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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 "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "modules/audio_coding/acm2/rent_a_codec.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.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/fileutils.h"
namespace webrtc {
namespace acm2 {
namespace {
bool CodecsEqual(const CodecInst& codec_a, const CodecInst& codec_b) {
if (strcmp(codec_a.plname, codec_b.plname) != 0 ||
codec_a.plfreq != codec_b.plfreq ||
codec_a.pltype != codec_b.pltype ||
codec_b.channels != codec_a.channels)
return false;
return true;
}
struct CodecIdInst {
explicit CodecIdInst(RentACodec::CodecId codec_id) {
const auto codec_ix = RentACodec::CodecIndexFromId(codec_id);
EXPECT_TRUE(codec_ix);
id = *codec_ix;
const auto codec_inst = RentACodec::CodecInstById(codec_id);
EXPECT_TRUE(codec_inst);
inst = *codec_inst;
}
int id;
CodecInst inst;
};
} // namespace
class AcmReceiverTestOldApi : public AudioPacketizationCallback,
public ::testing::Test {
protected:
AcmReceiverTestOldApi()
: timestamp_(0),
packet_sent_(false),
last_packet_send_timestamp_(timestamp_),
last_frame_type_(kEmptyFrame) {
config_.decoder_factory = CreateBuiltinAudioDecoderFactory();
}
~AcmReceiverTestOldApi() {}
void SetUp() override {
acm_.reset(AudioCodingModule::Create(config_));
receiver_.reset(new AcmReceiver(config_));
ASSERT_TRUE(receiver_.get() != NULL);
ASSERT_TRUE(acm_.get() != NULL);
codecs_ = RentACodec::Database();
acm_->InitializeReceiver();
acm_->RegisterTransportCallback(this);
rtp_header_.header.sequenceNumber = 0;
rtp_header_.header.timestamp = 0;
rtp_header_.header.markerBit = false;
rtp_header_.header.ssrc = 0x12345678; // Arbitrary.
rtp_header_.header.numCSRCs = 0;
rtp_header_.header.payloadType = 0;
rtp_header_.frameType = kAudioFrameSpeech;
rtp_header_.type.Audio.isCNG = false;
}
void TearDown() override {}
void InsertOnePacketOfSilence(int codec_id) {
CodecInst codec =
*RentACodec::CodecInstById(*RentACodec::CodecIdFromIndex(codec_id));
if (timestamp_ == 0) { // This is the first time inserting audio.
ASSERT_EQ(0, acm_->RegisterSendCodec(codec));
} else {
auto current_codec = acm_->SendCodec();
ASSERT_TRUE(current_codec);
if (!CodecsEqual(codec, *current_codec))
ASSERT_EQ(0, acm_->RegisterSendCodec(codec));
}
AudioFrame frame;
// Frame setup according to the codec.
frame.sample_rate_hz_ = codec.plfreq;
frame.samples_per_channel_ = codec.plfreq / 100; // 10 ms.
frame.num_channels_ = codec.channels;
frame.Mute();
packet_sent_ = false;
last_packet_send_timestamp_ = timestamp_;
while (!packet_sent_) {
frame.timestamp_ = timestamp_;
timestamp_ += rtc::checked_cast<uint32_t>(frame.samples_per_channel_);
ASSERT_GE(acm_->Add10MsData(frame), 0);
}
}
template <size_t N>
void AddSetOfCodecs(const RentACodec::CodecId(&ids)[N]) {
for (auto id : ids) {
const auto i = RentACodec::CodecIndexFromId(id);
ASSERT_TRUE(i);
ASSERT_EQ(0, receiver_->AddCodec(*i, codecs_[*i].pltype,
codecs_[*i].channels, codecs_[*i].plfreq,
nullptr, codecs_[*i].plname));
}
}
int SendData(FrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) override {
if (frame_type == kEmptyFrame)
return 0;
rtp_header_.header.payloadType = payload_type;
rtp_header_.frameType = frame_type;
if (frame_type == kAudioFrameSpeech)
rtp_header_.type.Audio.isCNG = false;
else
rtp_header_.type.Audio.isCNG = true;
rtp_header_.header.timestamp = timestamp;
int ret_val = receiver_->InsertPacket(
rtp_header_,
rtc::ArrayView<const uint8_t>(payload_data, payload_len_bytes));
if (ret_val < 0) {
assert(false);
return -1;
}
rtp_header_.header.sequenceNumber++;
packet_sent_ = true;
last_frame_type_ = frame_type;
return 0;
}
AudioCodingModule::Config config_;
std::unique_ptr<AcmReceiver> receiver_;
rtc::ArrayView<const CodecInst> codecs_;
std::unique_ptr<AudioCodingModule> acm_;
WebRtcRTPHeader rtp_header_;
uint32_t timestamp_;
bool packet_sent_; // Set when SendData is called reset when inserting audio.
