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Revert of Move fake_audio_device to its own target. (patchset #5 id:80001 of https://codereview.webrtc.org/2717983003/ )

Browse Source 
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: 03d850ddf9

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}
This commit is contained in:
perkj 2017-02-28 00:56:28 -08:00 committed by Commit bot
parent 03d850ddf9
commit 985371bda9
5 changed files with 10 additions and 24 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
View File

@ -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);
}

2
webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
View File

@ -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;
}

8
webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
View File

@ -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;
}

2
webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
View File

@ -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);
}

20
webrtc/test/BUILD.gn
View File

@ -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",