/external/clang/test/CodeGen/ |
H A D | 2003-11-19-BitFieldArray.c | 9 void g_io_channel_init (struct _GIOChannel *channel) { argument 10 channel->partial_write_buf[0];
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/external/chromium_org/remoting/tools/mac/ |
H A D | chromoting-set-channel.sh | 16 echo "Usage: ${ME} <channel>" >&2 17 echo "where <channel> is 'beta' or 'stable'" >&2 34 local channel="$1" 36 if [[ "${channel}" != "beta" && "${channel}" != "stable" ]]; then 41 local channeltag="${channel}" 42 if [[ "${channel}" == "stable" ]]; then 46 log "Switching Chrome Remote Desktop channel to ${channel}" 50 if [[ "${channel}" [all...] |
/external/chromium_org/chrome/browser/resources/help/ |
H A D | channel_change_page.css | 6 #channel-change-page { 11 .channel-change-page-channel label { 15 .channel-change-page-channel { 20 .show-when-selected-channel-requires-powerwash, 21 .show-when-selected-channel-requires-delayed-update, 22 .show-when-selected-channel-good, 23 .show-when-selected-channel-unstable { 27 .selected-channel [all...] |
/external/chromium_org/ppapi/native_client/src/trusted/plugin/ |
H A D | sel_ldr_launcher_chrome.cc | 16 void SelLdrLauncherChrome::set_channel(NaClHandle channel) { argument 18 channel_ = channel;
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/external/chromium_org/third_party/WebKit/Source/modules/mediastream/ |
H A D | RTCDataChannelEvent.idl | 28 readonly attribute RTCDataChannel channel;
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/external/chromium_org/third_party/webrtc/examples/android/media_demo/src/org/webrtc/webrtcdemo/ |
H A D | VideoEngine.java | 61 public native int startSend(int channel); argument 62 public native int stopRender(int channel); argument 63 public native int stopSend(int channel); argument 64 public native int startReceive(int channel); argument 65 public native int stopReceive(int channel); argument 67 public native int deleteChannel(int channel); argument 69 public native int setLocalReceiver(int channel, int port); argument 70 public native int setSendDestination(int channel, int port, String ipAddr); argument 73 public native int setReceiveCodec(int channel, VideoCodecInst codec); argument 74 public native int setSendCodec(int channel, VideoCodecIns argument 75 addRenderer(int channel, Object glSurface, int zOrder, float left, float top, float right, float bottom) argument 78 removeRenderer(int channel) argument 79 registerExternalReceiveCodec(int channel, int plType, MediaCodecVideoDecoder decoder, boolean internal_source) argument 81 deRegisterExternalReceiveCodec(int channel, int plType) argument 82 startRender(int channel) argument 86 connectCaptureDevice(int cameraId, int channel) argument 92 setNackStatus(int channel, boolean enable) argument 93 setKeyFrameRequestMethod(int channel, VieKeyFrameRequestMethod requestMethod) argument 97 setKeyFrameRequestMethod(int channel, int requestMethod) argument 99 getReceivedRtcpStatistics(int channel) argument 100 registerObserver(int channel, VideoDecodeEncodeObserver callback) argument 102 deregisterObserver(int channel) argument 109 startRtpDump(int channel, String file, RtpDirections direction) argument 113 startRtpDump(int channel, String file, int direction) argument 115 stopRtpDump(int channel, RtpDirections direction) argument 118 stopRtpDump(int channel, int direction) argument 119 setLocalSSRC(int channel, int ssrc) argument [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | voe_rtp_rtcp_impl.h | 24 virtual int SetRTCPStatus(int channel, bool enable); 26 virtual int GetRTCPStatus(int channel, bool& enabled); 28 virtual int SetRTCP_CNAME(int channel, const char cName[256]); 30 virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]); 32 virtual int GetRemoteRTCPData(int channel, 41 virtual int SetLocalSSRC(int channel, unsigned int ssrc); 43 virtual int GetLocalSSRC(int channel, unsigned int& ssrc); 45 virtual int GetRemoteSSRC(int channel, unsigned int& ssrc); 48 virtual int SetSendAudioLevelIndicationStatus(int channel, 51 virtual int SetReceiveAudioLevelIndicationStatus(int channel, [all...] |
