From b637a94b63e45be2c2e6f599ea9b14293f3fd321 Mon Sep 17 00:00:00 2001 From: "henrik.lundin" Date: Fri, 28 Apr 2017 00:59:45 -0700 Subject: [PATCH] NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921} --- webrtc/modules/audio_coding/BUILD.gn | 113 ++++++++++-------- .../neteq/tools/fake_decode_from_file.cc | 11 +- webrtc/modules/audio_processing/BUILD.gn | 2 +- webrtc/test/fuzzers/BUILD.gn | 2 +- 4 files changed, 74 insertions(+), 54 deletions(-) diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn index 8195e47aa6..4c2602ba49 100644 --- a/webrtc/modules/audio_coding/BUILD.gn +++ b/webrtc/modules/audio_coding/BUILD.gn @@ -1105,18 +1105,19 @@ rtc_static_library("neteq") { # rtc_include_tests conditional. The reason is that it supports fuzzer tests # that ultimately are built and run as a part of the Chromium ecosystem, which # does not set the rtc_include_tests flag. -rtc_source_set("neteq_test_minimal") { - testonly = true - - # TODO(kjellander): Remove (bugs.webrtc.org/6828) - # Has cyclic dependency with :neteq_unittest_tools - check_includes = false - +rtc_source_set("neteq_tools_minimal") { sources = [ + "neteq/tools/audio_sink.cc", + "neteq/tools/audio_sink.h", "neteq/tools/encode_neteq_input.cc", "neteq/tools/encode_neteq_input.h", + "neteq/tools/neteq_input.h", "neteq/tools/neteq_test.cc", "neteq/tools/neteq_test.h", + "neteq/tools/packet.cc", + "neteq/tools/packet.h", + "neteq/tools/packet_source.cc", + "neteq/tools/packet_source.h", ] if (!build_with_chromium && is_clang) { @@ -1127,9 +1128,11 @@ rtc_source_set("neteq_test_minimal") { deps = [ ":audio_encoder_interface", ":neteq", + "..:module_api", "../..:webrtc_common", "../../api/audio_codecs:builtin_audio_decoder_factory", "../../base:rtc_base_approved", + "../rtp_rtcp", ] } @@ -1236,7 +1239,7 @@ if (rtc_include_tests) { ] deps = [ ":neteq_test_support", - ":neteq_unittest_tools", + ":neteq_test_tools", ":webrtc_opus", "../..:webrtc_common", "../../base:protobuf_utils", @@ -1265,7 +1268,7 @@ if (rtc_include_tests) { ":audio_format_conversion", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", - ":neteq_unittest_tools", + ":neteq_tools", "../../base:rtc_base_approved", "../../test:test_support", "//testing/gtest", @@ -1285,7 +1288,7 @@ if (rtc_include_tests) { ":audio_coding", "../../api/audio_codecs:audio_codecs_api", ":audio_encoder_interface", - ":neteq_unittest_tools", + ":neteq_tools", "../../base:rtc_base_approved", "../../test:test_support", "//testing/gtest", @@ -1379,7 +1382,7 @@ if (rtc_include_tests) { ":isac", ":isac_fix", ":neteq", - ":neteq_unittest_tools", + ":neteq_tools", "../../api/audio_codecs:audio_codecs_api", "../../base:protobuf_utils", "../../common_audio", @@ -1456,7 +1459,7 @@ if (rtc_include_tests) { deps += [ ":neteq", - ":neteq_unittest_tools", + ":neteq_test_tools", "../..:webrtc_common", "../../base:rtc_base_approved", "../../system_wrappers:system_wrappers_default", @@ -1515,7 +1518,7 @@ if (rtc_include_tests) { deps = [ ":neteq", - ":neteq_unittest_tools", + ":neteq_test_tools", ":pcm16b", "..:module_api", "../..:webrtc_common", @@ -1542,7 +1545,7 @@ if (rtc_include_tests) { deps = [ ":neteq", - ":neteq_unittest_tools", + ":neteq_test_tools", "..:module_api", "../..:webrtc_common", "../../api/audio_codecs:builtin_audio_decoder_factory", @@ -1553,42 +1556,59 @@ if (rtc_include_tests) { ] } - config("neteq_unittest_tools_config") { + config("neteq_tools_config") { include_dirs = [ "tools" ] } - rtc_source_set("neteq_unittest_tools") { - testonly = true + rtc_source_set("neteq_tools") { sources = [ - "neteq/tools/audio_checksum.h", - "neteq/tools/audio_loop.cc", - "neteq/tools/audio_loop.h", - "neteq/tools/audio_sink.cc", - "neteq/tools/audio_sink.h", - "neteq/tools/constant_pcm_packet_source.cc", - "neteq/tools/constant_pcm_packet_source.h", "neteq/tools/fake_decode_from_file.cc", "neteq/tools/fake_decode_from_file.h", "neteq/tools/input_audio_file.cc", "neteq/tools/input_audio_file.h", - "neteq/tools/neteq_input.h", "neteq/tools/neteq_replacement_input.cc", "neteq/tools/neteq_replacement_input.h", - "neteq/tools/output_audio_file.h", - "neteq/tools/output_wav_file.h", - "neteq/tools/packet.cc", - "neteq/tools/packet.h", - "neteq/tools/packet_source.cc", - "neteq/tools/packet_source.h", "neteq/tools/resample_input_audio_file.cc", "neteq/tools/resample_input_audio_file.h", - "neteq/tools/rtp_file_source.cc", - "neteq/tools/rtp_file_source.h", - "neteq/tools/rtp_generator.cc", - "neteq/tools/rtp_generator.h", ] - public_configs = [ ":neteq_unittest_tools_config" ] + public_configs = [ ":neteq_tools_config" ] + + if (!build_with_chromium && is_clang) { + # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). + suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] + } + + deps = [ + "../..:webrtc_common", + "../../api/audio_codecs:audio_codecs_api", + "../../base:rtc_base_approved", + "../../common_audio", + "../rtp_rtcp", + ] + + public_deps = [ + ":neteq_tools_minimal", + ] + } + + rtc_source_set("neteq_test_tools") { + testonly = true + sources = [ + "neteq/tools/audio_checksum.h", + "neteq/tools/audio_loop.cc", + "neteq/tools/audio_loop.h", + "neteq/tools/constant_pcm_packet_source.cc", + "neteq/tools/constant_pcm_packet_source.h", + "neteq/tools/output_audio_file.h", + "neteq/tools/output_wav_file.h", + "neteq/tools/rtp_file_source.cc", + "neteq/tools/rtp_file_source.h", + "neteq/tools/rtp_generator.cc", + "neteq/tools/rtp_generator.h", + ] + + public_configs = [ ":neteq_tools_config" ] if (!build_with_chromium && is_clang) { # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). @@ -1596,11 +1616,9 @@ if (rtc_include_tests) { } deps = [ - ":audio_encoder_interface", ":pcm16b", "..:module_api", "../..:webrtc_common", - "../../api/audio_codecs:audio_codecs_api", "../../base:rtc_base_approved", "../../common_audio", "../../test:rtp_test_utils", @@ -1608,7 +1626,8 @@ if (rtc_include_tests) { ] public_deps = [ - ":neteq_test_minimal", + ":neteq_tools", + ":neteq_tools_minimal", ] if (rtc_enable_protobuf) { @@ -1620,7 +1639,7 @@ if (rtc_include_tests) { } } - rtc_source_set("neteq_test_tools") { + rtc_source_set("neteq_test_tools_deprecated") { testonly = true sources = [ "neteq/test/NETEQTEST_DummyRTPpacket.cc", @@ -1686,7 +1705,7 @@ if (rtc_include_tests) { ":ilbc", ":isac", ":neteq", - ":neteq_test_tools", + ":neteq_test_tools_deprecated", ":pcm16b", ":webrtc_opus", "../..:webrtc_common", @@ -1721,7 +1740,7 @@ if (rtc_include_tests) { ] deps = [ - ":neteq_test_tools", + ":neteq_test_tools_deprecated", ] } @@ -1748,7 +1767,7 @@ if (rtc_include_tests) { ] deps = [ - ":neteq_test_tools", + ":neteq_test_tools_deprecated", "../../test:test_support", "//testing/gtest", ] @@ -1776,7 +1795,7 @@ if (rtc_include_tests) { deps = [ ":neteq", - ":neteq_unittest_tools", + ":neteq_test_tools", ":pcm16b", "../../system_wrappers:system_wrappers_default", "//testing/gtest", @@ -1799,7 +1818,7 @@ if (rtc_include_tests) { deps = [ ":neteq", ":neteq_quality_test_support", - ":neteq_unittest_tools", + ":neteq_tools", ":webrtc_opus", "../../test:test_main", "//testing/gtest", @@ -1835,7 +1854,7 @@ if (rtc_include_tests) { ":ilbc", ":neteq", ":neteq_quality_test_support", - ":neteq_unittest_tools", + ":neteq_tools", "../..:webrtc_common", "../../base:rtc_base_approved", "../../system_wrappers:system_wrappers_default", @@ -2143,7 +2162,7 @@ if (rtc_include_tests) { ":legacy_encoded_audio_frame", ":neteq", ":neteq_test_support", - ":neteq_unittest_tools", + ":neteq_test_tools", ":pcm16b", ":red", ":rent_a_codec", diff --git a/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc b/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc index 2e452e1dea..1cc814fa88 100644 --- a/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc +++ b/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc @@ -29,7 +29,7 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded, RTC_DCHECK_GT(last_decoded_length_, 0); std::fill_n(decoded, last_decoded_length_, 0); *speech_type = kComfortNoise; - return last_decoded_length_; + return rtc::dchecked_cast(last_decoded_length_); } RTC_CHECK_GE(encoded_len, 12); @@ -42,8 +42,8 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded, if (last_decoded_length_ > 0) { // Use length of last decoded packet, but since this is the total for all // channels, we have to divide by 2 in the stereo case. - samples_to_decode = rtc::CheckedDivExact( - last_decoded_length_, static_cast(stereo_ ? 2uL : 1uL)); + samples_to_decode = rtc::dchecked_cast(rtc::CheckedDivExact( + last_decoded_length_, static_cast(stereo_ ? 2uL : 1uL))); } else { // This is the first packet to decode, and we do not know the length of // it. Set it to 10 ms. @@ -70,7 +70,7 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded, std::fill_n(decoded, last_decoded_length_, 0); *speech_type = kComfortNoise; cng_mode_ = true; - return last_decoded_length_; + return rtc::dchecked_cast(last_decoded_length_); } cng_mode_ = false; @@ -83,7 +83,8 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded, } *speech_type = kSpeech; - return last_decoded_length_ = samples_to_decode; + last_decoded_length_ = samples_to_decode; + return rtc::dchecked_cast(last_decoded_length_); } void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp, diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn index ff9a4d6474..93ca85b303 100644 --- a/webrtc/modules/audio_processing/BUILD.gn +++ b/webrtc/modules/audio_processing/BUILD.gn @@ -541,7 +541,7 @@ if (rtc_include_tests) { "../../common_audio:common_audio", "../../system_wrappers:system_wrappers", "../../test:test_support", - "../audio_coding:neteq_unittest_tools", + "../audio_coding:neteq_tools", "test/conversational_speech:unittest", "//testing/gmock", "//testing/gtest", diff --git a/webrtc/test/fuzzers/BUILD.gn b/webrtc/test/fuzzers/BUILD.gn index 74ea607ae2..92603eb9ce 100644 --- a/webrtc/test/fuzzers/BUILD.gn +++ b/webrtc/test/fuzzers/BUILD.gn @@ -258,7 +258,7 @@ webrtc_fuzzer_test("neteq_rtp_fuzzer") { ] deps = [ "../../modules/audio_coding:neteq", - "../../modules/audio_coding:neteq_test_minimal", + "../../modules/audio_coding:neteq_tools_minimal", "../../modules/audio_coding:pcm16b", "../../modules/rtp_rtcp", ]