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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/examples/androidvoip/jni | |
parent | Initial commit. (diff) | |
download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip |
Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/examples/androidvoip/jni')
3 files changed, 731 insertions, 0 deletions
diff --git a/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.cc b/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.cc new file mode 100644 index 0000000000..cf07e87e50 --- /dev/null +++ b/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.cc @@ -0,0 +1,514 @@ +/* + * Copyright 2020 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 "examples/androidvoip/jni/android_voip_client.h" + +#include <errno.h> +#include <sys/socket.h> +#include <algorithm> +#include <map> +#include <memory> +#include <unordered_map> +#include <unordered_set> +#include <utility> +#include <vector> + +#include "absl/memory/memory.h" +#include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "api/audio_codecs/builtin_audio_encoder_factory.h" +#include "api/task_queue/default_task_queue_factory.h" +#include "api/voip/voip_codec.h" +#include "api/voip/voip_engine_factory.h" +#include "api/voip/voip_network.h" +#include "examples/androidvoip/generated_jni/VoipClient_jni.h" +#include "rtc_base/logging.h" +#include "rtc_base/network.h" +#include "rtc_base/socket_server.h" +#include "sdk/android/native_api/audio_device_module/audio_device_android.h" +#include "sdk/android/native_api/jni/java_types.h" +#include "sdk/android/native_api/jni/jvm.h" +#include "sdk/android/native_api/jni/scoped_java_ref.h" + +namespace { + +#define RUN_ON_VOIP_THREAD(method, ...) \ + if (!voip_thread_->IsCurrent()) { \ + voip_thread_->PostTask( \ + std::bind(&AndroidVoipClient::method, this, ##__VA_ARGS__)); \ + return; \ + } \ + RTC_DCHECK_RUN_ON(voip_thread_.get()); + +// Connects a UDP socket to a public address and returns the local +// address associated with it. Since it binds to the "any" address +// internally, it returns the default local address on a multi-homed +// endpoint. Implementation copied from +// BasicNetworkManager::QueryDefaultLocalAddress. +rtc::IPAddress QueryDefaultLocalAddress(int family) { + const char kPublicIPv4Host[] = "8.8.8.8"; + const char kPublicIPv6Host[] = "2001:4860:4860::8888"; + const int kPublicPort = 53; + std::unique_ptr<rtc::Thread> thread = rtc::Thread::CreateWithSocketServer(); + + RTC_DCHECK(thread->socketserver() != nullptr); + RTC_DCHECK(family == AF_INET || family == AF_INET6); + + std::unique_ptr<rtc::Socket> socket( + thread->socketserver()->CreateSocket(family, SOCK_DGRAM)); + if (!socket) { + RTC_LOG_ERR(LS_ERROR) << "Socket creation failed"; + return rtc::IPAddress(); + } + + auto host = family == AF_INET ? kPublicIPv4Host : kPublicIPv6Host; + if (socket->Connect(rtc::SocketAddress(host, kPublicPort)) < 0) { + if (socket->GetError() != ENETUNREACH && + socket->GetError() != EHOSTUNREACH) { + RTC_LOG(LS_INFO) << "Connect failed with " << socket->GetError(); + } + return rtc::IPAddress(); + } + return socket->GetLocalAddress().ipaddr(); +} + +// Assigned payload type for supported built-in codecs. PCMU, PCMA, +// and G722 have set payload types. Whereas opus, ISAC, and ILBC +// have dynamic payload types. +enum class PayloadType : int { + kPcmu = 0, + kPcma = 8, + kG722 = 9, + kOpus = 96, + kIsac = 97, + kIlbc = 98, +}; + +// Returns the payload type corresponding to codec_name. Only +// supports the built-in codecs. +int GetPayloadType(const std::string& codec_name) { + RTC_DCHECK(codec_name == "PCMU" || codec_name == "PCMA" || + codec_name == "G722" || codec_name == "opus" || + codec_name == "ISAC" || codec_name == "ILBC"); + + if (codec_name == "PCMU") { + return static_cast<int>(PayloadType::kPcmu); + } else if (codec_name == "PCMA") { + return static_cast<int>(PayloadType::kPcma); + } else if (codec_name == "G722") { + return static_cast<int>(PayloadType::kG722); + } else if (codec_name == "opus") { + return static_cast<int>(PayloadType::kOpus); + } else if (codec_name == "ISAC") { + return static_cast<int>(PayloadType::kIsac); + } else if (codec_name == "ILBC") { + return static_cast<int>(PayloadType::kIlbc); + } + + RTC_DCHECK_NOTREACHED(); + return -1; +} + +} // namespace + +namespace webrtc_examples { + +void AndroidVoipClient::Init( + JNIEnv* env, + const webrtc::JavaParamRef<jobject>& application_context) { + webrtc::VoipEngineConfig config; + config.encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory(); + config.decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory(); + config.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory(); + config.audio_device_module = + webrtc::CreateJavaAudioDeviceModule(env, application_context.obj()); + config.audio_processing = webrtc::AudioProcessingBuilder().Create(); + + voip_thread_->Start(); + + // Due to consistent thread requirement on + // modules/audio_device/android/audio_device_template.h, + // code is invoked in the context of voip_thread_. + voip_thread_->BlockingCall([this, &config] { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + supported_codecs_ = config.encoder_factory->GetSupportedEncoders(); + env_ = webrtc::AttachCurrentThreadIfNeeded(); + voip_engine_ = webrtc::CreateVoipEngine(std::move(config)); + }); +} + +AndroidVoipClient::~AndroidVoipClient() { + voip_thread_->BlockingCall([this] { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + JavaVM* jvm = nullptr; + env_->GetJavaVM(&jvm); + if (!jvm) { + RTC_LOG(LS_ERROR) << "Failed to retrieve JVM"; + return; + } + jint res = jvm->DetachCurrentThread(); + if (res != JNI_OK) { + RTC_LOG(LS_ERROR) << "DetachCurrentThread failed: " << res; + } + }); + + voip_thread_->Stop(); +} + +AndroidVoipClient* AndroidVoipClient::Create( + JNIEnv* env, + const webrtc::JavaParamRef<jobject>& application_context, + const webrtc::JavaParamRef<jobject>& j_voip_client) { + // Using `new` to access a non-public constructor. + auto voip_client = + absl::WrapUnique(new AndroidVoipClient(env, j_voip_client)); + voip_client->Init(env, application_context); + return voip_client.release(); +} + +void AndroidVoipClient::GetSupportedCodecs(JNIEnv* env) { + RUN_ON_VOIP_THREAD(GetSupportedCodecs, env); + + std::vector<std::string> names; + for (const webrtc::AudioCodecSpec& spec : supported_codecs_) { + names.push_back(spec.format.name); + } + webrtc::ScopedJavaLocalRef<jstring> (*convert_function)( + JNIEnv*, const std::string&) = &webrtc::NativeToJavaString; + Java_VoipClient_onGetSupportedCodecsCompleted( + env_, j_voip_client_, NativeToJavaList(env_, names, convert_function)); +} + +void AndroidVoipClient::GetLocalIPAddress(JNIEnv* env) { + RUN_ON_VOIP_THREAD(GetLocalIPAddress, env); + + std::string local_ip_address; + rtc::IPAddress ipv4_address = QueryDefaultLocalAddress(AF_INET); + if (!ipv4_address.IsNil()) { + local_ip_address = ipv4_address.ToString(); + } else { + rtc::IPAddress ipv6_address = QueryDefaultLocalAddress(AF_INET6); + if (!ipv6_address.IsNil()) { + local_ip_address = ipv6_address.ToString(); + } + } + Java_VoipClient_onGetLocalIPAddressCompleted( + env_, j_voip_client_, webrtc::NativeToJavaString(env_, local_ip_address)); +} + +void AndroidVoipClient::SetEncoder(const std::string& encoder) { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + