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/*
* Copyright 2023 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 "audio/channel_send.h"
#include <utility>
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/scoped_refptr.h"
#include "api/test/mock_frame_transformer.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "call/rtp_transport_controller_send.h"
#include "rtc_base/gunit.h"
#include "test/gtest.h"
#include "test/mock_transport.h"
#include "test/scoped_key_value_config.h"
#include "test/time_controller/simulated_time_controller.h"
namespace webrtc {
namespace voe {
namespace {
using ::testing::Invoke;
using ::testing::NiceMock;
using ::testing::Return;
using ::testing::SaveArg;
constexpr int kRtcpIntervalMs = 1000;
constexpr int kSsrc = 333;
constexpr int kPayloadType = 1;
constexpr int kSampleRateHz = 48000;
constexpr int kRtpRateHz = 48000;
BitrateConstraints GetBitrateConfig() {
BitrateConstraints bitrate_config;
bitrate_config.min_bitrate_bps = 10000;
bitrate_config.start_bitrate_bps = 100000;
bitrate_config.max_bitrate_bps = 1000000;
return bitrate_config;
}
class ChannelSendTest : public ::testing::Test {
protected:
ChannelSendTest()
: time_controller_(Timestamp::Seconds(1)),
env_(CreateEnvironment(&field_trials_,
time_controller_.GetClock(),
time_controller_.CreateTaskQueueFactory())),
transport_controller_(
RtpTransportConfig{.env = env_,
.bitrate_config = GetBitrateConfig()}) {
channel_ = voe::CreateChannelSend(
time_controller_.GetClock(), time_controller_.GetTaskQueueFactory(),
&transport_, nullptr, &env_.event_log(), nullptr, crypto_options_,
false, kRtcpIntervalMs, kSsrc, nullptr, &transport_controller_,
env_.field_trials());
encoder_factory_ = CreateBuiltinAudioEncoderFactory();
SdpAudioFormat opus = SdpAudioFormat("opus", kRtpRateHz, 2);
std::unique_ptr<AudioEncoder> encoder =
encoder_factory_->MakeAudioEncoder(kPayloadType, opus, {});
channel_->SetEncoder(kPayloadType, opus, std::move(encoder));
transport_controller_.EnsureStarted();
channel_->RegisterSenderCongestionControlObjects(&transport_controller_);
ON_CALL(transport_, SendRtcp).WillByDefault(Return(true));
ON_CALL(transport_, SendRtp).WillByDefault(Return(true));
}
std::unique_ptr<AudioFrame> CreateAudioFrame() {
auto frame = std::make_unique<AudioFrame>();
frame->sample_rate_hz_ = kSampleRateHz;
frame->samples_per_channel_ = kSampleRateHz / 100;
frame->num_channels_ = 1;
frame->set_absolute_capture_timestamp_ms(
time_controller_.GetClock()->TimeInMilliseconds());
return frame;
}
void ProcessNextFrame() {
channel_->ProcessAndEncodeAudio(CreateAudioFrame());
// Advance time to process the task queue.
time_controller_.AdvanceTime(TimeDelta::Millis(10));
}
GlobalSimulatedTimeController time_controller_;
webrtc::test::ScopedKeyValueConfig field_trials_;
Environment env_;
NiceMock<MockTransport> transport_;
CryptoOptions crypto_options_;
RtpTransportControllerSend transport_controller_;
std::unique_ptr<ChannelSendInterface> channel_;
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
};
TEST_F(ChannelSendTest, StopSendShouldResetEncoder) {
channel_->StartSend();
// Insert two frames which should trigger a new packet.
EXPECT_CALL(transport_, SendRtp).Times(1);
ProcessNextFrame();
ProcessNextFrame();
EXPECT_CALL(transport_, SendRtp).Times(0);
ProcessNextFrame();
// StopSend should clear the previous audio frame stored in the encoder.
channel_->StopSend();
channel_->StartSend();
// The following frame should not trigger a new packet since the encoder
// needs 20 ms audio.
EXPECT_CALL(transport_, SendRtp).Times(0);
ProcessNextFrame();
}
TEST_F(ChannelSendTest, IncreaseRtpTimestampByPauseDuration) {
channel_->StartSend();
uint32_t timestamp;
int sent_packets = 0;
auto send_rtp = [&](rtc::ArrayView<const uint8_t> data,
const PacketOptions& options) {
++sent_packets;
RtpPacketReceived packet;
packet.Parse(data);
timestamp = packet.Timestamp();
return true;
};
EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(send_rtp));
ProcessNextFrame();
ProcessNextFrame();
EXPECT_EQ(sent_packets, 1);
uint32_t first_timestamp = timestamp;
channel_->StopSend();
time_controller_.AdvanceTime(TimeDelta::Seconds(10));
channel_->StartSend();
ProcessNextFrame();
ProcessNextFrame();
EXPECT_EQ(sent_packets, 2);
int64_t timestamp_gap_ms =
static_cast<int64_t>(timestamp - first_timestamp) * 1000 / kRtpRateHz;
EXPECT_EQ(timestamp_gap_ms, 10020);
}
TEST_F(ChannelSendTest, FrameTransformerGetsCorrectTimestamp) {
rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
rtc::make_ref_counted<MockFrameTransformer>();
channel_->SetEncoderToPacketizerFrameTransformer(mock_frame_transformer);
rtc::scoped_refptr<TransformedFrameCallback> callback;
EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback)
.WillOnce(SaveArg<0>(&callback));
EXPECT_CALL(*mock_frame_transformer, UnregisterTransformedFrameCallback);
absl::optional<uint32_t> sent_timestamp;
auto send_rtp = [&](rtc::ArrayView<const uint8_t> data,
const PacketOptions& options) {
RtpPacketReceived packet;
packet.Parse(data);
if (!sent_timestamp) {
sent_timestamp = packet.Timestamp();
}
return true;
};
EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(send_rtp));
channel_->StartSend();
int64_t transformable_frame_timestamp = -1;
EXPECT_CALL(*mock_frame_transformer, Transform)
.WillOnce([&](std::unique_ptr<TransformableFrameInterface> frame) {
transformable_frame_timestamp = frame->GetTimestamp();
callback->OnTransformedFrame(std::move(frame));
});
// Insert two frames which should trigger a new packet.
ProcessNextFrame();
ProcessNextFrame();
// Ensure the RTP timestamp on the frame passed to the transformer
// includes the RTP offset and matches the actual RTP timestamp on the sent
// packet.
EXPECT_EQ_WAIT(transformable_frame_timestamp,
0 + channel_->GetRtpRtcp()->StartTimestamp(), 1000);
EXPECT_TRUE_WAIT(sent_timestamp, 1000);
EXPECT_EQ(*sent_timestamp, transformable_frame_timestamp);
}
} // namespace
} // namespace voe
} // namespace webrtc
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