = it->second;
config.vad_mode = Vad::kVadNormal;
enc = CreateComfortNoiseEncoder(std::move(config));
}
// Actually start using the new encoder.
acm_->SetEncoder(std::move(enc));
return *info;
}
int InsertOnePacketOfSilence(const AudioCodecInfo& info) {
// Frame setup according to the codec.
AudioFrame frame;
frame.sample_rate_hz_ = info.sample_rate_hz;
frame.samples_per_channel_ = info.sample_rate_hz / 100; // 10 ms.
frame.num_channels_ = info.num_channels;
frame.Mute();
packet_sent_ = false;
last_packet_send_timestamp_ = timestamp_;
int num_10ms_frames = 0;
while (!packet_sent_) {
frame.timestamp_ = timestamp_;
timestamp_ += rtc::checked_cast<uint32_t>(frame.samples_per_channel_);
EXPECT_GE(acm_->Add10MsData(frame), 0);
++num_10ms_frames;
}
return num_10ms_frames;
}
int SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes,
int64_t absolute_capture_timestamp_ms) override {
if (frame_type == AudioFrameType::kEmptyFrame)
return 0;
rtp_header_.payloadType = payload_type;
rtp_header_.timestamp = timestamp;
int ret_val = receiver_->InsertPacket(
rtp_header_,
rtc::ArrayView<const uint8_t>(payload_data, payload_len_bytes));
if (ret_val < 0) {
RTC_DCHECK_NOTREACHED();
return -1;
}
rtp_header_.sequenceNumber++;
packet_sent_ = true;
last_frame_type_ = frame_type;
return 0;
}
const rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_ =
CreateBuiltinAudioEncoderFactory();
const rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_ =
CreateBuiltinAudioDecoderFactory();
AudioCodingModule::Config config_;
std::unique_ptr<AcmReceiver> receiver_;
std::unique_ptr<AudioCodingModule> acm_;
RTPHeader rtp_header_;
uint32_t timestamp_;
bool packet_sent_; // Set when SendData is called reset when inserting audio.
uint32_t last_packet_send_timestamp_;
AudioFrameType last_frame_type_;
};
#if defined(WEBRTC_ANDROID)
#define MAYBE_SampleRate DISABLED_SampleRate
#else
#define MAYBE_SampleRate SampleRate
#endif
TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
const std::map<int, SdpAudioFormat> codecs = {{0, {"OPUS", 48000, 2}}};
receiver_->SetCodecs(codecs);
constexpr int kOutSampleRateHz = 8000; // Different than codec sample rate.
for (size_t i = 0; i < codecs.size(); ++i) {
const int payload_type = rtc::checked_cast<int>(i);
const int num_10ms_frames =
InsertOnePacketOfSilence(SetEncoder(payload_type, codecs.at(i)));
for (int k = 0; k < num_10ms_frames; ++k) {
AudioFrame frame;
bool muted;
EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame, &muted));
}
EXPECT_EQ(encoder_factory_->QueryAudioEncoder(codecs.at(i))->sample_rate_hz,
receiver_->last_output_sample_rate_hz());
}
}
class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
protected:
AcmReceiverTestFaxModeOldApi() {
config_.neteq_config.for_test_no_time_stretching = true;
}
void RunVerifyAudioFrame(const SdpAudioFormat& codec) {
// Make sure "fax mode" is enabled. This will avoid delay changes unless the
// packet-loss concealment is made. We do this in order to make the
// timestamp increments predictable; in normal mode, NetEq may decide to do
// accelerate or pre-emptive expand operations after some time, offsetting
// the timestamp.
