4b133da5fd
RtpFileSource used to implement it's own RTP dump file reader, but with this change it simply uses RtpFileReader. One benefit is that pcap files are now also supported. All NetEq test tools that use RtpFileSource are updated. BUG=2692 R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22839004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7367 4adac7df-926f-26a2-2b94-8c16560cd09d
235 lines
6.7 KiB
Python
235 lines
6.7 KiB
Python
# Copyright (c) 2012 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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'variables': {
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'codecs': [
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'G711',
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'G722',
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'PCM16B',
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'iLBC',
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'iSAC',
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'iSACFix',
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'CNG',
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],
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'neteq_defines': [],
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'conditions': [
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['include_opus==1', {
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'codecs': ['webrtc_opus',],
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'neteq_defines': ['WEBRTC_CODEC_OPUS',],
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}],
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],
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'neteq_dependencies': [
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'<@(codecs)',
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'<(DEPTH)/third_party/opus/opus.gyp:opus',
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'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
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'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
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],
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},
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'targets': [
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{
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'target_name': 'neteq',
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'type': 'static_library',
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'dependencies': [
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'<@(neteq_dependencies)',
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],
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'defines': [
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'<@(neteq_defines)',
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],
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'include_dirs': [
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# Need Opus header files for the audio classifier.
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'<(DEPTH)/third_party/opus/src/celt',
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'<(DEPTH)/third_party/opus/src/src',
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],
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'direct_dependent_settings': {
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'include_dirs': [
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# Need Opus header files for the audio classifier.
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'<(DEPTH)/third_party/opus/src/celt',
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'<(DEPTH)/third_party/opus/src/src',
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],
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},
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'export_dependent_settings': [
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'<(DEPTH)/third_party/opus/opus.gyp:opus',
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],
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'sources': [
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'interface/audio_decoder.h',
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'interface/neteq.h',
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'accelerate.cc',
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'accelerate.h',
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'audio_classifier.cc',
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'audio_classifier.h',
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'audio_decoder_impl.cc',
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'audio_decoder_impl.h',
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'audio_decoder.cc',
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'audio_multi_vector.cc',
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'audio_multi_vector.h',
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'audio_vector.cc',
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'audio_vector.h',
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'background_noise.cc',
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'background_noise.h',
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'buffer_level_filter.cc',
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'buffer_level_filter.h',
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'comfort_noise.cc',
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'comfort_noise.h',
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'decision_logic.cc',
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'decision_logic.h',
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'decision_logic_fax.cc',
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'decision_logic_fax.h',
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'decision_logic_normal.cc',
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'decision_logic_normal.h',
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'decoder_database.cc',
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'decoder_database.h',
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'defines.h',
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'delay_manager.cc',
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'delay_manager.h',
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'delay_peak_detector.cc',
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'delay_peak_detector.h',
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'dsp_helper.cc',
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'dsp_helper.h',
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'dtmf_buffer.cc',
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'dtmf_buffer.h',
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'dtmf_tone_generator.cc',
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'dtmf_tone_generator.h',
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'expand.cc',
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'expand.h',
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'merge.cc',
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'merge.h',
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'neteq_impl.cc',
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'neteq_impl.h',
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'neteq.cc',
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'statistics_calculator.cc',
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'statistics_calculator.h',
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'normal.cc',
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'normal.h',
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'packet_buffer.cc',
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'packet_buffer.h',
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'payload_splitter.cc',
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'payload_splitter.h',
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'post_decode_vad.cc',
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'post_decode_vad.h',
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'preemptive_expand.cc',
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'preemptive_expand.h',
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'random_vector.cc',
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'random_vector.h',
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'rtcp.cc',
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'rtcp.h',
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'sync_buffer.cc',
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'sync_buffer.h',
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'timestamp_scaler.cc',
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'timestamp_scaler.h',
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'time_stretch.cc',
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'time_stretch.h',
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],
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},
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], # targets
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'conditions': [
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['include_tests==1', {
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'includes': ['neteq_tests.gypi',],
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'targets': [
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{
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'target_name': 'audio_decoder_unittests',
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'type': '<(gtest_target_type)',
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'dependencies': [
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'<@(codecs)',
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'<(DEPTH)/testing/gtest.gyp:gtest',
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'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
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'<(webrtc_root)/test/test.gyp:test_support_main',
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],
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'defines': [
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'AUDIO_DECODER_UNITTEST',
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'WEBRTC_CODEC_G722',
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'WEBRTC_CODEC_ILBC',
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'WEBRTC_CODEC_ISACFX',
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'WEBRTC_CODEC_ISAC',
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'WEBRTC_CODEC_PCM16',
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'<@(neteq_defines)',
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],
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'sources': [
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'audio_decoder_impl.cc',
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'audio_decoder_impl.h',
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'audio_decoder_unittest.cc',
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'audio_decoder.cc',
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'interface/audio_decoder.h',
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],
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'conditions': [
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['OS=="android"', {
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'dependencies': [
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'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
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],
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}],
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],
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}, # audio_decoder_unittests
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{
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'target_name': 'neteq_unittest_tools',
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'type': 'static_library',
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'dependencies': [
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'rtp_rtcp',
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'<(webrtc_root)/test/webrtc_test_common.gyp:webrtc_test_common',
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],
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'direct_dependent_settings': {
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'include_dirs': [
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'tools',
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],
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},
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'include_dirs': [
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'tools',
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],
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'sources': [
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'tools/audio_checksum.h',
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'tools/audio_loop.cc',
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'tools/audio_loop.h',
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'tools/audio_sink.h',
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'tools/input_audio_file.cc',
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'tools/input_audio_file.h',
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'tools/output_audio_file.h',
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'tools/packet.cc',
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'tools/packet.h',
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'tools/packet_source.h',
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'tools/rtp_file_source.cc',
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'tools/rtp_file_source.h',
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'tools/rtp_generator.cc',
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'tools/rtp_generator.h',
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],
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}, # neteq_unittest_tools
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], # targets
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'conditions': [
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['OS=="android"', {
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'targets': [
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{
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'target_name': 'audio_decoder_unittests_apk_target',
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'type': 'none',
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'dependencies': [
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'<(apk_tests_path):audio_decoder_unittests_apk',
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],
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},
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],
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}],
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['test_isolation_mode != "noop"', {
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'targets': [
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{
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'target_name': 'audio_decoder_unittests_run',
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'type': 'none',
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'dependencies': [
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'audio_decoder_unittests',
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],
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'includes': [
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'../../../build/isolate.gypi',
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'audio_decoder_unittests.isolate',
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],
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'sources': [
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'audio_decoder_unittests.isolate',
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],
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},
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],
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}],
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],
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}], # include_tests
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], # conditions
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}
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