return; + } + for (const webrtc::AudioCodecSpec& codec : supported_codecs_) { + if (codec.format.name == encoder) { + webrtc::VoipResult result = voip_engine_->Codec().SetSendCodec( + *channel_, GetPayloadType(codec.format.name), codec.format); + RTC_CHECK(result == webrtc::VoipResult::kOk); + return; + } + } +} + +void AndroidVoipClient::SetEncoder( + JNIEnv* env, + const webrtc::JavaParamRef<jstring>& j_encoder_string) { + const std::string& chosen_encoder = + webrtc::JavaToNativeString(env, j_encoder_string); + voip_thread_->PostTask( + [this, chosen_encoder] { SetEncoder(chosen_encoder); }); +} + +void AndroidVoipClient::SetDecoders(const std::vector<std::string>& decoders) { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + return; + } + std::map<int, webrtc::SdpAudioFormat> decoder_specs; + for (const webrtc::AudioCodecSpec& codec : supported_codecs_) { + if (std::find(decoders.begin(), decoders.end(), codec.format.name) != + decoders.end()) { + decoder_specs.insert({GetPayloadType(codec.format.name), codec.format}); + } + } + + webrtc::VoipResult result = + voip_engine_->Codec().SetReceiveCodecs(*channel_, decoder_specs); + RTC_CHECK(result == webrtc::VoipResult::kOk); +} + +void AndroidVoipClient::SetDecoders( + JNIEnv* env, + const webrtc::JavaParamRef<jobject>& j_decoder_strings) { + const std::vector<std::string>& chosen_decoders = + webrtc::JavaListToNativeVector<std::string, jstring>( + env, j_decoder_strings, &webrtc::JavaToNativeString); + voip_thread_->PostTask( + [this, chosen_decoders] { SetDecoders(chosen_decoders); }); +} + +void AndroidVoipClient::SetLocalAddress(const std::string& ip_address, + const int port_number) { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + rtp_local_address_ = rtc::SocketAddress(ip_address, port_number); + rtcp_local_address_ = rtc::SocketAddress(ip_address, port_number + 1); +} + +void AndroidVoipClient::SetLocalAddress( + JNIEnv* env, + const webrtc::JavaParamRef<jstring>& j_ip_address_string, + jint j_port_number_int) { + const std::string& ip_address = + webrtc::JavaToNativeString(env, j_ip_address_string); + voip_thread_->PostTask([this, ip_address, j_port_number_int] { + SetLocalAddress(ip_address, j_port_number_int); + }); +} + +void AndroidVoipClient::SetRemoteAddress(const std::string& ip_address, + const int port_number) { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + rtp_remote_address_ = rtc::SocketAddress(ip_address, port_number); + rtcp_remote_address_ = rtc::SocketAddress(ip_address, port_number + 1); +} + +void AndroidVoipClient::SetRemoteAddress( + JNIEnv* env, + const webrtc::JavaParamRef<jstring>& j_ip_address_string, + jint j_port_number_int) { + const std::string& ip_address = + webrtc::JavaToNativeString(env, j_ip_address_string); + voip_thread_->PostTask([this, ip_address, j_port_number_int] { + SetRemoteAddress(ip_address, j_port_number_int); + }); +} + +void AndroidVoipClient::StartSession(JNIEnv* env) { + RUN_ON_VOIP_THREAD(StartSession, env); + + // CreateChannel guarantees to return valid channel id. + channel_ = voip_engine_->Base().CreateChannel(this, absl::nullopt); + + rtp_socket_.reset(rtc::AsyncUDPSocket::Create(voip_thread_->socketserver(), + rtp_local_address_)); + if (!rtp_socket_) { + RTC_LOG_ERR(LS_ERROR) << "Socket creation failed"; + Java_VoipClient_onStartSessionCompleted(env_, j_voip_client_, + /*isSuccessful=*/false); + return; + } + rtp_socket_->SignalReadPacket.connect( + this, &AndroidVoipClient::OnSignalReadRTPPacket); + + rtcp_socket_.reset(rtc::AsyncUDPSocket::Create(voip_thread_->socketserver(), + rtcp_local_address_)); + if (!rtcp_socket_) { + RTC_LOG_ERR(LS_ERROR) << "Socket