| # Copyright (c) 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. |
| |
| import("//webrtc.gni") |
| |
| if (is_android) { |
| import("//third_party/jni_zero/jni_zero.gni") |
| rtc_android_apk("androidvoip") { |
| testonly = true |
| apk_name = "androidvoip" |
| android_manifest = "AndroidManifest.xml" |
| min_sdk_version = 21 |
| target_sdk_version = 31 |
| |
| sources = [ |
| "java/org/webrtc/examples/androidvoip/MainActivity.java", |
| "java/org/webrtc/examples/androidvoip/OnVoipClientTaskCompleted.java", |
| "java/org/webrtc/examples/androidvoip/VoipClient.java", |
| ] |
| |
| deps = [ |
| ":resources", |
| "//rtc_base:base_java", |
| "//sdk/android:base_java", |
| "//sdk/android:java_audio_device_module_java", |
| "//sdk/android:video_java", |
| "//third_party/androidx:androidx_core_core_java", |
| "//third_party/androidx:androidx_legacy_legacy_support_v4_java", |
| ] |
| |
| shared_libraries = [ ":examples_androidvoip_jni" ] |
| } |
| |
| generate_jni("generated_jni") { |
| testonly = true |
| sources = [ "java/org/webrtc/examples/androidvoip/VoipClient.java" ] |
| namespace = "webrtc_examples" |
| jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h" |
| } |
| |
| rtc_shared_library("examples_androidvoip_jni") { |
| testonly = true |
| sources = [ |
| "jni/android_voip_client.cc", |
| "jni/android_voip_client.h", |
| "jni/onload.cc", |
| ] |
| |
| suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ] |
| configs += [ "//build/config/android:hide_all_but_jni" ] |
| |
| deps = [ |
| ":generated_jni", |
| "../../rtc_base:async_packet_socket", |
| "../../rtc_base:async_udp_socket", |
| "../../rtc_base:logging", |
| "../../rtc_base:network", |
| "../../rtc_base:socket_address", |
| "../../rtc_base:socket_server", |
| "../../rtc_base:ssl", |
| "../../rtc_base:threading", |
| "//api:transport_api", |
| "//api/audio_codecs:audio_codecs_api", |
| "//api/audio_codecs:builtin_audio_decoder_factory", |
| "//api/audio_codecs:builtin_audio_encoder_factory", |
| "//api/task_queue:default_task_queue_factory", |
| "//api/voip:voip_api", |
| "//api/voip:voip_engine_factory", |
| "//rtc_base/third_party/sigslot:sigslot", |
| "//sdk/android:native_api_audio_device_module", |
| "//sdk/android:native_api_base", |
| "//sdk/android:native_api_jni", |
| "//third_party/abseil-cpp/absl/memory:memory", |
| ] |
| } |
| |
| android_resources("resources") { |
| testonly = true |
| custom_package = "org.webrtc.examples.androidvoip" |
| sources = [ |
| "res/layout/activity_main.xml", |
| "res/values/colors.xml", |
| "res/values/strings.xml", |
| ] |
| |
| # Needed for Bazel converter. |
| resource_dirs = [ "res" ] |
| assert(resource_dirs != []) # Mark as used. |
| } |
| } |