From 985371bda999c6db51286586c5850d2ff58f3511 Mon Sep 17 00:00:00 2001 From: perkj Date: Tue, 28 Feb 2017 00:56:28 -0800 Subject: [PATCH] Revert of Move fake_audio_device to its own target. (patchset #5 id:80001 of https://codereview.webrtc.org/2717983003/ ) Reason for revert: Breaks build DEPS. Original issue's description: > Move fake_audio_device to its own target. > The purpose is to make it usefull for test targets that does not need or can use test_common. > > For some reason this also triggered override issues in rtp module tests that are fixed in the same cl. > > BUG=none > > Review-Url: https://codereview.webrtc.org/2717983003 > Cr-Commit-Position: refs/heads/master@{#16889} > Committed: https://chromium.googlesource.com/external/webrtc/+/03d850ddf933e6109caa4d716f0dc69ad78d1014 TBR=ehmaldonado@webrtc.org,danilchap@webrtc.org,kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=none Review-Url: https://codereview.webrtc.org/2718083003 Cr-Commit-Position: refs/heads/master@{#16890} --- .../rtp_rtcp/source/nack_rtx_unittest.cc | 2 +- .../rtp_rtcp/source/rtcp_sender_unittest.cc | 2 +- .../rtp_rtcp/test/testAPI/test_api_audio.cc | 8 ++++---- .../rtp_rtcp/test/testAPI/test_api_rtcp.cc | 2 +- webrtc/test/BUILD.gn | 20 +++---------------- 5 files changed, 10 insertions(+), 24 deletions(-) diff --git a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc index 63378905d1..d772753fa5 100644 --- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc @@ -58,7 +58,7 @@ class TestRtpFeedback : public NullRtpFeedback { explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {} virtual ~TestRtpFeedback() {} - void OnIncomingSSRCChanged(uint32_t ssrc) override { + void OnIncomingSSRCChanged(const uint32_t ssrc) override { rtp_rtcp_->SetRemoteSSRC(ssrc); } diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc index 40b8e061b9..0aa6361e26 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc @@ -211,7 +211,7 @@ class TestTransport : public Transport, return true; } int OnReceivedPayloadData(const uint8_t* payload_data, - size_t payload_size, + const size_t payload_size, const WebRtcRTPHeader* rtp_header) override { return 0; } diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc index 5e5d8eea49..2801c65b90 100644 --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc +++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc @@ -70,11 +70,11 @@ class VerifyingAudioReceiver : public NullRtpData { class RTPCallback : public NullRtpFeedback { public: - int32_t OnInitializeDecoder(int8_t payloadType, + int32_t OnInitializeDecoder(const int8_t payloadType, const char payloadName[RTP_PAYLOAD_NAME_SIZE], - int frequency, - size_t channels, - uint32_t rate) override { + const int frequency, + const size_t channels, + const uint32_t rate) override { EXPECT_EQ(0u, rate) << "The rate should be zero"; return 0; } diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc index 96379283b1..25ff98d7f2 100644 --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc +++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc @@ -56,7 +56,7 @@ class TestRtpFeedback : public NullRtpFeedback { explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {} virtual ~TestRtpFeedback() {} - void OnIncomingSSRCChanged(uint32_t ssrc) override { + void OnIncomingSSRCChanged(const uint32_t ssrc) override { rtp_rtcp_->SetRemoteSSRC(ssrc); } diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn index 12eb3cacc3..07ffbee412 100644 --- a/webrtc/test/BUILD.gn +++ b/webrtc/test/BUILD.gn @@ -333,22 +333,6 @@ rtc_source_set("direct_transport") { ] } -rtc_source_set("fake_audio_device") { - testonly = true - sources = [ - "fake_audio_device.cc", - "fake_audio_device.h", - ] - 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 = [ - "../base:rtc_base_approved", - "../modules/audio_device:audio_device", - ] -} - rtc_source_set("test_common") { testonly = true sources = [ @@ -362,6 +346,8 @@ rtc_source_set("test_common") { "drifting_clock.h", "encoder_settings.cc", "encoder_settings.h", + "fake_audio_device.cc", + "fake_audio_device.h", "fake_decoder.cc", "fake_decoder.h", "fake_encoder.cc", @@ -393,13 +379,13 @@ rtc_source_set("test_common") { deps = [ ":direct_transport", - ":fake_audio_device", ":rtp_test_utils", ":test_support", "..:webrtc_common", "../audio", "../base:rtc_base_approved", "../call", + "../modules/audio_device:mock_audio_device", "../modules/audio_mixer:audio_mixer_impl", "../modules/audio_processing", "../video",