mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 22:30:40 +01:00
87e2d785a0
This cl deprecates the FrameType enum, and adds aliases AudioFrameType and VideoFrameType. After downstream usage is updated, the enums will be separated and be moved out of common_types.h. Bug: webrtc:6883 Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27011}
378 lines
14 KiB
C++
378 lines
14 KiB
C++
/*
|
|
* 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 "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_(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.
|
|
AudioCodecInfo info = encoder_factory_->QueryAudioEncoder(format).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,
|
|
const RTPFragmentationHeader* fragmentation) override {
|
|
if (frame_type == 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) {
|
|
assert(false);
|
|
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, {"ISAC", 16000, 1}},
|
|
{1, {"ISAC", 32000, 1}}};
|
|
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_VerifyAudioFrameISAC DISABLED_VerifyAudioFrameISAC
|
|
#else
|
|
#define MAYBE_VerifyAudioFrameISAC VerifyAudioFrameISAC
|
|
#endif
|
|
TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameISAC) {
|
|
RunVerifyAudioFrame({"ISAC", 16000, 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);
|
|
encoder_factory_->QueryAudioEncoder(codec).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_ISAC)
|
|
TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) {
|
|
const std::map<int, SdpAudioFormat> codecs = {{0, {"ISAC", 16000, 1}},
|
|
{1, {"PCMA", 8000, 1}},
|
|
{2, {"ISAC", 32000, 1}},
|
|
{3, {"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(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(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_ != 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
|
|
|
|
} // namespace acm2
|
|
|
|
} // namespace webrtc
|