/external/webrtc/sdk/android/src/jni/ |
D | jvm.cc | 36 RTC_CHECK(g_jvm) << "JNI_OnLoad failed to run?"; in GetJVM() 44 RTC_CHECK(((env != nullptr) && (status == JNI_OK)) || in GetEnv() 61 RTC_CHECK(GetEnv() == prev_jni_ptr) in ThreadDestructor() 64 RTC_CHECK(status == JNI_OK) << "Failed to detach thread: " << status; in ThreadDestructor() 65 RTC_CHECK(!GetEnv()) << "Detaching was a successful no-op???"; in ThreadDestructor() 69 RTC_CHECK(!pthread_key_create(&g_jni_ptr, &ThreadDestructor)) in CreateJNIPtrKey() 74 RTC_CHECK(!g_jvm) << "InitGlobalJniVariables!"; in InitGlobalJniVariables() 76 RTC_CHECK(g_jvm) << "InitGlobalJniVariables handed NULL?"; in InitGlobalJniVariables() 78 RTC_CHECK(!pthread_once(&g_jni_ptr_once, &CreateJNIPtrKey)) << "pthread_once"; in InitGlobalJniVariables() 110 RTC_CHECK(!pthread_getspecific(g_jni_ptr)) in AttachCurrentThreadIfNeeded() [all …]
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/external/webrtc/modules/audio_processing/test/ |
D | debug_dump_replayer.cc | 83 RTC_CHECK(false); in RunNextEvent() 100 RTC_CHECK(msg.has_num_input_channels()); in OnInitEvent() 101 RTC_CHECK(msg.has_output_sample_rate()); in OnInitEvent() 102 RTC_CHECK(msg.has_num_output_channels()); in OnInitEvent() 103 RTC_CHECK(msg.has_reverse_sample_rate()); in OnInitEvent() 104 RTC_CHECK(msg.has_num_reverse_channels()); in OnInitEvent() 120 RTC_CHECK(apm_.get()); in OnStreamEvent() 150 RTC_CHECK(apm_.get()); in OnReverseStreamEvent() 176 RTC_CHECK(apm_.get()); in OnRuntimeSettingEvent() 183 RTC_CHECK(msg.has_aec_delay_agnostic_enabled()); in MaybeRecreateApm() [all …]
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D | aec_dump_based_simulator.cc | 84 RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface || in PrepareProcessStreamCall() 96 RTC_CHECK(interface_used_ == InterfaceType::kFloatInterface || in PrepareProcessStreamCall() 162 RTC_CHECK(msg.has_level()); in PrepareProcessStreamCall() 182 RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface || in PrepareReverseProcessStreamCall() 193 RTC_CHECK(interface_used_ == InterfaceType::kFloatInterface || in PrepareReverseProcessStreamCall() 252 RTC_CHECK(event_msg.has_init()); in HandleEvent() 256 RTC_CHECK(event_msg.has_stream()); in HandleEvent() 261 RTC_CHECK(event_msg.has_reverse_stream()); in HandleEvent() 265 RTC_CHECK(event_msg.has_config()); in HandleEvent() 272 RTC_CHECK(false); in HandleEvent() [all …]
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/external/webrtc/modules/audio_coding/neteq/tools/ |
D | neteq_rtpplay.cc | 310 RTC_CHECK(ValidateOutputFilesOptions( in main() 314 RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcmu))); in main() 315 RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcma))); in main() 316 RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_ilbc))); in main() 317 RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_isac))); in main() 318 RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_isac_swb))); in main() 319 RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_opus))); in main() 320 RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcm16b))); in main() 321 RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcm16b_wb))); in main() 322 RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcm16b_swb32))); in main() [all …]
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/external/webrtc/common_audio/ |
