◆ NetEqStereoTest()
webrtc::NetEqStereoTest::NetEqStereoTest |
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◆ ~NetEqStereoTest()
webrtc::NetEqStereoTest::~NetEqStereoTest |
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◆ GetArrivalTime()
virtual int webrtc::NetEqStereoTest::GetArrivalTime |
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int |
send_time | ) |
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◆ GetNewPackets()
int webrtc::NetEqStereoTest::GetNewPackets |
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◆ Lost()
virtual bool webrtc::NetEqStereoTest::Lost |
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◆ RunTest()
void webrtc::NetEqStereoTest::RunTest |
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int |
num_loops | ) |
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◆ SetUp()
virtual void webrtc::NetEqStereoTest::SetUp |
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◆ TearDown()
virtual void webrtc::NetEqStereoTest::TearDown |
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◆ VerifyOutput()
virtual void webrtc::NetEqStereoTest::VerifyOutput |
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size_t |
num_samples | ) |
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◆ encoded_
uint8_t* webrtc::NetEqStereoTest::encoded_ |
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◆ encoded_multi_channel_
uint8_t* webrtc::NetEqStereoTest::encoded_multi_channel_ |
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◆ frame_size_ms_
const int webrtc::NetEqStereoTest::frame_size_ms_ |
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◆ frame_size_samples_
const size_t webrtc::NetEqStereoTest::frame_size_samples_ |
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◆ input_
int16_t* webrtc::NetEqStereoTest::input_ |
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◆ input_file_
◆ input_multi_channel_
int16_t* webrtc::NetEqStereoTest::input_multi_channel_ |
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◆ kMaxBlockSize
const size_t webrtc::NetEqStereoTest::kMaxBlockSize = 480 |
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◆ kPayloadTypeMono
const uint8_t webrtc::NetEqStereoTest::kPayloadTypeMono = 95 |
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◆ kPayloadTypeMulti
const uint8_t webrtc::NetEqStereoTest::kPayloadTypeMulti = 96 |
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◆ kTimeStepMs
const int webrtc::NetEqStereoTest::kTimeStepMs = 10 |
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◆ last_arrival_time_
int webrtc::NetEqStereoTest::last_arrival_time_ |
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◆ last_send_time_
int webrtc::NetEqStereoTest::last_send_time_ |
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◆ multi_payload_size_bytes_
size_t webrtc::NetEqStereoTest::multi_payload_size_bytes_ |
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◆ neteq_
NetEq* webrtc::NetEqStereoTest::neteq_ |
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◆ neteq_mono_
NetEq* webrtc::NetEqStereoTest::neteq_mono_ |
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◆ num_channels_
const size_t webrtc::NetEqStereoTest::num_channels_ |
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◆ output_
◆ output_multi_channel_
AudioFrame webrtc::NetEqStereoTest::output_multi_channel_ |
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◆ output_size_samples_
const size_t webrtc::NetEqStereoTest::output_size_samples_ |
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◆ payload_size_bytes_
size_t webrtc::NetEqStereoTest::payload_size_bytes_ |
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◆ rtp_generator_
◆ rtp_generator_mono_
◆ rtp_header_
◆ rtp_header_mono_
◆ sample_rate_hz_
const int webrtc::NetEqStereoTest::sample_rate_hz_ |
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◆ samples_per_ms_
const int webrtc::NetEqStereoTest::samples_per_ms_ |
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The documentation for this class was generated from the following file: