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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/rtc_event_log/rtc_event_log.h"
#include "api/scoped_refptr.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "call/rtp_transport_controller_send.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 {
constexpr int kRtcpIntervalMs = 1000;
constexpr int kSsrc = 333;
constexpr int kPayloadType = 1;
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;
}
std::unique_ptr<AudioFrame> CreateAudioFrame() {
auto frame = std::make_unique<AudioFrame>();
frame->samples_per_channel_ = 480;
frame->sample_rate_hz_ = 48000;
frame->num_channels_ = 1;
return frame;
}
class ChannelSendTest : public ::testing::Test {
protected:
ChannelSendTest()
: time_controller_(Timestamp::Seconds(1)),
transport_controller_(
time_controller_.GetClock(),
RtpTransportConfig{
.bitrate_config = GetBitrateConfig(),
.event_log = &event_log_,
.task_queue_factory = time_controller_.GetTaskQueueFactory(),
.trials = &field_trials_,
}) {
transport_controller_.EnsureStarted();
}
std::unique_ptr<ChannelSendInterface> CreateChannelSend() {
return voe::CreateChannelSend(
time_controller_.GetClock(), time_controller_.GetTaskQueueFactory(),
&transport_, nullptr, &event_log_, nullptr, crypto_options_, false,
kRtcpIntervalMs, kSsrc, nullptr, nullptr, field_trials_);
}
GlobalSimulatedTimeController time_controller_;
webrtc::test::ScopedKeyValueConfig field_trials_;
RtcEventLogNull event_log_;
MockTransport transport_;
RtpTransportControllerSend transport_controller_;
CryptoOptions crypto_options_;
};
TEST_F(ChannelSendTest, StopSendShouldResetEncoder) {
std::unique_ptr<ChannelSendInterface> channel = CreateChannelSend();
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory =
CreateBuiltinAudioEncoderFactory();
std::unique_ptr<AudioEncoder> encoder = encoder_factory->MakeAudioEncoder(
kPayloadType, SdpAudioFormat("opus", 48000, 2), {});
channel->SetEncoder(kPayloadType, std::move(encoder));
channel->RegisterSenderCongestionControlObjects(&transport_controller_,
nullptr);
channel->StartSend();
// Insert two frames which should trigger a new packet.
EXPECT_CALL(transport_, SendRtp).Times(1);
channel->ProcessAndEncodeAudio(CreateAudioFrame());
time_controller_.AdvanceTime(webrtc::TimeDelta::Zero());
channel->ProcessAndEncodeAudio(CreateAudioFrame());
time_controller_.AdvanceTime(webrtc::TimeDelta::Zero());
EXPECT_CALL(transport_, SendRtp).Times(0);
channel->ProcessAndEncodeAudio(CreateAudioFrame());
time_controller_.AdvanceTime(webrtc::TimeDelta::Zero());
// 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.
channel->ProcessAndEncodeAudio(CreateAudioFrame());
time_controller_.AdvanceTime(webrtc::TimeDelta::Zero());
}
} // namespace
} // namespace voe
} // namespace webrtc
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