H A D | voe_codec_impl.h | 28 virtual int SetSendCodec(int channel, const CodecInst& codec); 30 virtual int GetSendCodec(int channel, CodecInst& codec); 32 virtual int GetRecCodec(int channel, CodecInst& codec); 35 int channel, int type, 38 virtual int SetRecPayloadType(int channel, 41 virtual int GetRecPayloadType(int channel, CodecInst& codec); 43 virtual int SetFECStatus(int channel, bool enable); 45 virtual int GetFECStatus(int channel, bool& enabled); 47 virtual int SetVADStatus(int channel, 52 virtual int GetVADStatus(int channel, [all...] |
H A D | voe_video_sync_impl.h | 25 virtual int SetMinimumPlayoutDelay(int channel, int delayMs); 27 virtual int SetInitialPlayoutDelay(int channel, int delay_ms); 29 virtual int GetDelayEstimate(int channel, 33 virtual int GetLeastRequiredDelayMs(int channel) const; 35 virtual int SetInitTimestamp(int channel, unsigned int timestamp); 37 virtual int SetInitSequenceNumber(int channel, short sequenceNumber); 39 virtual int GetPlayoutTimestamp(int channel, unsigned int& timestamp); 41 virtual int GetRtpRtcp(int channel, RtpRtcp** rtpRtcpModule,
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/external/e2fsprogs/lib/ext2fs/ |
H A D | io_manager.c | 22 errcode_t io_channel_set_options(io_channel channel, const char *opts) argument 27 EXT2_CHECK_MAGIC(channel, EXT2_ET_MAGIC_IO_CHANNEL); 32 if (!channel->manager->set_option) 50 retval = (channel->manager->set_option)(channel, ptr, arg); 59 errcode_t io_channel_write_byte(io_channel channel, unsigned long offset, argument 62 EXT2_CHECK_MAGIC(channel, EXT2_ET_MAGIC_IO_CHANNEL); 64 if (channel->manager->write_byte) 65 return channel->manager->write_byte(channel, offse 71 io_channel_read_blk64(io_channel channel, unsigned long long block, int count, void *data) argument 87 io_channel_write_blk64(io_channel channel, unsigned long long block, int count, const void *data) argument 103 io_channel_discard(io_channel channel, unsigned long long block, unsigned long long count) argument [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/include/ |
H A D | voe_codec.h | 14 // - Voice Activity Detection (VAD) on a per channel basis. 60 // Sets the |codec| for the |channel| to be used for sending. 61 virtual int SetSendCodec(int channel, const CodecInst& codec) = 0; 64 // |channel|. 65 virtual int GetSendCodec(int channel, CodecInst& codec) = 0; 67 // Sets the |codec| as secondary codec for |channel|. Registering a 74 virtual int SetSecondarySendCodec(int channel, const CodecInst& codec, 77 // Removes the secondary codec from |channel|. This will terminate 79 virtual int RemoveSecondarySendCodec(int channel) = 0; 81 // Gets |codec| which is used as secondary codec in |channel| 109 SetFECStatus(int channel, bool enable) argument 116 GetFECStatus(int channel, bool& enabled) argument 133 SetOpusMaxPlaybackRate(int channel, int frequency_hz) argument 138 SetAMREncFormat(int channel, AmrMode mode) argument 139 SetAMRDecFormat(int channel, AmrMode mode) argument 140 SetAMRWbEncFormat(int channel, AmrMode mode) argument 141 SetAMRWbDecFormat(int channel, AmrMode mode) argument 142 SetISACInitTargetRate(int channel, int rateBps, bool useFixedFrameSize = false) argument 144 SetISACMaxRate(int channel, int rateBps) argument 145 SetISACMaxPayloadSize(int channel, int sizeBytes) argument [all...] |