creation failed"; + Java_VoipClient_onStartSessionCompleted(env_, j_voip_client_, + /*isSuccessful=*/false); + return; + } + rtcp_socket_->SignalReadPacket.connect( + this, &AndroidVoipClient::OnSignalReadRTCPPacket); + Java_VoipClient_onStartSessionCompleted(env_, j_voip_client_, + /*isSuccessful=*/true); +} + +void AndroidVoipClient::StopSession(JNIEnv* env) { + RUN_ON_VOIP_THREAD(StopSession, env); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + Java_VoipClient_onStopSessionCompleted(env_, j_voip_client_, + /*isSuccessful=*/false); + return; + } + if (voip_engine_->Base().StopSend(*channel_) != webrtc::VoipResult::kOk || + voip_engine_->Base().StopPlayout(*channel_) != webrtc::VoipResult::kOk) { + Java_VoipClient_onStopSessionCompleted(env_, j_voip_client_, + /*isSuccessful=*/false); + return; + } + + rtp_socket_->Close(); + rtcp_socket_->Close(); + + webrtc::VoipResult result = voip_engine_->Base().ReleaseChannel(*channel_); + RTC_CHECK(result == webrtc::VoipResult::kOk); + + channel_ = absl::nullopt; + Java_VoipClient_onStopSessionCompleted(env_, j_voip_client_, + /*isSuccessful=*/true); +} + +void AndroidVoipClient::StartSend(JNIEnv* env) { + RUN_ON_VOIP_THREAD(StartSend, env); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + Java_VoipClient_onStartSendCompleted(env_, j_voip_client_, + /*isSuccessful=*/false); + return; + } + bool sending_started = + (voip_engine_->Base().StartSend(*channel_) == webrtc::VoipResult::kOk); + Java_VoipClient_onStartSendCompleted(env_, j_voip_client_, sending_started); +} + +void AndroidVoipClient::StopSend(JNIEnv* env) { + RUN_ON_VOIP_THREAD(StopSend, env); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + Java_VoipClient_onStopSendCompleted(env_, j_voip_client_, + /*isSuccessful=*/false); + return; + } + bool sending_stopped = + (voip_engine_->Base().StopSend(*channel_) == webrtc::VoipResult::kOk); + Java_VoipClient_onStopSendCompleted(env_, j_voip_client_, sending_stopped); +} + +void AndroidVoipClient::StartPlayout(JNIEnv* env) { + RUN_ON_VOIP_THREAD(StartPlayout, env); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + Java_VoipClient_onStartPlayoutCompleted(env_, j_voip_client_, + /*isSuccessful=*/false); + return; + } + bool playout_started = + (voip_engine_->Base().StartPlayout(*channel_) == webrtc::VoipResult::kOk); + Java_VoipClient_onStartPlayoutCompleted(env_, j_voip_client_, + playout_started); +} + +void AndroidVoipClient::StopPlayout(JNIEnv* env) { + RUN_ON_VOIP_THREAD(StopPlayout, env); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + Java_VoipClient_onStopPlayoutCompleted(env_, j_voip_client_, + /*isSuccessful=*/false); + return; + } + bool playout_stopped = + (voip_engine_->Base().StopPlayout(*channel_) == webrtc::VoipResult::kOk); + Java_VoipClient_onStopPlayoutCompleted(env_, j_voip_client_, playout_stopped); +} + +void AndroidVoipClient::Delete(JNIEnv* env) { + delete this; +} + +void AndroidVoipClient::SendRtpPacket(const std::vector<uint8_t>& packet_copy) { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + if (!rtp_socket_->SendTo(packet_copy.data(), packet_copy.size(), + rtp_remote_address_, rtc::PacketOptions())) { + RTC_LOG(LS_ERROR) << "Failed to send RTP packet"; + } +} + +bool AndroidVoipClient::SendRtp(const uint8_t* packet, + size_t length, + const webrtc::PacketOptions& options) { + std::vector<uint8_t> packet_copy(packet, packet + length); + voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] { + SendRtpPacket(packet_copy); + }); + return true; +} + +void AndroidVoipClient::SendRtcpPacket( + const std::vector<uint8_t>& packet_copy) { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + if (!rtcp_socket_->SendTo(packet_copy.data(), packet_copy.size(), + rtcp_remote_address_, rtc::PacketOptions())) { + RTC_LOG(LS_ERROR) << "Failed to send RTCP packet"; + } +} + +bool AndroidVoipClient::SendRtcp(const uint8_t* packet, size_t length) { + std::vector<uint8_t> packet_copy(packet, packet + length); + voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] { + SendRtcpPacket(packet_copy); + }); + return true; +} + +void AndroidVoipClient::ReadRTPPacket(const std::vector<uint8_t>& packet_copy) { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + return; + } + webrtc::VoipResult result = voip_engine_->Network().ReceivedRTPPacket( + *channel_, + rtc::ArrayView<const uint8_t>(packet_copy.data(), packet_copy.size())); + RTC_CHECK(result == webrtc::VoipResult::kOk); +} + +void AndroidVoipClient::OnSignalReadRTPPacket(rtc::AsyncPacketSocket* socket, + const char* rtp_packet, + size_t size, + const rtc::SocketAddress& addr, + const int64_t& timestamp) { + std::vector<uint8_t> packet_copy(rtp_packet, rtp_packet + size); + voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] { + ReadRTPPacket(packet_copy); + }); +} + +void AndroidVoipClient::ReadRTCPPacket( + const std::vector<uint8_t>& packet_copy) { + RTC_DCHECK_RUN_ON(voip_thread_.get()); + + if (!channel_) { + RTC_LOG(LS_ERROR) << "Channel has not been created"; + return; + } + webrtc::VoipResult result = voip_engine_->Network().ReceivedRTCPPacket( + *channel_, + rtc::ArrayView<const uint8_t>(packet_copy.data(), packet_copy.size())); + RTC_CHECK(result == webrtc::VoipResult::kOk); +} + +void AndroidVoipClient::OnSignalReadRTCPPacket(rtc::AsyncPacketSocket* socket, + const char* rtcp_packet, + size_t size, + const rtc::SocketAddress& addr, + const int64_t& timestamp) { + std::vector<uint8_t> packet_copy(rtcp_packet, rtcp_packet + size); + voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] { + ReadRTCPPacket(packet_copy); + }); +} + +static jlong JNI_VoipClient_CreateClient( + JNIEnv* env, + const webrtc::JavaParamRef<jobject>& application_context, + const webrtc::JavaParamRef<jobject>& j_voip_client) { + return webrtc::NativeToJavaPointer( + AndroidVoipClient::Create(env, application_context, j_voip_client)); +} + +} // namespace webrtc_examples diff --git a/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.h b/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.h new file mode 100644 index 0000000000..8e1edd5ef9 --- /dev/null +++ b/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.h @@ -0,0 +1,189 @@ +/* + * Copyright 2020 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. + */ + +#ifndef EXAMPLES_ANDROIDVOIP_JNI_ANDROID_VOIP_CLIENT_H_ +#define EXAMPLES_ANDROIDVOIP_JNI_ANDROID_VOIP_CLIENT_H_ + +#include <jni.h> + +#include <memory> +#include <string> +#include <vector> + +#include "api/audio_codecs/audio_format.h" +#include "api/call/transport.h" +#include "api/voip/voip_base.h" +#include "api/voip/voip_engine.h" +#include "rtc_base/async_packet_socket.h" +#include "rtc_base/async_udp_socket.h" +#include "rtc_base/socket_address.h" +#include "rtc_base/third_party/sigslot/sigslot.h" +#include "rtc_base/thread.h" +#include "sdk/android/native_api/jni/scoped_java_ref.h" + +namespace webrtc_examples { + +// AndroidVoipClient facilitates the use of the VoIP API defined in +// api/voip/voip_engine.h. One instance of AndroidVoipClient should +// suffice for most VoIP applications. AndroidVoipClient implements +// webrtc::Transport to send RTP/RTCP packets to the remote endpoint. +// It also creates methods (slots) for sockets to connect to in +// order to receive RTP/RTCP packets. AndroidVoipClient does all +// operations with rtc::Thread (voip_thread_), this is to comply +// with consistent thread usage requirement with ProcessThread used +// within VoipEngine, as well as providing asynchronicity to the +// caller. AndroidVoipClient is meant to be used by Java through JNI. +class AndroidVoipClient : public webrtc::Transport, + public sigslot::has_slots<> { + public: + // Returns a pointer to an AndroidVoipClient object. Clients should + // use this factory method to create AndroidVoipClient objects. The + // method will return a nullptr in case of initialization errors. + // It is the client's responsibility to delete the pointer when + // they are done with it (this class provides a Delete() method). + static AndroidVoipClient* Create( + JNIEnv* env, + const webrtc::JavaParamRef<jobject>& application_context, + const webrtc::JavaParamRef<jobject>& j_voip_client); + + ~AndroidVoipClient() override; + + // Provides client with a Java List of Strings containing names of + // the built-in supported codecs through callback. + void GetSupportedCodecs(JNIEnv* env); + + // Provides client with a Java String of the default local IPv4 address + // through callback. If IPv4 address is not found, provide the default + // local IPv6 address. If IPv6 address is not found, provide an empty + // string. + void GetLocalIPAddress(JNIEnv* env); + + // Sets the encoder used by the VoIP API. + void SetEncoder(JNIEnv* env, + const webrtc::JavaParamRef<jstring>& j_encoder_string); + + // Sets the decoders used by the VoIP API. + void SetDecoders(JNIEnv* env, + const webrtc::JavaParamRef<jobject>& j_decoder_strings); + + // Sets two local/remote addresses, one for RTP packets, and another for + // RTCP packets. The RTP address will have IP address j_ip_address_string + // and port number j_port_number_int, the RTCP address will have IP address + // j_ip_address_string and port number j_port_number_int+1. + void SetLocalAddress(JNIEnv* env, + const webrtc::JavaParamRef<jstring>& j_ip_address_string, + jint j_port_number_int); + void SetRemoteAddress( + JNIEnv* env, + const webrtc::JavaParamRef<jstring>& j_ip_address_string, + jint j_port_number_int); + + // Starts a VoIP session, then calls a callback method with a boolean + // value indicating if the session has started successfully. The VoIP + // operations below can only be used after a session has already started. + void StartSession(JNIEnv* env); + + // Stops the current session, then calls a callback method with a + // boolean value indicating if the session has stopped successfully. + void StopSession(JNIEnv* env); + + // Starts sending RTP/RTCP packets to the remote endpoint, then calls + // a callback method with a boolean value indicating if sending + // has started successfully. + void StartSend(JNIEnv* env); + + // Stops sending RTP/RTCP packets to the remote endpoint, then calls + // a callback method with a boolean value indicating if sending + // has stopped successfully. + void StopSend(JNIEnv* env); + + // Starts playing out the voice data received from the remote endpoint, + // then calls a callback method with a boolean value indicating if + // playout has started successfully. + void StartPlayout(JNIEnv* env); + + // Stops playing out the voice data received from the remote endpoint, + // then calls a callback method with a boolean value indicating if + // playout has stopped successfully. + void StopPlayout(JNIEnv* env); + + // Deletes this object. Used by client when they are done. + void Delete(JNIEnv* env); + + // Implementation for Transport. + bool SendRtp(const uint8_t* packet, + size_t length, + const webrtc::PacketOptions& options) override; + bool SendRtcp(const uint8_t* packet, size_t length) override; + + // Slots for sockets