| /dports/databases/cayley/cayley-0.7.5-2-gcf576ba/vendor/google.golang.org/genproto/googleapis/cloud/datalabeling/v1beta1/ |
| H A D | dataset.pb.go | 510 …AudioPayload *AudioPayload `protobuf:"bytes,5,opt,name=audio_payload,json=audioPayload,proto3,oneo… member
549 func (m *DataItem) GetAudioPayload() *AudioPayload {
551 return x.AudioPayload
1034 …AudioPayload *AudioPayload `protobuf:"bytes,8,opt,name=audio_payload,json=audioPayload,proto3,oneo… member
1075 return x.AudioPayload
1324 type AudioPayload struct { struct
1337 func (m *AudioPayload) Reset() { *m = AudioPayload{} } argument
1339 func (*AudioPayload) ProtoMessage() {} argument
1353 func (m *AudioPayload) XXX_Size() int { argument
1356 func (m *AudioPayload) XXX_DiscardUnknown() { argument
[all …]
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| /dports/mail/thunderbird/thunderbird-91.8.0/third_party/libwebrtc/webrtc/modules/rtp_rtcp/include/ |
| H A D | rtp_rtcp_defines.h | 43 struct AudioPayload { struct
56 explicit PayloadUnion(const AudioPayload& payload); argument
67 const AudioPayload& audio_payload() const { in audio_payload()
75 AudioPayload& audio_payload() { in audio_payload()
85 rtc::Optional<AudioPayload> audio_payload_;
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| H A D | rtp_rtcp_defines.cc | 15 PayloadUnion::PayloadUnion(const AudioPayload& payload) in PayloadUnion()
|
| /dports/www/firefox-esr/firefox-91.8.0/third_party/libwebrtc/webrtc/modules/rtp_rtcp/include/ |
| H A D | rtp_rtcp_defines.h | 43 struct AudioPayload { struct
56 explicit PayloadUnion(const AudioPayload& payload); argument
67 const AudioPayload& audio_payload() const { in audio_payload()
75 AudioPayload& audio_payload() { in audio_payload()
85 rtc::Optional<AudioPayload> audio_payload_;
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| H A D | rtp_rtcp_defines.cc | 15 PayloadUnion::PayloadUnion(const AudioPayload& payload) in PayloadUnion()
|
| /dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/ |
| H A D | rtp_rtcp_defines.h | 43 struct AudioPayload { struct
56 explicit PayloadUnion(const AudioPayload& payload); argument
67 const AudioPayload& audio_payload() const { in audio_payload()
75 AudioPayload& audio_payload() { in audio_payload()
85 rtc::Optional<AudioPayload> audio_payload_;
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| H A D | rtp_rtcp_defines.cc | 15 PayloadUnion::PayloadUnion(const AudioPayload& payload) in PayloadUnion()
|
| /dports/mail/thunderbird/thunderbird-91.8.0/third_party/libwebrtc/webrtc/modules/rtp_rtcp/source/ |
| H A D | rtp_receiver_unittest.cc | 89 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
140 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
192 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
266 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
319 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
352 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
435 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
|
| H A D | rtp_receiver_audio.h | 79 const AudioPayload& audio_specific,
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| H A D | rtp_receiver_audio.cc | 208 const AudioPayload& audio_specific, in ParseAudioCodecSpecific()
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| H A D | rtp_sender_audio.cc | 70 PayloadUnion(AudioPayload{ in RegisterAudioPayload()
|
| /dports/www/firefox-esr/firefox-91.8.0/third_party/libwebrtc/webrtc/modules/rtp_rtcp/source/ |
| H A D | rtp_receiver_unittest.cc | 89 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
140 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
192 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
266 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
319 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
352 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
435 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
|
| H A D | rtp_receiver_audio.h | 79 const AudioPayload& audio_specific,
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| H A D | rtp_receiver_audio.cc | 208 const AudioPayload& audio_specific, in ParseAudioCodecSpecific()
|
| H A D | rtp_sender_audio.cc | 70 PayloadUnion(AudioPayload{ in RegisterAudioPayload()
|
| /dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/ |
| H A D | rtp_receiver_unittest.cc | 89 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
140 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
192 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
266 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
319 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
352 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
435 AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}}; in TEST_F()
|
| H A D | rtp_receiver_audio.h | 79 const AudioPayload& audio_specific,
|
| H A D | rtp_receiver_audio.cc | 208 const AudioPayload& audio_specific, in ParseAudioCodecSpecific()
|
| H A D | rtp_sender_audio.cc | 70 PayloadUnion(AudioPayload{ in RegisterAudioPayload()
|
| /dports/www/firefox-legacy/firefox-52.8.0esr/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/ |
| H A D | rtp_rtcp_defines.h | 33 struct AudioPayload
48 AudioPayload Audio;
|
| /dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/ |
| H A D | rtp_rtcp_defines.h | 41 struct AudioPayload { struct
54 AudioPayload Audio; argument
|
| /dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/ |
| H A D | rtp_receiver_audio.h | 79 const AudioPayload& audio_specific,
|
| H A D | rtp_receiver_audio.cc | 213 const AudioPayload& audio_specific, in ParseAudioCodecSpecific()
|
| /dports/www/firefox-legacy/firefox-52.8.0esr/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/ |
| H A D | rtp_receiver_audio.h | 107 const AudioPayload& audio_specific,
|
| H A D | rtp_receiver_audio.cc | 292 const AudioPayload& audio_specific,
|