D | wav_file.cc | 72 RTC_CHECK(file_.is_open()) in WavReader() 77 RTC_CHECK(ReadWavHeader(&readable, &num_channels_, &sample_rate_, &format_, in WavReader() 81 RTC_CHECK(FormatSupported(format_)) << "Non-implemented wav-format"; in WavReader() 85 RTC_CHECK(file_.SeekTo(data_start_pos_)) in Reset() 118 RTC_CHECK(num_samples_read == 0 || (num_bytes_read % num_samples_read) == 0) in ReadSamples() 120 RTC_CHECK(num_samples_read == chunk_size || file_.ReadEof()) in ReadSamples() 164 RTC_CHECK(num_samples_read == 0 || (num_bytes_read % num_samples_read) == 0) in ReadSamples() 166 RTC_CHECK(num_samples_read == chunk_size || file_.ReadEof()) in ReadSamples() 204 RTC_CHECK(file_.is_open()) << "Invalid file. Could not create wav file."; in WavWriter() 206 RTC_CHECK(CheckWavParameters(num_channels_, sample_rate_, format_, in WavWriter() [all …]
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D | wav_header.cc | 135 RTC_CHECK(false); in MapWavFormatToHeaderField() 146 RTC_CHECK(false) << "Unsupported WAV format"; in MapHeaderFieldToWavFormat() 209 RTC_CHECK(buf); in WritePcmWavHeader() 210 RTC_CHECK(header_size); in WritePcmWavHeader() 240 RTC_CHECK(buf); in WriteIeeeFloatWavHeader() 241 RTC_CHECK(header_size); in WriteIeeeFloatWavHeader() 282 RTC_CHECK(false); in GetFormatBytesPerSample() 359 RTC_CHECK(buf); in WriteWavHeader() 360 RTC_CHECK(header_size); in WriteWavHeader() 363 RTC_CHECK(CheckWavParameters(num_channels, sample_rate, format, in WriteWavHeader()
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/external/webrtc/test/ |
D | rtp_file_writer.cc | 32 RTC_CHECK(file_ != NULL); in RtpDumpWriter() 46 RTC_CHECK(WriteUint16(len)); in WritePacket() 47 RTC_CHECK(WriteUint16(plen)); in WritePacket() 48 RTC_CHECK(WriteUint32(offset)); in WritePacket() 57 RTC_CHECK(WriteUint32(0)); in Init() 58 RTC_CHECK(WriteUint32(0)); in Init() 59 RTC_CHECK(WriteUint32(0)); in Init() 60 RTC_CHECK(WriteUint16(0)); in Init() 61 RTC_CHECK(WriteUint16(0)); in Init()
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/external/webrtc/modules/audio_coding/audio_network_adaptor/ |
D | controller_manager.cc | 49 RTC_CHECK(config.has_fec_enabling_threshold()); in CreateFecControllerPlrBased() 50 RTC_CHECK(config.has_fec_disabling_threshold()); in CreateFecControllerPlrBased() 51 RTC_CHECK(config.has_time_constant_ms()); in CreateFecControllerPlrBased() 54 RTC_CHECK(fec_enabling_threshold.has_low_bandwidth_bps()); in CreateFecControllerPlrBased() 55 RTC_CHECK(fec_enabling_threshold.has_low_bandwidth_packet_loss()); in CreateFecControllerPlrBased() 56 RTC_CHECK(fec_enabling_threshold.has_high_bandwidth_bps()); in CreateFecControllerPlrBased() 57 RTC_CHECK(fec_enabling_threshold.has_high_bandwidth_packet_loss()); in CreateFecControllerPlrBased() 60 RTC_CHECK(fec_disabling_threshold.has_low_bandwidth_bps()); in CreateFecControllerPlrBased() 61 RTC_CHECK(fec_disabling_threshold.has_low_bandwidth_packet_loss()); in CreateFecControllerPlrBased() 62 RTC_CHECK(fec_disabling_threshold.has_high_bandwidth_bps()); in CreateFecControllerPlrBased() [all …]
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/external/webrtc/test/pc/e2e/ |
D | peer_configurer.cc | 118 RTC_CHECK(p->name); in ValidateParams() 120 RTC_CHECK(inserted) << "Duplicate name=" << p->name.value(); in ValidateParams() 131 RTC_CHECK(video_config.stream_label); in ValidateParams() 134 RTC_CHECK(inserted) << "Duplicate video_config.stream_label=" in ValidateParams() 142 RTC_CHECK(sync_group_inserted) in ValidateParams() 157 RTC_CHECK(!video_config.max_encode_bitrate_bps) in ValidateParams() 159 RTC_CHECK(!video_config.min_encode_bitrate_bps) in ValidateParams() 173 RTC_CHECK(inserted) << "Duplicate audio_config.stream_label=" in ValidateParams() 181 RTC_CHECK(sync_group_inserted) in ValidateParams() 188 RTC_CHECK(!p->audio_config.value().input_file_name); in ValidateParams() [all …]
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/external/webrtc/modules/rtp_rtcp/source/ |
D | rtcp_transceiver.cc | 56 RTC_CHECK(rtcp_transceiver_); in AddMediaReceiverRtcpObserver() 67 RTC_CHECK(rtcp_transceiver_); in RemoveMediaReceiverRtcpObserver() 76 RTC_CHECK(rtcp_transceiver_); in SetReadyToSend() 83 RTC_CHECK(rtcp_transceiver_); in ReceivePacket() 91 RTC_CHECK(rtcp_transceiver_); in SendCompoundPacket() 98 RTC_CHECK(rtcp_transceiver_); in SetRemb() 107 RTC_CHECK(rtcp_transceiver_); in UnsetRemb() 114 RTC_CHECK(rtcp_transceiver_); in SendCombinedRtcpPacket() 124 RTC_CHECK(rtcp_transceiver_); in SendNack() 133 RTC_CHECK(rtcp_transceiver_); in SendPictureLossIndication() [all …]
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/external/webrtc/sdk/android/src/jni/pc/ |
D | ice_candidate.cc | 57 RTC_CHECK(!sdp.empty()) << "got an empty ICE candidate"; in NativeToJavaCandidate() 68 RTC_CHECK(candidate.ToString(&sdp)) << "got so far: " << sdp; in NativeToJavaIceCandidate() 99 RTC_CHECK(false) << "Unexpected IceTransportsType enum_name " << enum_name; in JavaToNativeIceTransportsType() 117 RTC_CHECK(false) << "Unexpected BundlePolicy enum_name " << enum_name; in JavaToNativeBundlePolicy() 132 RTC_CHECK(false) << "Unexpected RtcpMuxPolicy enum_name " << enum_name; in JavaToNativeRtcpMuxPolicy() 147 RTC_CHECK(false) << "Unexpected TcpCandidatePolicy enum_name " << enum_name; in JavaToNativeTcpCandidatePolicy() 163 RTC_CHECK(false) << "Unexpected CandidateNetworkPolicy enum_name " in JavaToNativeCandidateNetworkPolicy() 177 RTC_CHECK(false) << "Unexpected KeyType enum_name " << enum_name; in JavaToNativeKeyType() 192 RTC_CHECK(false) << "Unexpected ContinualGatheringPolicy enum name " in JavaToNativeContinualGatheringPolicy() 211 RTC_CHECK(false) << " Unexpected PortPrunePolicy enum name " << enum_name; in JavaToNativePortPrunePolicy() [all …]
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/external/webrtc/modules/utility/source/ |
D | helpers_android.cc | 30 RTC_CHECK(((env != NULL) && (status == JNI_OK)) || in GetEnv() 57 RTC_CHECK(m) << name << ", " << signature; in GetMethodID() 68 RTC_CHECK(m) << name << ", " << signature; in GetStaticMethodID() 75 RTC_CHECK(c) << name; in FindClass() 82 RTC_CHECK(ret); in NewGlobalRef() 101 RTC_CHECK(attached_) << "AttachCurrentThread failed: " << res; in AttachThreadScoped() 109 RTC_CHECK(res == JNI_OK) << "DetachCurrentThread failed: " << res; in ~AttachThreadScoped() 110 RTC_CHECK(!GetEnv(jvm_)); in ~AttachThreadScoped()
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/external/webrtc/rtc_base/ |
D | checks_unittest.cc | 17 RTC_CHECK(true) << "i=" << ++i; in TEST() 40 RTC_CHECK(true) << "Shouldn't crash" << 1; in TEST() 41 EXPECT_DEATH(RTC_CHECK(false) << "Hi there!", in TEST() 64 RTC_CHECK(true) << "Shouldn't crash" << 1; in TEST() 65 EXPECT_DEATH(RTC_CHECK(false) << "Hi there!", in TEST()
