| # Copyright (c) 2011 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. |
| |
| LOCAL_PATH := $(call my-dir) |
| |
| include $(CLEAR_VARS) |
| |
| include $(LOCAL_PATH)/../../../../android-webrtc.mk |
| |
| LOCAL_MODULE_CLASS := STATIC_LIBRARIES |
| LOCAL_MODULE := libwebrtc_rtp_rtcp |
| LOCAL_MODULE_TAGS := optional |
| LOCAL_CPP_EXTENSION := .cc |
| LOCAL_GENERATED_SOURCES := |
| LOCAL_SRC_FILES := \ |
| bitrate.cc \ |
| rtp_rtcp_impl.cc \ |
| rtcp_receiver.cc \ |
| rtcp_receiver_help.cc \ |
| rtcp_sender.cc \ |
| rtcp_utility.cc \ |
| rtp_receiver.cc \ |
| rtp_sender.cc \ |
| rtp_utility.cc \ |
| rtp_header_extension.cc \ |
| ssrc_database.cc \ |
| tmmbr_help.cc \ |
| dtmf_queue.cc \ |
| rtp_receiver_audio.cc \ |
| rtp_sender_audio.cc \ |
| bandwidth_management.cc \ |
| forward_error_correction.cc \ |
| forward_error_correction_internal.cc \ |
| overuse_detector.cc \ |
| remote_rate_control.cc \ |
| rtp_packet_history.cc \ |
| receiver_fec.cc \ |
| rtp_receiver_video.cc \ |
| rtp_sender_video.cc \ |
| rtp_format_vp8.cc \ |
| transmission_bucket.cc \ |
| vp8_partition_aggregator.cc |
| |
| # Flags passed to both C and C++ files. |
| LOCAL_CFLAGS := \ |
| $(MY_WEBRTC_COMMON_DEFS) |
| |
| LOCAL_C_INCLUDES := \ |
| $(LOCAL_PATH)/../interface \ |
| $(LOCAL_PATH)/../../.. \ |
| $(LOCAL_PATH)/../../interface \ |
| $(LOCAL_PATH)/../../../system_wrappers/interface |
| |
| LOCAL_SHARED_LIBRARIES := \ |
| libcutils \ |
| libdl \ |
| libstlport |
| |
| ifndef NDK_ROOT |
| include external/stlport/libstlport.mk |
| endif |
| include $(BUILD_STATIC_LIBRARY) |