#include <rtp_generator.h>
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| TimestampJumpRtpGenerator (int samples_per_ms, uint16_t start_seq_number, uint32_t start_timestamp, uint32_t jump_from_timestamp, uint32_t jump_to_timestamp) |
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uint32_t | GetRtpHeader (uint8_t payload_type, size_t payload_length_samples, WebRtcRTPHeader *rtp_header) override |
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| TimestampJumpRtpGenerator (int samples_per_ms, uint16_t start_seq_number, uint32_t start_timestamp, uint32_t jump_from_timestamp, uint32_t jump_to_timestamp) |
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uint32_t | GetRtpHeader (uint8_t payload_type, size_t payload_length_samples, WebRtcRTPHeader *rtp_header) override |
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| RtpGenerator (int samples_per_ms, uint16_t start_seq_number=0, uint32_t start_timestamp=0, uint32_t start_send_time_ms=0, uint32_t ssrc=0x12345678) |
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virtual | ~RtpGenerator () |
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void | set_drift_factor (double factor) |
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| RtpGenerator (int samples_per_ms, uint16_t start_seq_number=0, uint32_t start_timestamp=0, uint32_t start_send_time_ms=0, uint32_t ssrc=0x12345678) |
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virtual | ~RtpGenerator () |
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void | set_drift_factor (double factor) |
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◆ TimestampJumpRtpGenerator() [1/2]
webrtc::test::TimestampJumpRtpGenerator::TimestampJumpRtpGenerator |
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int |
samples_per_ms, |
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uint16_t |
start_seq_number, |
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uint32_t |
start_timestamp, |
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uint32_t |
jump_from_timestamp, |
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uint32_t |
jump_to_timestamp |
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) |
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inline |
◆ TimestampJumpRtpGenerator() [2/2]
webrtc::test::TimestampJumpRtpGenerator::TimestampJumpRtpGenerator |
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int |
samples_per_ms, |
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uint16_t |
start_seq_number, |
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uint32_t |
start_timestamp, |
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uint32_t |
jump_from_timestamp, |
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uint32_t |
jump_to_timestamp |
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) |
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inline |
◆ GetRtpHeader() [1/2]
◆ GetRtpHeader() [2/2]
The documentation for this class was generated from the following files:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
- Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_generator.cc