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diff --git a/third_party/libwebrtc/test/scenario/audio_stream.cc b/third_party/libwebrtc/test/scenario/audio_stream.cc
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+++ b/third_party/libwebrtc/test/scenario/audio_stream.cc
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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
+
+std::vector<RtpExtension> GetAudioRtpExtensions(
+ const AudioStreamConfig& config) {
+ std::vector<RtpExtension> extensions;
+ if (config.stream.in_bandwidth_estimation) {
+ extensions.push_back({RtpExtension::kTransportSequenceNumberUri,
+ kTransportSequenceNumberExtensionId});
+ }
+ if (config.stream.abs_send_time) {
+ extensions.push_back(
+ {RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId});
+ }
+ return extensions;
+}
+
+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;
+
+ bool use_fixed_rate = !config.encoder.min_rate && !config.encoder.max_rate;
+ if (use_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 (use_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 = GetAudioRtpExtensions(config);
+
+ sender_->SendTask([&] {
+ send_stream_ = sender_->call_->CreateAudioSendStream(send_config);
+ 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) {
+ AudioReceiveStreamInterface::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;
+ recv_config.rtp.extensions = GetAudioRtpExtensions(config);
+ 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(); });
+}
+
+AudioReceiveStreamInterface::Stats ReceiveAudioStream::GetStats() const {
+ AudioReceiveStreamInterface::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