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D | openssl_utility_unittest.cc | 179 RTC_CHECK(x509); in CreateSSLWithPeerCertificate() 185 RTC_CHECK(key); in CreateSSLWithPeerCertificate() 193 RTC_CHECK(SSL_use_certificate(server, x509)); in CreateSSLWithPeerCertificate() 194 RTC_CHECK(SSL_use_PrivateKey(server, key)); in CreateSSLWithPeerCertificate() 206 RTC_CHECK(client_err == SSL_ERROR_NONE || in CreateSSLWithPeerCertificate() 212 RTC_CHECK(server_err == SSL_ERROR_NONE || in CreateSSLWithPeerCertificate()
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D | rtc_certificate_generator_unittest.cc | 30 RTC_CHECK(signaling_thread_); in RTCCertificateGeneratorFixture() 31 RTC_CHECK(worker_thread_->Start()); in RTCCertificateGeneratorFixture() 41 RTC_CHECK(signaling_thread_->IsCurrent()); in OnSuccess() 42 RTC_CHECK(certificate); in OnSuccess() 47 RTC_CHECK(signaling_thread_->IsCurrent()); in OnFailure() 53 RTC_CHECK(signaling_thread_->IsCurrent()); in GenerateAsyncCompleted()
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/external/webrtc/examples/unityplugin/ |
D | class_reference_holder.cc | 38 RTC_CHECK(g_class_reference_holder == nullptr); in LoadGlobalClassReferenceHolder() 54 RTC_CHECK(classes_.empty()) << "Must call FreeReferences() before dtor!"; in ~ClassReferenceHolder() 67 RTC_CHECK(it != classes_.end()) << "Unexpected GetClass() call for: " << name; in GetClass() 74 RTC_CHECK(localRef) << name; in LoadClass() 77 RTC_CHECK(globalRef) << name; in LoadClass() 79 RTC_CHECK(inserted) << "Duplicate class name: " << name; in LoadClass()
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/external/webrtc/pc/test/ |
D | fake_data_channel_provider.h | 32 RTC_CHECK(ready_to_send_); in SendData() 33 RTC_CHECK(transport_available_); in SendData() 49 RTC_CHECK(connected_channels_.find(data_channel) == in ConnectDataChannel() 60 RTC_CHECK(connected_channels_.find(data_channel) != in DisconnectDataChannel() 67 RTC_CHECK(sid >= 0); in AddSctpDataStream() 76 RTC_CHECK(sid >= 0); in RemoveSctpDataStream() 117 RTC_CHECK(transport_available_); in set_ready_to_send()
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/external/webrtc/test/network/ |
D | fake_network_socket_server.cc | 123 RTC_CHECK(local_addr_.IsNil()) in Bind() 147 RTC_CHECK(remote_addr_.IsNil()) in Connect() 149 RTC_CHECK(!local_addr_.IsNil()) in Connect() 158 RTC_CHECK(state_ == CS_CONNECTED) << "Socket cannot send: not connected"; in Send() 166 RTC_CHECK(!local_addr_.IsNil()) in SendTo() 193 RTC_CHECK(pending_); in RecvFrom() 203 RTC_CHECK(data_read == pending_->size()) in RecvFrom() 217 RTC_CHECK(false) << "Listen() isn't valid for SOCK_DGRAM"; in Listen() 221 RTC_CHECK(false) << "Accept() isn't valid for SOCK_DGRAM"; in Accept() 242 RTC_CHECK(error == 0); in SetError() [all …]
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D | network_emulation_manager.cc | 88 RTC_CHECK(ip) << "All auto generated IPv4 addresses exhausted"; in CreateEndpoint() 92 RTC_CHECK(ip) << "All auto generated IPv6 addresses exhausted"; in CreateEndpoint() 99 RTC_CHECK(res) << "IP=" << ip->ToString() << " already in use"; in CreateEndpoint() 111 RTC_CHECK(network_manager); in EnableEndpoint() 118 RTC_CHECK(network_manager); in DisableEndpoint() 129 RTC_CHECK(!via_nodes.empty()); in CreateRoute() 156 RTC_CHECK(route->active) << "Route already cleared"; in ClearRoute() 173 RTC_CHECK(!via_nodes.empty()); in CreateTrafficRoute() 286 RTC_CHECK(insertion_result) in CreateEmulatedNetworkManagerInterface()