H A D | voe_rtp_rtcp.h | 55 int channel, unsigned int CSRC, bool added) = 0; 58 int channel, unsigned int SSRC) = 0; 69 int channel, unsigned char subType, 132 virtual int SetLocalSSRC(int channel, unsigned int ssrc) = 0; 134 // Gets the local RTP SSRC of a specified |channel|. 135 virtual int GetLocalSSRC(int channel, unsigned int& ssrc) = 0; 138 virtual int GetRemoteSSRC(int channel, unsigned int& ssrc) = 0; 140 // Sets the status of rtp-audio-level-indication on a specific |channel|. 141 virtual int SetSendAudioLevelIndicationStatus(int channel, 146 // |channel| 147 SetReceiveAudioLevelIndicationStatus(int channel, bool enable, unsigned char id = 1) argument 176 GetRTCP_CNAME(int channel, char cName[256]) argument 208 SetREDStatus( int channel, bool enable, int redPayloadtype = -1) argument 214 GetREDStatus( int channel, bool& enabled, int& redPayloadtype) argument 220 SetFECStatus( int channel, bool enable, int redPayloadtype = -1) argument 228 GetFECStatus( int channel, bool& enabled, int& redPayloadtype) argument 262 SetVideoEngineBWETarget(int channel, ViENetwork* vie_network, int video_channel) argument 268 RegisterRTPObserver(int channel, VoERTPObserver& observer) argument 270 DeRegisterRTPObserver(int channel) argument 271 RegisterRTCPObserver( int channel, VoERTCPObserver& observer) argument 273 DeRegisterRTCPObserver(int channel) argument 274 GetRemoteCSRCs(int channel, unsigned int arrCSRC[15]) argument 276 InsertExtraRTPPacket( int channel, unsigned char payloadType, bool markerBit, const char* payloadData, unsigned short payloadSize) argument 279 GetRemoteRTCPSenderInfo( int channel, SenderInfo* sender_info) argument 281 SendApplicationDefinedRTCPPacket( int channel, unsigned char subType, unsigned int name, const char* data, unsigned short dataLengthInBytes) argument 284 GetLastRemoteTimeStamp(int channel, uint32_t* lastRemoteTimeStamp) argument [all...] |
H A D | voe_network.h | 48 // notifications for a specified |channel| when the observer interface 50 virtual void OnPeriodicDeadOrAlive(int channel, bool alive) = 0; 72 // specified |channel|. 74 int channel, Transport& transport) = 0; 77 // specified |channel|. 78 virtual int DeRegisterExternalTransport(int channel) = 0; 83 virtual int ReceivedRTPPacket(int channel, 86 virtual int ReceivedRTPPacket(int channel, argument 97 int channel, const void* data, unsigned int length) = 0;
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H A D | voe_video_sync.h | 62 // maintained by the jitter buffer, unless channel condition (jitter in 66 virtual int SetMinimumPlayoutDelay(int channel, int delay_ms) = 0; 73 virtual int SetInitialPlayoutDelay(int channel, int delay_ms) = 0; 76 // the |playout_buffer_delay_ms| for a specified |channel|. 77 virtual int GetDelayEstimate(int channel, 85 virtual int GetLeastRequiredDelayMs(int channel) const = 0; 88 virtual int SetInitTimestamp(int channel, unsigned int timestamp) = 0; 91 virtual int SetInitSequenceNumber(int channel, short sequenceNumber) = 0; 94 virtual int GetPlayoutTimestamp(int channel, unsigned int& timestamp) = 0; 96 virtual int GetRtpRtcp (int channel, RtpRtc [all...] |
/external/chromium_org/third_party/WebKit/Tools/Scripts/webkitpy/common/config/ |
H A D | irc.py | 28 channel = "#blink" variable
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/external/chromium_org/chrome/common/ |
H A D | chrome_version_info_win.cc | 22 base::string16 channel; local 28 &channel); 32 channel += L" SyzyASan"; 34 return base::UTF16ToASCII(channel); 43 std::wstring channel(L"unknown"); 49 channel = GoogleUpdateSettings::GetChromeChannel(is_system_install); 52 if (channel.empty()) { 54 } else if (channel == L"beta") { 56 } else if (channel == L"dev") { 58 } else if (channel [all...] |
