GN: Add video_engine_tests

Adds separate source_sets for the video_engine_tests subtargets inside
audio, call and video and merges them together into video_engine_tests.

BUG=webrtc:5949
R=kjellander@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/2064523002 .

Cr-Commit-Position: refs/heads/master@{#13127}
This commit is contained in:
Peter Boström
2016-06-14 12:52:54 +02:00
parent 075af92730
commit 0208322ee3
18 changed files with 122 additions and 17 deletions

View File

@ -69,7 +69,7 @@ bool LoopBackTransport::SendRtcp(const uint8_t* data, size_t len) {
int32_t TestRtpReceiver::OnReceivedPayloadData(
const uint8_t* payload_data,
const size_t payload_size,
size_t payload_size,
const webrtc::WebRtcRTPHeader* rtp_header) {
EXPECT_LE(payload_size, sizeof(payload_data_));
memcpy(payload_data_, payload_data, payload_size);

View File

@ -55,7 +55,7 @@ class TestRtpReceiver : public NullRtpData {
public:
int32_t OnReceivedPayloadData(
const uint8_t* payload_data,
const size_t payload_size,
size_t payload_size,
const webrtc::WebRtcRTPHeader* rtp_header) override;
const uint8_t* payload_data() const { return payload_data_; }

View File

@ -28,7 +28,7 @@ class VerifyingAudioReceiver : public NullRtpData {
public:
int32_t OnReceivedPayloadData(
const uint8_t* payloadData,
const size_t payloadSize,
size_t payloadSize,
const webrtc::WebRtcRTPHeader* rtpHeader) override {
if (rtpHeader->header.payloadType == 98 ||
rtpHeader->header.payloadType == 99) {