to connect to. + void OnSignalReadRTPPacket(rtc::AsyncPacketSocket* socket, + const char* rtp_packet, + size_t size, + const rtc::SocketAddress& addr, + const int64_t& timestamp); + void OnSignalReadRTCPPacket(rtc::AsyncPacketSocket* socket, + const char* rtcp_packet, + size_t size, + const rtc::SocketAddress& addr, + const int64_t& timestamp); + + private: + AndroidVoipClient(JNIEnv* env, + const webrtc::JavaParamRef<jobject>& j_voip_client) + : voip_thread_(rtc::Thread::CreateWithSocketServer()), + j_voip_client_(env, j_voip_client) {} + + void Init(JNIEnv* env, + const webrtc::JavaParamRef<jobject>& application_context); + + // Overloaded methods having native C++ variables as arguments. + void SetEncoder(const std::string& encoder); + void SetDecoders(const std::vector<std::string>& decoders); + void SetLocalAddress(const std::string& ip_address, int port_number); + void SetRemoteAddress(const std::string& ip_address, int port_number); + + // Methods to send and receive RTP/RTCP packets. Takes in a + // copy of a packet as a vector to prolong the lifetime of + // the packet as these methods will be called asynchronously. + void SendRtpPacket(const std::vector<uint8_t>& packet_copy); + void SendRtcpPacket(const std::vector<uint8_t>& packet_copy); + void ReadRTPPacket(const std::vector<uint8_t>& packet_copy); + void ReadRTCPPacket(const std::vector<uint8_t>& packet_copy); + + // Used to invoke operations and send/receive RTP/RTCP packets. + std::unique_ptr<rtc::Thread> voip_thread_; + // Reference to the VoipClient java instance used to + // invoke callbacks when operations are finished. + webrtc::ScopedJavaGlobalRef<jobject> j_voip_client_ + RTC_GUARDED_BY(voip_thread_); + // A list of AudioCodecSpec supported by the built-in + // encoder/decoder factories. + std::vector<webrtc::AudioCodecSpec> supported_codecs_ + RTC_GUARDED_BY(voip_thread_); + // A JNI context used by the voip_thread_. + JNIEnv* env_ RTC_GUARDED_BY(voip_thread_); + // The entry point to all VoIP APIs. + std::unique_ptr<webrtc::VoipEngine> voip_engine_ RTC_GUARDED_BY(voip_thread_); + // Used by the VoIP API to facilitate a VoIP session. + absl::optional<webrtc::ChannelId> channel_ RTC_GUARDED_BY(voip_thread_); + // Members below are used for network related operations. + std::unique_ptr<rtc::AsyncUDPSocket> rtp_socket_ RTC_GUARDED_BY(voip_thread_); + std::unique_ptr<rtc::AsyncUDPSocket> rtcp_socket_ + RTC_GUARDED_BY(voip_thread_); + rtc::SocketAddress rtp_local_address_ RTC_GUARDED_BY(voip_thread_); + rtc::SocketAddress rtcp_local_address_ RTC_GUARDED_BY(voip_thread_); + rtc::SocketAddress rtp_remote_address_ RTC_GUARDED_BY(voip_thread_); + rtc::SocketAddress rtcp_remote_address_ RTC_GUARDED_BY(voip_thread_); +}; + +} // namespace webrtc_examples + +#endif // EXAMPLES_ANDROIDVOIP_JNI_ANDROID_VOIP_CLIENT_H_ diff --git a/third_party/libwebrtc/examples/androidvoip/jni/onload.cc b/third_party/libwebrtc/examples/androidvoip/jni/onload.cc new file mode 100644 index 0000000000..b952de348b --- /dev/null +++ b/third_party/libwebrtc/examples/androidvoip/jni/onload.cc @@ -0,0 +1,28 @@ +/* + * Copyright 2020 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 <jni.h> + +#include "rtc_base/ssl_adapter.h" +#include "sdk/android/native_api/base/init.h" + +namespace webrtc_examples { + +extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM* jvm, void* reserved) { + webrtc::InitAndroid(jvm); + RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()"; + return JNI_VERSION_1_6; +} + +extern "C" void JNIEXPORT JNICALL JNI_OnUnLoad(JavaVM* jvm, void* reserved) { + RTC_CHECK(rtc::CleanupSSL()) << "Failed to CleanupSSL()"; +} + +} // namespace webrtc_examples |