blob: d924275803604de1c3d471d1374fe65c07f3aa0c [file] [log] [blame]
/*
* 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 "rtc_base/bitrate_allocation_strategy.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 {
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,
/*media_transport=*/nullptr);
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::ms(20))
sdp_params["ptime"] =
std::to_string(config.encoder.initial_frame_length.ms());
// 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.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;
}
if (field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
TimeDelta min_frame_length = TimeDelta::ms(20);
// Note, depends on WEBRTC_OPUS_SUPPORT_120MS_PTIME being set, which is
// the default.
TimeDelta max_frame_length = TimeDelta::ms(120);
DataSize rtp_overhead = DataSize::bytes(12);
// Note that this does not include rtp extension overhead and will not
// follow updates in the transport overhead over time.
DataSize total_overhead =
sender_->transport_.packet_overhead() + rtp_overhead;
min_rate += total_overhead / max_frame_length;
// In WebRTCVoiceEngine the max rate is also based on the max frame
// length.
max_rate += total_overhead / min_frame_length;
}
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, 8}};
}
if (config.encoder.priority_rate) {
send_config.track_id = sender->GetNextPriorityId();
sender_->call_->SetBitrateAllocationStrategy(
absl::make_unique<rtc::AudioPriorityBitrateAllocationStrategy>(
send_config.track_id,
config.encoder.priority_rate->bps<uint32_t>()));
}
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_->call_->DestroyAudioSendStream(send_stream_);
}
void SendAudioStream::Start() {
send_stream_->Start();
sender_->call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
}
ColumnPrinter SendAudioStream::StatsPrinter() {
return ColumnPrinter::Lambda(
"audio_target_rate",
[this](rtc::SimpleStringBuilder& 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 = CallTest::kReceiverLocalAudioSsrc;
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, 8}};
}
receiver_->AddExtensions(recv_config.rtp.extensions);
recv_config.decoder_factory = decoder_factory;
recv_config.decoder_map = {
{CallTest::kAudioSendPayloadType, {"opus", 48000, 2}}};
recv_config.sync_group = config.render.sync_group;
receive_stream_ = receiver_->call_->CreateAudioReceiveStream(recv_config);
}
ReceiveAudioStream::~ReceiveAudioStream() {
receiver_->call_->DestroyAudioReceiveStream(receive_stream_);
}
void ReceiveAudioStream::Start() {
receive_stream_->Start();
receiver_->call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
}
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_),
receive_stream_(receiver,
config,
&send_stream_,
decoder_factory,
&receiver->transport_) {}
} // namespace test
} // namespace webrtc