uint32_t last_packet_send_timestamp_;
FrameType last_frame_type_;
};
#if defined(WEBRTC_ANDROID)
#define MAYBE_AddCodecGetCodec DISABLED_AddCodecGetCodec
#else
#define MAYBE_AddCodecGetCodec AddCodecGetCodec
#endif
TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecGetCodec) {
// Add codec.
for (size_t n = 0; n < codecs_.size(); ++n) {
if (n & 0x1) { // Just add codecs with odd index.
EXPECT_EQ(
0, receiver_->AddCodec(rtc::checked_cast<int>(n), codecs_[n].pltype,
codecs_[n].channels, codecs_[n].plfreq, NULL,
codecs_[n].plname));
}
}
// Get codec and compare.
for (size_t n = 0; n < codecs_.size(); ++n) {
CodecInst my_codec;
if (n & 0x1) {
// Codecs with odd index should match the reference.
EXPECT_EQ(0, receiver_->DecoderByPayloadType(codecs_[n].pltype,
&my_codec));
EXPECT_TRUE(CodecsEqual(codecs_[n], my_codec));
} else {
// Codecs with even index are not registered.
EXPECT_EQ(-1, receiver_->DecoderByPayloadType(codecs_[n].pltype,
&my_codec));
}
}
}
#if defined(WEBRTC_ANDROID)
#define MAYBE_AddCodecChangePayloadType DISABLED_AddCodecChangePayloadType
#else
#define MAYBE_AddCodecChangePayloadType AddCodecChangePayloadType
#endif
TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecChangePayloadType) {
const CodecIdInst codec1(RentACodec::CodecId::kPCMA);
CodecInst codec2 = codec1.inst;
++codec2.pltype;
CodecInst test_codec;
// Register the same codec with different payloads.
EXPECT_EQ(0, receiver_->AddCodec(codec1.id, codec1.inst.pltype,
codec1.inst.channels, codec1.inst.plfreq,
nullptr, codec1.inst.plname));
EXPECT_EQ(0, receiver_->AddCodec(codec1.id, codec2.pltype, codec2.channels,
codec2.plfreq, NULL, codec2.plname));
// Both payload types should exist.
EXPECT_EQ(0,
receiver_->DecoderByPayloadType(codec1.inst.pltype, &test_codec));
EXPECT_EQ(true, CodecsEqual(codec1.inst, test_codec));
EXPECT_EQ(0, receiver_->DecoderByPayloadType(codec2.pltype, &test_codec));
EXPECT_EQ(true, CodecsEqual(codec2, test_codec));
}
#if defined(WEBRTC_ANDROID)
#define MAYBE_AddCodecChangeCodecId DISABLED_AddCodecChangeCodecId
#else
#define MAYBE_AddCodecChangeCodecId AddCodecChangeCodecId
#endif
TEST_F(AcmReceiverTestOldApi, AddCodecChangeCodecId) {
const CodecIdInst codec1(RentACodec::CodecId::kPCMU);
CodecIdInst codec2(RentACodec::CodecId::kPCMA);
codec2.inst.pltype = codec1.inst.pltype;
CodecInst test_codec;
// Register the same payload type with different codec ID.
EXPECT_EQ(0, receiver_->AddCodec(codec1.id, codec1.inst.pltype,
codec1.inst.channels, codec1.inst.plfreq,
nullptr, codec1.inst.plname));
EXPECT_EQ(0, receiver_->AddCodec(codec2.id, codec2.inst.pltype,
codec2.inst.channels, codec2.inst.plfreq,
nullptr, codec2.inst.plname));
// Make sure that the last codec is used.
EXPECT_EQ(0,
receiver_->DecoderByPayloadType(codec2.inst.pltype, &test_codec));
EXPECT_EQ(true, CodecsEqual(codec2.inst, test_codec));
}
#if defined(WEBRTC_ANDROID)
#define MAYBE_AddCodecRemoveCodec DISABLED_AddCodecRemoveCodec
#else
#define MAYBE_AddCodecRemoveCodec AddCodecRemoveCodec
#endif
TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecRemoveCodec) {
const CodecIdInst codec(RentACodec::CodecId::kPCMA);
const int payload_type = codec.inst.pltype;
EXPECT_EQ(
0, receiver_->AddCodec(codec.id, codec.inst.pltype, codec.inst.channels,
codec.inst.plfreq, nullptr, codec.inst.plname));
// Remove non-existing codec should not fail. ACM1 legacy.
EXPECT_EQ(0, receiver_->RemoveCodec(payload_type + 1));
// Remove an existing codec.
EXPECT_EQ(0, receiver_->RemoveCodec(payload_type));
// Ask for the removed codec, must fail.
CodecInst ci;
EXPECT_EQ(-1, receiver_->DecoderByPayloadType(payload_type, &ci));
}
#if defined(WEBRTC_ANDROID)
#define MAYBE_SampleRate DISABLED_SampleRate
#else
#define MAYBE_SampleRate SampleRate
#endif
TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
const RentACodec::CodecId kCodecId[] = {RentACodec::CodecId::kISAC,
RentACodec::CodecId::kISACSWB};
AddSetOfCodecs(kCodecId);
AudioFrame frame;
const int kOutSampleRateHz = 8000; // Different than codec sample rate.
for (const auto codec_id : kCodecId) {
const CodecIdInst codec(codec_id);
const int num_10ms_frames = codec.inst.pacsize / (codec.inst.plfreq / 100);
InsertOnePacketOfSilence(codec.id);
for (int k = 0; k < num_10ms_frames; ++k) {
bool muted;
EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame, &muted));
}
EXPECT_EQ(codec.inst.plfreq, receiver_->last_output_sample_rate_hz());
}
}
class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
protected:
AcmReceiverTestFaxModeOldApi() {
config_.neteq_config.playout_mode = kPlayoutFax;
}
void RunVerifyAudioFrame(RentACodec::CodecId codec_id) {
// 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_EQ(kPlayoutFax, config_.neteq_config.playout_mode);
const RentACodec::CodecId kCodecId[] = {codec_id};
AddSetOfCodecs(kCodecId);
const CodecIdInst codec(codec_id);
const int output_sample_rate_hz = codec.inst.plfreq;
const size_t output_channels = codec.inst.channels;
const size_t samples_per_ms = rtc::checked_cast<size_t>(
rtc::CheckedDivExact(output_sample_rate_hz, 1000));
const int num_10ms_frames = rtc::CheckedDivExact(
codec.inst.pacsize, rtc::checked_cast<int>(10 * samples_per_ms));
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) {
InsertOnePacketOfSilence(codec.id);
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(RentACodec::CodecId::kPCMU);
}
#if defined(WEBRTC_ANDROID)
#define MAYBE_VerifyAudioFrameISAC DISABLED_VerifyAudioFrameISAC
#else
#define MAYBE_VerifyAudioFrameISAC VerifyAudioFrameISAC
#endif
TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameISAC) {
RunVerifyAudioFrame(RentACodec::CodecId::kISAC);
}
#if defined(WEBRTC_ANDROID)
#define MAYBE_VerifyAudioFrameOpus DISABLED_VerifyAudioFrameOpus
