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/*
* Copyright 2018 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 "test/scenario/audio_stream.h"
#include "absl/memory/memory.h"
#include "test/call_test.h"
#if WEBRTC_ENABLE_PROTOBUF
RTC_PUSH_IGNORING_WUNDEF()
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
#else
#include "modules/audio_coding/audio_network_adaptor/config.pb.h"
#endif
RTC_POP_IGNORING_WUNDEF()
#endif
namespace webrtc {
namespace test {
namespace {
enum : int { // The first valid value is 1.
kTransportSequenceNumberExtensionId = 1,
kAbsSendTimeExtensionId
};
absl::optional<std::string> CreateAdaptationString(
AudioStreamConfig::NetworkAdaptation config) {
#if WEBRTC_ENABLE_PROTOBUF
audio_network_adaptor::config::ControllerManager cont_conf;
if (config.frame.max_rate_for_60_ms.IsFinite()) {
auto controller =
cont_conf.add_controllers()->mutable_frame_length_controller();
controller->set_fl_decreasing_packet_loss_fraction(
config.frame.min_packet_loss_for_decrease);
controller->set_fl_increasing_packet_loss_fraction(
config.frame.max_packet_loss_for_increase);
controller->set_fl_20ms_to_60ms_bandwidth_bps(
config.frame.min_rate_for_20_ms.bps<int32_t>());
controller->set_fl_60ms_to_20ms_bandwidth_bps(
config.frame.max_rate_for_60_ms.bps<int32_t>());
if (config.frame.max_rate_for_120_ms.IsFinite()) {
controller->set_fl_60ms_to_120ms_bandwidth_bps(
config.frame.min_rate_for_60_ms.bps<int32_t>());
controller->set_fl_120ms_to_60ms_bandwidth_bps(
config.frame.max_rate_for_120_ms.bps<int32_t>());
}
}
cont_conf.add_controllers()->mutable_bitrate_controller();
std::string config_string = cont_conf.SerializeAsString();
return config_string;
#else
RTC_LOG(LS_ERROR) << "audio_network_adaptation is enabled"
" but WEBRTC_ENABLE_PROTOBUF is false.\n"
"Ignoring settings.";
return absl::nullopt;
#endif // WEBRTC_ENABLE_PROTOBUF
}
} // namespace
SendAudioStream::SendAudioStream(
CallClient* sender,
AudioStreamConfig config,
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
Transport* send_transport)
: sender_(sender), config_(config) {
AudioSendStream::Config send_config(send_transport);
ssrc_ = sender->GetNextAudioSsrc();
send_config.rtp.ssrc = ssrc_;
SdpAudioFormat::Parameters sdp_params;
if (config.source.channels == 2)
sdp_params["stereo"] = "1";
if (config.encoder.initial_frame_length != TimeDelta::Millis(20))
sdp_params["ptime"] =
std::to_string(config.encoder.initial_frame_length.ms());
if (config.encoder.enable_dtx)
sdp_params["usedtx"] = "1";
// SdpAudioFormat::num_channels indicates that the encoder is capable of
// stereo, but the actual channel count used is based on the "stereo"
// parameter.
