| /external/webrtc/api/audio_codecs/opus/ |
| D | audio_encoder_multi_channel_opus_config.h | 40 enum class ApplicationMode { kVoip, kAudio }; enumerator
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| D | audio_encoder_opus_config.h | 41 enum class ApplicationMode { kVoip, kAudio }; enumerator
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| /external/webrtc/modules/pacing/ |
| D | pacing_controller_unittest.cc | 263 case RtpPacketMediaType::kAudio: in BuildRtpPacket()
359 MediaStream audio{/*type*/ RtpPacketMediaType::kAudio,
923 Send(RtpPacketMediaType::kAudio, ssrc, sequence_number++, capture_time_ms, in TEST_P()
1022 SendAndExpectPacket(RtpPacketMediaType::kAudio, ssrc, sequence_number++, in TEST_P()
1211 Send(RtpPacketMediaType::kAudio, ssrc_high_priority, sequence_number++, in TEST_P()
1221 Send(RtpPacketMediaType::kAudio, ssrc_high_priority, sequence_number++, in TEST_P()
1705 RtpPacketMediaType::kRetransmission, RtpPacketMediaType::kAudio}) { in TEST_P()
1749 pacer_->EnqueuePacket(BuildRtpPacket(RtpPacketMediaType::kAudio)); in TEST_P()
1770 if (packet->packet_type() == RtpPacketMediaType::kAudio) { in TEST_P()
1869 SendAndExpectPacket(RtpPacketMediaType::kAudio, kSsrc, sequence_number++, in TEST_P()
[all …]
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| D | round_robin_packet_queue.cc | 239 if (single_packet_queue_->Type() == RtpPacketMediaType::kAudio) { in LeadingAudioPacketEnqueueTime()
251 if (top_packet.Type() == RtpPacketMediaType::kAudio) { in LeadingAudioPacketEnqueueTime()
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| D | paced_sender_unittest.cc | 96 case RtpPacketMediaType::kAudio: in BuildRtpPacket()
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| D | task_queue_paced_sender_unittest.cc | 95 case RtpPacketMediaType::kAudio: in BuildRtpPacket()
240 pacer.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kAudio, 1)); in TEST()
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| D | pacing_controller.cc | 65 case RtpPacketMediaType::kAudio: in GetPriorityForType()
693 bool audio_packet = packet_type == RtpPacketMediaType::kAudio; in OnPacketSent()
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| /external/openscreen/cast/streaming/ |
| D | answer_messages.cc | 22 static constexpr char kAudio[] = "audio"; variable
328 if (!AudioConstraints::ParseAndValidate(root[kAudio], &(out->audio)) || in ParseAndValidate()
342 root[kAudio] = audio.ToJson(); in ToJson()
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| /external/webrtc/modules/audio_coding/include/ |
| D | audio_coding_module_typedefs.h | 51 kAudio = 1, enumerator
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_sender_audio.cc | 305 packet->set_packet_type(RtpPacketMediaType::kAudio); in SendAudio()
390 packet->set_packet_type(RtpPacketMediaType::kAudio); in SendTelephoneEventPacket()
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| D | rtp_sender_egress.cc | 234 const bool is_media = packet->packet_type() == RtpPacketMediaType::kAudio || in SendPacket()
387 case RtpPacketMediaType::kAudio: in HasCorrectSsrc()
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| /external/libwebm/webm_parser/src/ |
| D | ancestory.cc | 82 Id::kAudio, in ById()
209 case Id::kAudio: in ById()
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| D | track_entry_parser.h | 53 MakeChild<AudioParser>(Id::kAudio, &TrackEntry::audio), in TrackEntryParser()
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| /external/libwebm/webm_parser/include/webm/ |
| D | id.h | 754 kAudio = 0xE1, enumerator
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| /external/libbrillo/brillo/ |
| D | mime_utils.h | 22 BRILLO_EXPORT extern const char kAudio[]; // audio
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| D | mime_utils.cc | 17 const char mime::types::kAudio[] = "audio"; member in brillo::mime::types
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| /external/libwebm/testing/ |
| D | mkvparser_tests.cc | 159 } else if (track->GetType() == Track::kAudio) { in TEST_F()
422 } else if (track->GetType() == Track::kAudio) { in TEST_F()
472 EXPECT_TRUE(track_type == Track::kVideo || track_type == Track::kAudio); in TEST_F()
522 EXPECT_EQ(Track::kAudio, track->GetType()); in TEST_F()
572 EXPECT_EQ(Track::kAudio, track_type); in TEST_F()
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| /external/webrtc/api/audio_codecs/ |
| D | audio_encoder.h | 176 enum class Application { kSpeech, kAudio }; enumerator
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| /external/libwebm/webm_parser/tests/ |
| D | audio_parser_test.cc | 22 class AudioParserTest : public ElementParserTest<AudioParser, Id::kAudio> {};
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| /external/webrtc/modules/audio_coding/codecs/opus/ |
| D | audio_encoder_opus.cc | 255 : AudioEncoderOpusConfig::ApplicationMode::kAudio; in SdpToConfig()
447 case Application::kAudio: in SetApplication()
448 conf.application = AudioEncoderOpusConfig::ApplicationMode::kAudio; in SetApplication()
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| D | audio_encoder_opus_unittest.cc | 79 : AudioEncoderOpusConfig::ApplicationMode::kAudio; in CreateCodec()
151 EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kAudio, in TEST_P()
169 EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kAudio, in TEST_P()
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| D | audio_encoder_multi_channel_opus_impl.cc | 257 : AudioEncoderMultiChannelOpusConfig::ApplicationMode::kAudio; in SdpToConfig()
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| /external/webrtc/test/pc/e2e/analyzer/audio/ |
| D | default_audio_quality_analyzer.cc | 35 !(*stat->kind == RTCMediaStreamTrackKind::kAudio) || in OnStatsReports()
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| /external/webrtc/modules/rtp_rtcp/source/deprecated/ |
| D | deprecated_rtp_sender_egress.cc | 166 const bool is_media = packet->packet_type() == RtpPacketMediaType::kAudio || in SendPacket()
291 case RtpPacketMediaType::kAudio: in HasCorrectSsrc()
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| /external/libwebm/ |
| D | mkvmuxer_sample.cc | 605 } else if (track_type == Track::kAudio && output_audio) { in main()
717 if ((track_type == Track::kAudio && output_audio) || in main()
738 muxer_frame.set_track_number(track_type == Track::kAudio ? aud_track in main()
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