H A D | chrome_version_info_chromeos.cc | 36 void VersionInfo::SetChannel(const std::string& channel) { argument 38 if (channel == "stable-channel") { 40 } else if (channel == "beta-channel") { 42 } else if (channel == "dev-channel") { 44 } else if (channel == "canary-channel") {
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/external/qemu/telephony/ |
H A D | sysdeps_posix.c | 98 /*** channel allocation ***/ 108 SysChannel channel = _s_free_channels; local 109 assert( channel != NULL && "out of free channels" ); 110 _s_free_channels = channel->next; 111 channel->next = NULL; 112 channel->active = 0; 113 channel->closed = 0; 114 channel->pending = 0; 115 channel->wanted = 0; 116 return channel; 199 sys_channel_on( SysChannel channel, int events, SysChannelCallback callback, void* opaque ) argument 233 sys_channel_read( SysChannel channel, void* buffer, int size ) argument 258 sys_channel_write( SysChannel channel, const void* buffer, int size ) argument 467 SysChannel channel; local 512 SysChannel channel; local 546 SysChannel channel; local 592 SysChannel channel = sys_channel_alloc(); local 613 SysChannel channel = sys_channel_alloc(); local [all...] |
H A D | sysdeps_qemu.c | 181 SysChannel channel = _s_free_channels; local 182 if (channel != NULL) { 183 _s_free_channels = channel->next; 184 channel->next = NULL; 185 channel->fd = -1; 186 channel->callback = NULL; 187 channel->opaque = NULL; 189 return channel; 193 sys_channel_free( SysChannel channel ) 195 if (channel 207 SysChannel channel = _channel; local 216 SysChannel channel = _channel; local 222 sys_channel_on( SysChannel channel, int events, SysChannelCallback event_callback, void* event_opaque ) argument 242 sys_channel_read( SysChannel channel, void* buffer, int size ) argument 269 sys_channel_write( SysChannel channel, const void* buffer, int size ) argument 319 SysChannel channel = sys_channel_alloc(); local 339 SysChannel channel = sys_channel_alloc(); local 364 SysChannel channel = sys_channel_alloc(); local [all...] |
/external/srec/srec/cfront/ |
H A D | chelfep.c | 47 static featdata smoothed_c0(front_cep *cepobj, front_channel *channel); 50 int make_frame(front_channel *channel, front_wave *waveobj, argument 62 MEMMOVE(channel->cep + (channel->mel_dim + 1), channel->cep, 63 (Q2 - 1) *(channel->mel_dim + 1), sizeof(float)); 71 filterbank_emulation(channel, waveobj, freqobj, cepobj, 74 read channel->fbo and dump it. */ 78 cepstrum_params(channel, waveobj, freqobj, cepobj); 81 (void) make_std_frame(channel, cepob 110 smoothed_c0(front_cep *cepobj, front_channel *channel) argument [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | sync_buffer_unittest.cc | 27 for (size_t channel = 0; channel < kChannels; ++channel) { 29 EXPECT_EQ(0, sync_buffer[channel][i]); 58 for (size_t channel = 0; channel < kChannels; ++channel) { 60 new_data[channel][i] = i; 71 for (size_t channel = 0; channel < kChannel [all...] |
/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
H A D | fakewebrtcvideoengine.h | 56 // WebRtc channel id and capture id share the same number space. 58 // renderer for a channel or it is adding a renderer for a capturer. 318 bool remb_contribute_; // This channel contributes to the remb report. 