send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
CallTest::kAudioSendPayloadType, {"opus", 48000, 2, sdp_params});
RTC_DCHECK_LE(config.source.channels, 2);
send_config.encoder_factory = encoder_factory;
if (config.encoder.fixed_rate)
send_config.send_codec_spec->target_bitrate_bps =
config.encoder.fixed_rate->bps();
if (!config.adapt.binary_proto.empty()) {
send_config.audio_network_adaptor_config = config.adapt.binary_proto;
} else if (config.network_adaptation) {
send_config.audio_network_adaptor_config =
CreateAdaptationString(config.adapt);
}
if (config.encoder.allocate_bitrate ||
config.stream.in_bandwidth_estimation) {
DataRate min_rate = DataRate::Infinity();
DataRate max_rate = DataRate::Infinity();
if (config.encoder.fixed_rate) {
min_rate = *config.encoder.fixed_rate;
max_rate = *config.encoder.fixed_rate;
} else {
min_rate = *config.encoder.min_rate;
max_rate = *config.encoder.max_rate;
}
send_config.min_bitrate_bps = min_rate.bps();
send_config.max_bitrate_bps = max_rate.bps();
}
if (config.stream.in_bandwidth_estimation) {
send_config.send_codec_spec->transport_cc_enabled = true;
send_config.rtp.extensions = {{RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId}};
}
if (config.stream.abs_send_time) {
send_config.rtp.extensions.push_back(
{RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId});
}
sender_->SendTask([&] {
send_stream_ = sender_->call_->CreateAudioSendStream(send_config);
if (field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
sender->call_->OnAudioTransportOverheadChanged(
sender_->transport_->packet_overhead().bytes());
}
});
}
SendAudioStream::~SendAudioStream() {
sender_->SendTask(
[this] { sender_->call_->DestroyAudioSendStream(send_stream_); });
}
void SendAudioStream::Start() {
sender_->SendTask([this] {
send_stream_->Start();
sender_->call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
});
}
void SendAudioStream::Stop() {
sender_->SendTask([this] { send_stream_->Stop(); });
}
void SendAudioStream::SetMuted(bool mute) {
sender_->SendTask([this, mute] { send_stream_->SetMuted(mute); });
}
ColumnPrinter SendAudioStream::StatsPrinter() {
return ColumnPrinter::Lambda(
"audio_target_rate",
[this](rtc::SimpleStringBuilder& sb) {
sender_->SendTask([this, &sb] {
AudioSendStream::Stats stats = send_stream_->GetStats();
sb.AppendFormat("%.0lf", stats.target_bitrate_bps / 8.0);
});
},
64);
}
ReceiveAudioStream::ReceiveAudioStream(
CallClient* receiver,
AudioStreamConfig config,
SendAudioStream* send_stream,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
Transport* feedback_transport)
: receiver_(receiver), config_(config) {
AudioReceiveStream::Config recv_config;
recv_config.rtp.local_ssrc = receiver_->GetNextAudioLocalSsrc();
recv_config.rtcp_send_transport = feedback_transport;
recv_config.rtp.remote_ssrc = send_stream->ssrc_;
receiver->ssrc_media_types_[recv_config.rtp.remote_ssrc] = MediaType::AUDIO;
if (config.stream.in_bandwidth_estimation) {
recv_config.rtp.transport_cc = true;
recv_config.rtp.extensions = {{RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId}};
}
recv_config.decoder_factory = decoder_factory;
recv_config.decoder_map = {
{CallTest::kAudioSendPayloadType, {"opus", 48000, 2}}};
recv_config.sync_group = config.render.sync_group;
receiver_->SendTask([&] {
receive_stream_ = receiver_->call_->CreateAudioReceiveStream(recv_config);
});
}
ReceiveAudioStream::~ReceiveAudioStream() {
receiver_->SendTask(
[&] { receiver_->call_->DestroyAudioReceiveStream(receive_stream_); });
}
void ReceiveAudioStream::Start() {
receiver_->SendTask([&] {
receive_stream_->Start();
receiver_->call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
});
}
void ReceiveAudioStream::Stop() {
receiver_->SendTask([&] { receive_stream_->Stop(); });
}
AudioReceiveStream::Stats ReceiveAudioStream::GetStats() const {
AudioReceiveStream::Stats result;
receiver_->SendTask([&] {
result = receive_stream_->GetStats(/*get_and_clear_legacy_stats=*/true);
});
return result;
}
AudioStreamPair::~AudioStreamPair() = default;
AudioStreamPair::AudioStreamPair(
CallClient* sender,
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
CallClient* receiver,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
AudioStreamConfig config)
: config_(config),
send_stream_(sender, config, encoder_factory, sender->transport_.get()),
receive_stream_(receiver,
config,
&send_stream_,
decoder_factory,
receiver->transport_.get()) {}
} // namespace test
} // namespace webrtc
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