EXPECT_TRUE(config_.neteq_config.for_test_no_time_stretching);
constexpr int payload_type = 17;
receiver_->SetCodecs({{payload_type, codec}});
const AudioCodecInfo info = SetEncoder(payload_type, codec);
const int output_sample_rate_hz = info.sample_rate_hz;
const size_t output_channels = info.num_channels;
const size_t samples_per_ms = rtc::checked_cast<size_t>(
rtc::CheckedDivExact(output_sample_rate_hz, 1000));
const AudioFrame::VADActivity expected_vad_activity =
output_sample_rate_hz > 16000 ? AudioFrame::kVadActive
: AudioFrame::kVadPassive;
// Expect the first output timestamp to be 5*fs/8000 samples before the
// first inserted timestamp (because of NetEq's look-ahead). (This value is
// defined in Expand::overlap_length_.)
uint32_t expected_output_ts =
last_packet_send_timestamp_ -
rtc::CheckedDivExact(5 * output_sample_rate_hz, 8000);
AudioFrame frame;
bool muted;
EXPECT_EQ(0, receiver_->GetAudio(output_sample_rate_hz, &frame, &muted));
// Expect timestamp = 0 before first packet is inserted.
EXPECT_EQ(0u, frame.timestamp_);
for (int i = 0; i < 5; ++i) {
const int num_10ms_frames = InsertOnePacketOfSilence(info);
for (int k = 0; k < num_10ms_frames; ++k) {
EXPECT_EQ(0,
receiver_->GetAudio(output_sample_rate_hz, &frame, &muted));
EXPECT_EQ(expected_output_ts, frame.timestamp_);
expected_output_ts += rtc::checked_cast<uint32_t>(10 * samples_per_ms);
EXPECT_EQ(10 * samples_per_ms, frame.samples_per_channel_);
EXPECT_EQ(output_sample_rate_hz, frame.sample_rate_hz_);
EXPECT_EQ(output_channels, frame.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, frame.speech_type_);
EXPECT_EQ(expected_vad_activity, frame.vad_activity_);
EXPECT_FALSE(muted);
}
}
}
};
#if defined(WEBRTC_ANDROID)
#define MAYBE_VerifyAudioFramePCMU DISABLED_VerifyAudioFramePCMU
#else
#define MAYBE_VerifyAudioFramePCMU VerifyAudioFramePCMU
#endif
TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFramePCMU) {
RunVerifyAudioFrame({"PCMU", 8000, 1});
}
#if defined(WEBRTC_ANDROID)
#define MAYBE_VerifyAudioFrameOpus DISABLED_VerifyAudioFrameOpus
#else
#define MAYBE_VerifyAudioFrameOpus VerifyAudioFrameOpus
#endif
TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameOpus) {
RunVerifyAudioFrame({"opus", 48000, 2});
}
#if defined(WEBRTC_ANDROID)
#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
#else
#define MAYBE_PostdecodingVad PostdecodingVad
#endif
TEST_F(AcmReceiverTestOldApi, MAYBE_PostdecodingVad) {
EXPECT_TRUE(config_.neteq_config.enable_post_decode_vad);
constexpr int payload_type = 34;
const SdpAudioFormat codec = {"L16", 16000, 1};
const AudioCodecInfo info = SetEncoder(payload_type, codec);
receiver_->SetCodecs({{payload_type, codec}});
constexpr int kNumPackets = 5;
AudioFrame frame;
for (int n = 0; n < kNumPackets; ++n) {
const int num_10ms_frames = InsertOnePacketOfSilence(info);
for (int k = 0; k < num_10ms_frames; ++k) {
bool muted;
ASSERT_EQ(0, receiver_->GetAudio(info.sample_rate_hz, &frame, &muted));
}
}
EXPECT_EQ(AudioFrame::kVadPassive, frame.vad_activity_);
}
class AcmReceiverTestPostDecodeVadPassiveOldApi : public AcmReceiverTestOldApi {
protected:
AcmReceiverTestPostDecodeVadPassiveOldApi() {
config_.neteq_config.enable_post_decode_vad = false;
}
};
#if defined(WEBRTC_ANDROID)
#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
#else
#define MAYBE_PostdecodingVad PostdecodingVad
#endif
TEST_F(AcmReceiverTestPostDecodeVadPassiveOldApi, MAYBE_PostdecodingVad) {
EXPECT_FALSE(config_.neteq_config.enable_post_decode_vad);
constexpr int payload_type = 34;
const SdpAudioFormat codec = {"L16", 16000, 1};
const AudioCodecInfo info = SetEncoder(payload_type, codec);
auto const value = encoder_factory_->QueryAudioEncoder(codec);
ASSERT_TRUE(value.has_value());
receiver_->SetCodecs({{payload_type, codec}});
const int kNumPackets = 5;
AudioFrame frame;
for (int n = 0; n < kNumPackets; ++n) {
const int num_10ms_frames = InsertOnePacketOfSilence(info);
for (int k = 0; k < num_10ms_frames; ++k) {
bool muted;
ASSERT_EQ(0, receiver_->GetAudio(info.sample_rate_hz, &frame, &muted));
}
}
EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
}
#if defined(WEBRTC_ANDROID)
#define MAYBE_LastAudioCodec DISABLED_LastAudioCodec
#else
#define MAYBE_LastAudioCodec LastAudioCodec
#endif
#if defined(WEBRTC_CODEC_OPUS)
TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) {
const std::map<int, SdpAudioFormat> codecs = {
{0, {"PCMU", 8000, 1}}, {1, {"PCMA", 8000, 1}}, {2, {"L16", 32000, 1}}};
const std::map<int, int> cng_payload_types = {
{8000, 100}, {16000, 101}, {32000, 102}};
{
std::map<int, SdpAudioFormat> receive_codecs = codecs;
for (const auto& cng_type : cng_payload_types) {
receive_codecs.emplace(std::make_pair(
cng_type.second, SdpAudioFormat("CN", cng_type.first, 1)));
}
receiver_->SetCodecs(receive_codecs);
}
// No audio payload is received.
EXPECT_EQ(absl::nullopt, receiver_->LastDecoder());
// Start with sending DTX.
packet_sent_ = false;
InsertOnePacketOfSilence(
SetEncoder(0, codecs.at(0), cng_payload_types)); // Enough to test
// with one codec.
ASSERT_TRUE(packet_sent_);
EXPECT_EQ(AudioFrameType::kAudioFrameCN, last_frame_type_);
// Has received, only, DTX. Last Audio codec is undefined.
EXPECT_EQ(absl::nullopt, receiver_->LastDecoder());
EXPECT_EQ(absl::nullopt, receiver_->last_packet_sample_rate_hz());
for (size_t i = 0; i < codecs.size(); ++i) {
// Set DTX off to send audio payload.
packet_sent_ = false;
const int payload_type = rtc::checked_cast<int>(i);
const AudioCodecInfo info_without_cng =
SetEncoder(payload_type, codecs.at(i));
InsertOnePacketOfSilence(info_without_cng);
// Sanity check if Actually an audio payload received, and it should be
// of type "speech."
ASSERT_TRUE(packet_sent_);
ASSERT_EQ(AudioFrameType::kAudioFrameSpeech, last_frame_type_);
EXPECT_EQ(info_without_cng.sample_rate_hz,
receiver_->last_packet_sample_rate_hz());
// Set VAD on to send DTX. Then check if the "Last Audio codec" returns
// the expected codec. Encode repeatedly until a DTX is sent.
const AudioCodecInfo info_with_cng =
SetEncoder(payload_type, codecs.at(i), cng_payload_types);
while (last_frame_type_ != AudioFrameType::kAudioFrameCN) {
packet_sent_ = false;
InsertOnePacketOfSilence(info_with_cng);
ASSERT_TRUE(packet_sent_);
}
EXPECT_EQ(info_with_cng.sample_rate_hz,
receiver_->last_packet_sample_rate_hz());
EXPECT_EQ(codecs.at(i), receiver_->LastDecoder()->second);
}
}
#endif
// Check if the statistics are initialized correctly. Before any call to ACM
// all fields have to be zero.
