#include <TestStereo.h>
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| | TestPackStereo () |
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| | ~TestPackStereo () |
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| void | RegisterReceiverACM (AudioCodingModule *acm) |
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| int32_t | SendData (const FrameType frame_type, const uint8_t payload_type, const uint32_t timestamp, const uint8_t *payload_data, const size_t payload_size, const RTPFragmentationHeader *fragmentation) override |
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| uint16_t | payload_size () |
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| uint32_t | timestamp_diff () |
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| void | reset_payload_size () |
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| void | set_codec_mode (StereoMonoMode mode) |
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| void | set_lost_packet (bool lost) |
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| | TestPackStereo () |
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| | ~TestPackStereo () |
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| void | RegisterReceiverACM (AudioCodingModule *acm) |
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| int32_t | SendData (const FrameType frame_type, const uint8_t payload_type, const uint32_t timestamp, const uint8_t *payload_data, const size_t payload_size, const RTPFragmentationHeader *fragmentation) override |
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| uint16_t | payload_size () |
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| uint32_t | timestamp_diff () |
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| void | reset_payload_size () |
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| void | set_codec_mode (StereoMonoMode mode) |
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| void | set_lost_packet (bool lost) |
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| virtual | ~AudioPacketizationCallback () |
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| virtual | ~AudioPacketizationCallback () |
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◆ TestPackStereo() [1/2]
| webrtc::TestPackStereo::TestPackStereo |
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◆ ~TestPackStereo() [1/2]
| webrtc::TestPackStereo::~TestPackStereo |
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◆ TestPackStereo() [2/2]
| webrtc::TestPackStereo::TestPackStereo |
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◆ ~TestPackStereo() [2/2]
| webrtc::TestPackStereo::~TestPackStereo |
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◆ payload_size() [1/2]
| uint16_t webrtc::TestPackStereo::payload_size |
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◆ payload_size() [2/2]
| uint16_t webrtc::TestPackStereo::payload_size |
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◆ RegisterReceiverACM() [1/2]
◆ RegisterReceiverACM() [2/2]
◆ reset_payload_size() [1/2]
| void webrtc::TestPackStereo::reset_payload_size |
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◆ reset_payload_size() [2/2]
| void webrtc::TestPackStereo::reset_payload_size |
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◆ SendData() [1/2]
◆ SendData() [2/2]
◆ set_codec_mode() [1/2]
◆ set_codec_mode() [2/2]
◆ set_lost_packet() [1/2]
| void webrtc::TestPackStereo::set_lost_packet |
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bool |
lost | ) |
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◆ set_lost_packet() [2/2]
| void webrtc::TestPackStereo::set_lost_packet |
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bool |
lost | ) |
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◆ timestamp_diff() [1/2]
| uint32_t webrtc::TestPackStereo::timestamp_diff |
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◆ timestamp_diff() [2/2]
| uint32_t webrtc::TestPackStereo::timestamp_diff |
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The documentation for this class was generated from the following files:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_coding/test/TestStereo.h
- Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestStereo.cc