157 lines
6.4 KiB
C++
157 lines
6.4 KiB
C++
/*
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* Copyright 2020 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef EXAMPLES_ANDROIDVOIP_JNI_ANDROID_VOIP_CLIENT_H_
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#define EXAMPLES_ANDROIDVOIP_JNI_ANDROID_VOIP_CLIENT_H_
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#include <jni.h>
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/audio_codecs/audio_format.h"
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#include "api/call/transport.h"
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#include "api/voip/voip_base.h"
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#include "api/voip/voip_engine.h"
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#include "rtc_base/async_packet_socket.h"
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#include "rtc_base/async_udp_socket.h"
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#include "rtc_base/socket_address.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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#include "rtc_base/thread.h"
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#include "sdk/android/native_api/jni/scoped_java_ref.h"
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namespace webrtc_examples {
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// AndroidVoipClient facilitates the use of the VoIP API defined in
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// api/voip/voip_engine.h. One instance of AndroidVoipClient should
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// suffice for most VoIP applications. AndroidVoipClient implements
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// webrtc::Transport to send RTP/RTCP packets to the remote endpoint.
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// It also creates methods (slots) for sockets to connect to in
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// order to receive RTP/RTCP packets. AndroidVoipClient does all
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// VoipBase related operations with rtc::Thread (voip_thread_), this
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// is to comply with consistent thread usage requirement with
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// ProcessThread used within VoipEngine. AndroidVoipClient is meant
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// to be used by Java through JNI.
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class AndroidVoipClient : public webrtc::Transport,
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public sigslot::has_slots<> {
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public:
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// Returns a pointer to an AndroidVoipClient object. Clients should
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// use this factory method to create AndroidVoipClient objects. The
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// method will return a nullptr in case of initialization errors.
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// It is the client's responsibility to delete the pointer when
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// they are done with it (this class provides a Delete() method).
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static AndroidVoipClient* Create(
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JNIEnv* env,
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const webrtc::JavaParamRef<jobject>& application_context);
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~AndroidVoipClient() override;
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// Returns a Java List of Strings containing names of the built-in
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// supported codecs.
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webrtc::ScopedJavaLocalRef<jobject> GetSupportedCodecs(JNIEnv* env);
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// Returns a Java String of the default local IPv4 address. If IPv4
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// address is not found, returns the default local IPv6 address. If
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// IPv6 address is not found, returns an empty string.
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webrtc::ScopedJavaLocalRef<jstring> GetLocalIPAddress(JNIEnv* env);
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// Sets the encoder used by the VoIP API.
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void SetEncoder(JNIEnv* env,
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const webrtc::JavaRef<jstring>& j_encoder_string);
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// Sets the decoders used by the VoIP API.
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void SetDecoders(JNIEnv* env,
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const webrtc::JavaParamRef<jobject>& j_decoder_strings);
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// Sets two local/remote addresses, one for RTP packets, and another for
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// RTCP packets. The RTP address will have IP address j_ip_address_string
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// and port number j_port_number_int, the RTCP address will have IP address
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// j_ip_address_string and port number j_port_number_int+1.
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void SetLocalAddress(JNIEnv* env,
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const webrtc::JavaRef<jstring>& j_ip_address_string,
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jint j_port_number_int);
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void SetRemoteAddress(JNIEnv* env,
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const webrtc::JavaRef<jstring>& j_ip_address_string,
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jint j_port_number_int);
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// Starts a VoIP session. The VoIP operations below can only be
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// used after a session has already started. Returns true if session
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// started successfully and false otherwise.
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jboolean StartSession(JNIEnv* env);
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// Stops the current session. Returns true if session stopped
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// successfully and false otherwise.
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jboolean StopSession(JNIEnv* env);
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// Starts sending RTP/RTCP packets to the remote endpoint. Returns
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// the return value of StartSend in api/voip/voip_base.h.
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jboolean StartSend(JNIEnv* env);
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// Stops sending RTP/RTCP packets to the remote endpoint. Returns
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// the return value of StopSend in api/voip/voip_base.h.
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jboolean StopSend(JNIEnv* env);
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// Starts playing out the voice data received from the remote endpoint.
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// Returns the return value of StartPlayout in api/voip/voip_base.h.
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jboolean StartPlayout(JNIEnv* env);
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// Stops playing out the voice data received from the remote endpoint.
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// Returns the return value of StopPlayout in api/voip/voip_base.h.
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jboolean StopPlayout(JNIEnv* env);
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// Deletes this object. Used by client when they are done.
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void Delete(JNIEnv* env);
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// Implementation for Transport.
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bool SendRtp(const uint8_t* packet,
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size_t length,
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const webrtc::PacketOptions& options) override;
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bool SendRtcp(const uint8_t* packet, size_t length) override;
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// Slots for sockets to connect to.
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void OnSignalReadRTPPacket(rtc::AsyncPacketSocket* socket,
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const char* rtp_packet,
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size_t size,
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const rtc::SocketAddress& addr,
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const int64_t& timestamp);
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void OnSignalReadRTCPPacket(rtc::AsyncPacketSocket* socket,
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const char* rtcp_packet,
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size_t size,
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const rtc::SocketAddress& addr,
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const int64_t& timestamp);
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private:
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AndroidVoipClient(JNIEnv* env,
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const webrtc::JavaParamRef<jobject>& application_context);
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// Used to invoke VoipBase operations and send/receive
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// RTP/RTCP packets.
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std::unique_ptr<rtc::Thread> voip_thread_;
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// A list of AudioCodecSpec supported by the built-in
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// encoder/decoder factories.
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std::vector<webrtc::AudioCodecSpec> supported_codecs_;
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// The entry point to all VoIP APIs.
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std::unique_ptr<webrtc::VoipEngine> voip_engine_;
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// Used by the VoIP API to facilitate a VoIP session.
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absl::optional<webrtc::ChannelId> channel_;
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// Members below are used for network related operations.
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std::unique_ptr<rtc::AsyncUDPSocket> rtp_socket_;
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std::unique_ptr<rtc::AsyncUDPSocket> rtcp_socket_;
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rtc::SocketAddress rtp_local_address_;
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rtc::SocketAddress rtcp_local_address_;
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rtc::SocketAddress rtp_remote_address_;
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rtc::SocketAddress rtcp_remote_address_;
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};
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} // namespace webrtc_examples
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#endif // EXAMPLES_ANDROIDVOIP_JNI_ANDROID_VOIP_CLIENT_H_
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