#if defined(WEBRTC_ANDROID)
#define MAYBE_InitializedToZero DISABLED_InitializedToZero
#else
#define MAYBE_InitializedToZero InitializedToZero
#endif
TEST_F(AcmReceiverTestOldApi, MAYBE_InitializedToZero) {
AudioDecodingCallStats stats;
receiver_->GetDecodingCallStatistics(&stats);
EXPECT_EQ(0, stats.calls_to_neteq);
EXPECT_EQ(0, stats.calls_to_silence_generator);
EXPECT_EQ(0, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(0, stats.decoded_neteq_plc);
EXPECT_EQ(0, stats.decoded_plc_cng);
EXPECT_EQ(0, stats.decoded_muted_output);
}
// Insert some packets and pull audio. Check statistics are valid. Then,
// simulate packet loss and check if PLC and PLC-to-CNG statistics are
// correctly updated.
#if defined(WEBRTC_ANDROID)
#define MAYBE_NetEqCalls DISABLED_NetEqCalls
#else
#define MAYBE_NetEqCalls NetEqCalls
#endif
TEST_F(AcmReceiverTestOldApi, MAYBE_NetEqCalls) {
AudioDecodingCallStats stats;
const int kNumNormalCalls = 10;
const int kSampleRateHz = 16000;
const int kNumSamples10ms = kSampleRateHz / 100;
const int kFrameSizeMs = 10; // Multiple of 10.
const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms;
const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
const uint8_t kPayloadType = 111;
RTPHeader rtp_header;
AudioFrame audio_frame;
bool muted;
receiver_->SetCodecs(
{{kPayloadType, SdpAudioFormat("L16", kSampleRateHz, 1)}});
rtp_header.sequenceNumber = 0xABCD;
rtp_header.timestamp = 0xABCDEF01;
rtp_header.payloadType = kPayloadType;
rtp_header.markerBit = false;
rtp_header.ssrc = 0x1234;
rtp_header.numCSRCs = 0;
rtp_header.payload_type_frequency = kSampleRateHz;
for (int num_calls = 0; num_calls < kNumNormalCalls; ++num_calls) {
const uint8_t kPayload[kPayloadSizeBytes] = {0};
ASSERT_EQ(0, receiver_->InsertPacket(rtp_header, kPayload));
++rtp_header.sequenceNumber;
rtp_header.timestamp += kFrameSizeSamples;
ASSERT_EQ(0, receiver_->GetAudio(-1, &audio_frame, &muted));
EXPECT_FALSE(muted);
}
receiver_->GetDecodingCallStatistics(&stats);
EXPECT_EQ(kNumNormalCalls, stats.calls_to_neteq);
EXPECT_EQ(0, stats.calls_to_silence_generator);
EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(0, stats.decoded_neteq_plc);
EXPECT_EQ(0, stats.decoded_plc_cng);
EXPECT_EQ(0, stats.decoded_muted_output);
const int kNumPlc = 3;
const int kNumPlcCng = 5;
// Simulate packet-loss. NetEq first performs PLC then PLC fades to CNG.
for (int n = 0; n < kNumPlc + kNumPlcCng; ++n) {
ASSERT_EQ(0, receiver_->GetAudio(-1, &audio_frame, &muted));
EXPECT_FALSE(muted);
}
receiver_->GetDecodingCallStatistics(&stats);
EXPECT_EQ(kNumNormalCalls + kNumPlc + kNumPlcCng, stats.calls_to_neteq);
EXPECT_EQ(0, stats.calls_to_silence_generator);
EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
EXPECT_EQ(0, stats.decoded_cng);
EXPECT_EQ(kNumPlc, stats.decoded_neteq_plc);
EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng);
EXPECT_EQ(0, stats.decoded_muted_output);
// TODO(henrik.lundin) Add a test with muted state enabled.
}
} // namespace acm2
} // namespace webrtc
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