webkit
2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
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Protected Member Functions | |
NetEqStereoTestLosses () | |
virtual bool | Lost () |
virtual void | VerifyOutput (size_t num_samples) |
Protected Member Functions inherited from webrtc::NetEqStereoTest | |
NetEqStereoTest () | |
~NetEqStereoTest () | |
virtual void | SetUp () |
virtual void | TearDown () |
int | GetNewPackets () |
virtual int | GetArrivalTime (int send_time) |
void | RunTest (int num_loops) |
Protected Attributes | |
int | frame_index_ |
Protected Attributes inherited from webrtc::NetEqStereoTest | |
const size_t | num_channels_ |
const int | sample_rate_hz_ |
const int | samples_per_ms_ |
const int | frame_size_ms_ |
const size_t | frame_size_samples_ |
const size_t | output_size_samples_ |
NetEq * | neteq_mono_ |
NetEq * | neteq_ |
test::RtpGenerator | rtp_generator_mono_ |
test::RtpGenerator | rtp_generator_ |
int16_t * | input_ |
int16_t * | input_multi_channel_ |
uint8_t * | encoded_ |
uint8_t * | encoded_multi_channel_ |
AudioFrame | output_ |
AudioFrame | output_multi_channel_ |
WebRtcRTPHeader | rtp_header_mono_ |
WebRtcRTPHeader | rtp_header_ |
size_t | payload_size_bytes_ |
size_t | multi_payload_size_bytes_ |
int | last_send_time_ |
int | last_arrival_time_ |
std::unique_ptr< test::InputAudioFile > | input_file_ |
Static Protected Attributes | |
static const int | kLossInterval = 10 |
Static Protected Attributes inherited from webrtc::NetEqStereoTest | |
static const int | kTimeStepMs = 10 |
static const size_t | kMaxBlockSize = 480 |
static const uint8_t | kPayloadTypeMono = 95 |
static const uint8_t | kPayloadTypeMulti = 96 |
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inlineprotected |
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inlineprotectedvirtual |
Reimplemented from webrtc::NetEqStereoTest.
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inlineprotectedvirtual |
Reimplemented from webrtc::NetEqStereoTest.
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protected |
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staticprotected |