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/external/webrtc/api/voip/ |
D | voip_engine_factory.cc | 21 RTC_CHECK(config.encoder_factory); in CreateVoipEngine() 22 RTC_CHECK(config.decoder_factory); in CreateVoipEngine() 23 RTC_CHECK(config.task_queue_factory); in CreateVoipEngine() 24 RTC_CHECK(config.audio_device_module); in CreateVoipEngine()
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/external/webrtc/test/fuzzers/ |
D | rtp_dependency_descriptor_fuzzer.cc | 50 RTC_CHECK(structure1_ptr); in FuzzOneInput() 62 RTC_CHECK(RtpDependencyDescriptorExtension::Write(write_buffer, structure1, in FuzzOneInput() 68 RTC_CHECK(RtpDependencyDescriptorExtension::Parse( in FuzzOneInput() 85 RTC_CHECK(descriptor1.resolution == descriptor2.resolution); in FuzzOneInput() 86 RTC_CHECK(descriptor1.frame_dependencies == descriptor2.frame_dependencies); in FuzzOneInput()
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D | rtp_packetizer_av1_fuzzer.cc | 49 RTC_CHECK(packetizer.NextPacket(&rtp_packet)); in FuzzOneInput() 55 RTC_CHECK(packetizer.NextPacket(&rtp_packet)); in FuzzOneInput() 60 RTC_CHECK(packetizer.NextPacket(&rtp_packet)) in FuzzOneInput() 66 RTC_CHECK(packetizer.NextPacket(&rtp_packet)); in FuzzOneInput()
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/external/webrtc/modules/video_coding/codecs/test/ |
D | videoprocessor.cc | 60 RTC_CHECK(!nalu_indices.empty()); in GetMaxNaluSizeBytes() 198 RTC_CHECK(TaskQueueBase::Current()) in VideoProcessor() 200 RTC_CHECK(stats_); in VideoProcessor() 201 RTC_CHECK(encoder_); in VideoProcessor() 202 RTC_CHECK(decoders_); in VideoProcessor() 204 RTC_CHECK(input_frame_reader_); in VideoProcessor() 205 RTC_CHECK(encoded_frame_writers_); in VideoProcessor() 206 RTC_CHECK(!decoded_frame_writers || in VideoProcessor() 257 RTC_CHECK(buffer) << "Tried to read too many frames from the file."; in ProcessFrame() 363 RTC_CHECK(first_encoded_frame_[spatial_idx] || in FrameEncoded() [all …]
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/external/webrtc/test/pc/e2e/analyzer/video/ |
D | single_process_encoded_image_data_injector.cc | 33 RTC_CHECK(source.size() >= ExtractionInfo::kUsedBufferSize); in InjectData() 90 RTC_CHECK(!id || *id == next_id) in ExtractData() 100 RTC_CHECK(ext_vector_it != extraction_cache_.end()) in ExtractData() 104 RTC_CHECK(info_it != ext_vector_it->second.infos.end()) in ExtractData() 116 RTC_CHECK(id); in ExtractData() 131 RTC_CHECK(pos < size); in ExtractData()
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/external/webrtc/modules/pacing/ |
D | round_robin_packet_queue.cc | 185 RTC_CHECK(queued_packet.EnqueueTimeIterator() != enqueue_times_.end()); in Pop() 201 RTC_CHECK(size_packets_ > 0 || queue_time_sum_ == TimeDelta::Zero()); in Pop() 207 RTC_CHECK(!IsSsrcScheduled(stream->ssrc)); in Pop() 264 RTC_CHECK(!enqueue_times_.empty()); in OldestEnqueueTime() 335 RTC_CHECK(!IsSsrcScheduled(stream->ssrc)); in Push() 346 RTC_CHECK(stream->priority_it != stream_priorities_.end()); in Push() 388 RTC_CHECK(!stream_priorities_.empty()); in GetHighestPriorityStream() 392 RTC_CHECK(stream_info_it != streams_.end()); in GetHighestPriorityStream() 393 RTC_CHECK(stream_info_it->second.priority_it == stream_priorities_.begin()); in GetHighestPriorityStream() 394 RTC_CHECK(!stream_info_it->second.packet_queue.empty()); in GetHighestPriorityStream()
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