virtual int32_t OnReceivedPayloadData(
const uint8_t* payloadData,
const uint16_t payloadSize,
- const webrtc::WebRtcRTPHeader* rtpHeader) {
+ const webrtc::WebRtcRTPHeader* rtpHeader) OVERRIDE {
if (rtpHeader->header.payloadType == 98 ||
rtpHeader->header.payloadType == 99) {
EXPECT_EQ(4, payloadSize);
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
const uint8_t channels,
- const uint32_t rate) {
+ const uint32_t rate) OVERRIDE {
if (payloadType == 96) {
EXPECT_EQ(test_rate, rate) <<
"The rate should be 64K for this payloadType";
}
};
-class AudioFeedback : public NullRtpAudioFeedback {
- virtual void OnReceivedTelephoneEvent(const int32_t id,
- const uint8_t event,
- const bool end) {
- static uint8_t expectedEvent = 0;
-
- if (end) {
- uint8_t oldEvent = expectedEvent-1;
- if (expectedEvent == 32) {
- oldEvent = 15;
- }
- EXPECT_EQ(oldEvent, event);
- } else {
- EXPECT_EQ(expectedEvent, event);
- expectedEvent++;
- }
- if (expectedEvent == 16) {
- expectedEvent = 32;
- }
- }
-};
-
class RtpRtcpAudioTest : public ::testing::Test {
protected:
RtpRtcpAudioTest() : fake_clock(123456) {
}
~RtpRtcpAudioTest() {}
- virtual void SetUp() {
- audioFeedback = new AudioFeedback();
+ virtual void SetUp() OVERRIDE {
+ audioFeedback = new NullRtpAudioFeedback();
data_receiver1 = new VerifyingAudioReceiver();
data_receiver2 = new VerifyingAudioReceiver();
rtp_callback = new RTPCallback();
rtp_receiver1_.get(), receive_statistics1_.get());
}
- virtual void TearDown() {
+ virtual void TearDown() OVERRIDE {
delete module1;
delete module2;
delete transport1;
VerifyingAudioReceiver* data_receiver2;
LoopBackTransport* transport1;
LoopBackTransport* transport2;
- AudioFeedback* audioFeedback;
+ NullRtpAudioFeedback* audioFeedback;
RTPCallback* rtp_callback;
uint32_t test_ssrc;
uint32_t test_timestamp;