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/*
* 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(rtc::ArrayView<const uint8_t> packet,
const webrtc::PacketOptions& options) override;
bool SendRtcp(rtc::ArrayView<const uint8_t> packet) 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_
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