#else
#define MAYBE_VerifyAudioFrameOpus VerifyAudioFrameOpus
#endif
TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameOpus) {
RunVerifyAudioFrame(RentACodec::CodecId::kOpus);
}
#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);
const CodecIdInst codec(RentACodec::CodecId::kPCM16Bwb);
ASSERT_EQ(
0, receiver_->AddCodec(codec.id, codec.inst.pltype, codec.inst.channels,
codec.inst.plfreq, nullptr, ""));
const int kNumPackets = 5;
const int num_10ms_frames = codec.inst.pacsize / (codec.inst.plfreq / 100);
AudioFrame frame;
for (int n = 0; n < kNumPackets; ++n) {
InsertOnePacketOfSilence(codec.id);
for (int k = 0; k < num_10ms_frames; ++k) {
bool muted;
ASSERT_EQ(0, receiver_->GetAudio(codec.inst.plfreq, &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);
const CodecIdInst codec(RentACodec::CodecId::kPCM16Bwb);
ASSERT_EQ(
0, receiver_->AddCodec(codec.id, codec.inst.pltype, codec.inst.channels,
codec.inst.plfreq, nullptr, ""));
const int kNumPackets = 5;
const int num_10ms_frames = codec.inst.pacsize / (codec.inst.plfreq / 100);
AudioFrame frame;
for (int n = 0; n < kNumPackets; ++n) {
InsertOnePacketOfSilence(codec.id);
for (int k = 0; k < num_10ms_frames; ++k) {
bool muted;
ASSERT_EQ(0, receiver_->GetAudio(codec.inst.plfreq, &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_ISAC)
TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) {
const RentACodec::CodecId kCodecId[] = {
RentACodec::CodecId::kISAC, RentACodec::CodecId::kPCMA,
RentACodec::CodecId::kISACSWB, RentACodec::CodecId::kPCM16Bswb32kHz};
AddSetOfCodecs(kCodecId);
const RentACodec::CodecId kCngId[] = {
// Not including full-band.
RentACodec::CodecId::kCNNB, RentACodec::CodecId::kCNWB,
RentACodec::CodecId::kCNSWB};
AddSetOfCodecs(kCngId);
// Register CNG at sender side.
for (auto id : kCngId)
ASSERT_EQ(0, acm_->RegisterSendCodec(CodecIdInst(id).inst));
CodecInst codec;
// No audio payload is received.
EXPECT_EQ(-1, receiver_->LastAudioCodec(&codec));
// Start with sending DTX.
ASSERT_EQ(0, acm_->SetVAD(true, true, VADVeryAggr));
packet_sent_ = false;
InsertOnePacketOfSilence(CodecIdInst(kCodecId[0]).id); // Enough to test
// with one codec.
ASSERT_TRUE(packet_sent_);
EXPECT_EQ(kAudioFrameCN, last_frame_type_);
// Has received, only, DTX. Last Audio codec is undefined.
EXPECT_EQ(-1, receiver_->LastAudioCodec(&codec));
EXPECT_FALSE(receiver_->last_packet_sample_rate_hz());
for (auto id : kCodecId) {
const CodecIdInst c(id);
// Set DTX off to send audio payload.
acm_->SetVAD(false, false, VADAggr);
packet_sent_ = false;
InsertOnePacketOfSilence(c.id);
// Sanity check if Actually an audio payload received, and it should be
// of type "speech."
ASSERT_TRUE(packet_sent_);
ASSERT_EQ(kAudioFrameSpeech, last_frame_type_);
EXPECT_EQ(c.inst.plfreq, receiver_->last_packet_sample_rate_hz());
// Set VAD on to send DTX. Then check if the "Last Audio codec" returns
// the expected codec.
acm_->SetVAD(true, true, VADAggr);
// Do as many encoding until a DTX is sent.
while (last_frame_type_ != kAudioFrameCN) {
packet_sent_ = false;
InsertOnePacketOfSilence(c.id);
ASSERT_TRUE(packet_sent_);
}
EXPECT_EQ(c.inst.plfreq, receiver_->last_packet_sample_rate_hz());
EXPECT_EQ(0, receiver_->LastAudioCodec(&codec));
EXPECT_TRUE(CodecsEqual(c.inst, codec));
}
}
#endif
} // namespace acm2
} // namespace webrtc