| /external/webrtc/api/transport/rtp/ |
| D | rtp_source.h | 30 absl::optional<uint8_t> audio_level; member
41 absl::optional<uint8_t> audio_level, in RtpSource()
76 absl::optional<uint8_t> audio_level() const { in audio_level() function
|
| /external/webrtc/api/ |
| D | rtp_packet_info.cc | 25 absl::optional<uint8_t> audio_level, in RtpPacketInfo()
|
| D | rtp_packet_info.h | 56 absl::optional<uint8_t> audio_level() const { return audio_level_; } in audio_level() function
|
| /external/webrtc/test/fuzzers/ |
| D | rtp_packet_fuzzer.cc | 75 uint8_t audio_level; in FuzzOneInput() local
|
| /external/webrtc/modules/rtp_rtcp/source/ |
| D | source_tracker.h | 91 absl::optional<uint8_t> audio_level; member
|
| D | rtp_sender_audio_unittest.cc | 126 uint8_t audio_level; in TEST_F() local
|
| D | rtp_packet_unittest.cc | 576 uint8_t audio_level; in TEST() local
608 uint8_t audio_level; in TEST() local
623 uint8_t audio_level; in TEST() local
677 uint8_t audio_level; in TEST() local
|
| D | rtp_header_extensions.cc | 170 uint8_t* audio_level) { in Parse()
181 uint8_t audio_level) { in Write()
|
| D | rtp_header_extensions.h | 72 static size_t ValueSize(bool voice_activity, uint8_t audio_level) { in ValueSize()
|
| /external/webrtc/audio/ |
| D | audio_send_stream.h | 186 int audio_level = 0; member
|
| D | audio_send_stream_tests.cc | 118 uint8_t audio_level = 0; in TEST_F() local
|
| D | audio_send_stream_unittest.cc | 345 std::unique_ptr<AudioFrame> CreateAudioFrame1kHzSineWave(int16_t audio_level, in CreateAudioFrame1kHzSineWave()
|
| /external/webrtc/api/stats/ |
| D | rtcstats_objects.h | 314 RTCStatsMember<double> audio_level; // Receive-only variable
430 RTCStatsMember<double> audio_level; variable
589 RTCStatsMember<double> audio_level; variable
|
| /external/webrtc/call/ |
| D | audio_send_stream.h | 57 int16_t audio_level = 0; member
|
| D | audio_receive_stream.h | 51 int32_t audio_level = -1; member
|
| /external/webrtc/logging/rtc_event_log/encoder/ |
| D | rtc_event_log_encoder_new_format.cc | 434 uint8_t audio_level; in EncodeRtpPacket() local
611 uint8_t audio_level; in EncodeRtpPacket() local
629 uint8_t audio_level; in EncodeRtpPacket() local
|
| /external/webrtc/modules/audio_coding/neteq/tools/ |
| D | neteq_test_factory.h | 104 int audio_level = default_audio_level(); member
|
| /external/webrtc/logging/rtc_event_log/ |
| D | rtc_event_log2.proto | 147 optional uint32 audio_level = 19; field
215 optional uint32 audio_level = 19; field
|
| D | rtc_event_log_parser.cc | 390 const uint8_t audio_level = in StoreRtpPackets() local
591 const uint8_t audio_level = in StoreRtpPackets() local
|
| D | rtc_event_log_unittest_helper.cc | 916 uint8_t audio_level; in VerifyLoggedRtpHeader() local
|
| /external/webrtc/media/base/ |
| D | media_channel.h | 468 int audio_level = 0; member
485 int audio_level = 0; member
|
| /external/webrtc/pc/ |
| D | rtc_stats_collector.cc | 221 double DoubleAudioLevelFromIntAudioLevel(int audio_level) { in DoubleAudioLevelFromIntAudioLevel()
|