319 bool remb_bw_partition_; // This channel is allocated part of total bw. 411 bool IsChannel(int channel) const { 412 return (channels_.find(channel) != channels_.end()); 435 int GetCaptureId(int channel) const { 436 WEBRTC_ASSERT_CHANNEL(channel); 437 return channels_.find(channel)->second->capture_id_; 439 int GetOriginalChannelId(int channel) cons 491 GetSendRtpExtensionId(int channel, const std::string& extension) argument 500 GetReceiveRtpExtensionId(int channel, const std::string& extension) argument 510 GetTransmissionSmoothingStatus(int channel) argument 514 GetSenderTargetDelay(int channel) argument 518 GetReceiverTargetDelay(int channel) argument 552 GetSsrc(int channel, int idx) const argument 560 GetRtxSsrc(int channel, int idx) const argument 568 ReceiveCodecRegistered(int channel, const webrtc::VideoCodec& codec) const argument 575 ExternalDecoderRegistered(int channel, unsigned int pl_type) const argument 586 ExternalEncoderRegistered(int channel, unsigned int pl_type) const argument 605 SetSendBitrates(int channel, unsigned int video_bitrate, unsigned int fec_bitrate, unsigned int nack_bitrate) argument 612 SetSendBandwidthEstimate(int channel, unsigned int send_bandwidth) argument 616 SetReceiveBandwidthEstimate(int channel, unsigned int receive_bandwidth) argument 622 GetRtxSendPayloadType(int channel) argument 626 GetRtxRecvPayloadType(int channel) argument 630 GetRemoteRtxSsrc(int channel) argument 634 GetSuspendBelowMinBitrateStatus(int channel) argument 642 GetReservedTransmitBitrate(int channel) argument 655 WEBRTC_FUNC(CreateChannel, (int& channel)) argument 671 WEBRTC_FUNC(CreateChannel, (int& channel, int original_channel)) argument 679 WEBRTC_FUNC(CreateReceiveChannel, (int& channel, int original_channel)) argument 682 WEBRTC_FUNC(DeleteChannel, (const int channel)) argument 690 WEBRTC_FUNC(RegisterCpuOveruseObserver, (int channel, webrtc::CpuOveruseObserver* observer)) argument 701 WEBRTC_FUNC(SetCpuOveruseOptions, (int channel, const webrtc::CpuOveruseOptions& options)) argument 709 WEBRTC_FUNC(StartSend, (const int channel)) argument 714 WEBRTC_FUNC(StopSend, (const int channel)) argument 719 WEBRTC_FUNC(StartReceive, (const int channel)) argument 724 WEBRTC_FUNC(StopReceive, (const int channel)) argument 765 WEBRTC_FUNC(SetSendCodec, (const int channel, const webrtc::VideoCodec& codec)) argument 772 WEBRTC_FUNC_CONST(GetSendCodec, (const int channel, webrtc::VideoCodec& codec)) argument 778 WEBRTC_FUNC(SetReceiveCodec, (const int channel, const webrtc::VideoCodec& codec)) argument 794 WEBRTC_FUNC_CONST(GetCodecTargetBitrate, (const int channel, unsigned int* codec_target_bitrate)) argument 824 WEBRTC_VOID_FUNC(SuspendBelowMinBitrate, (int channel)) argument 855 WEBRTC_FUNC(ConnectCaptureDevice, (const int capture_id, const int channel)) argument 863 WEBRTC_FUNC(DisconnectCaptureDevice, (const int channel)) argument 887 WEBRTC_VOID_FUNC(SetNetworkTransmissionState, (const int channel, const bool is_transmitting)) argument 895 WEBRTC_FUNC(ReceivedRTPPacket, (const int channel, const void* packet, const int length, const webrtc::PacketTime& packet_time)) argument 908 IsIPv6Enabled(int channel) argument 974 WEBRTC_FUNC(SetLocalSSRC, (const int channel, const unsigned int ssrc, const webrtc::StreamType usage, const unsigned char idx)) argument 992 WEBRTC_FUNC_CONST(SetRemoteSSRCType, (const int channel, const webrtc::StreamType usage, const unsigned int ssrc)) argument 1002 WEBRTC_FUNC_CONST(GetLocalSSRC, (const int channel, unsigned int& ssrc)) argument 1012 WEBRTC_FUNC(SetRtxSendPayloadType, (const int channel, const uint8 payload_type)) argument 1023 WEBRTC_FUNC(SetRtxReceivePayloadType, (const int channel, const uint8 payload_type)) argument 1031 WEBRTC_FUNC(SetRTCPStatus, (const int channel, const webrtc::ViERTCPMode mode)) argument 1038 WEBRTC_FUNC(SetRTCPCName, (const int channel, const char rtcp_cname[KMaxRTCPCNameLength])) argument 1044 WEBRTC_FUNC_CONST(GetRTCPCName, (const int channel, char rtcp_cname[KMaxRTCPCNameLength])) argument 1054 WEBRTC_FUNC(SetNACKStatus, (const int channel, const bool enable)) argument 1062 WEBRTC_FUNC(SetHybridNACKFECStatus, (const int channel, const bool enable, const unsigned char red_type, const unsigned char fec_type)) argument 1074 WEBRTC_FUNC(SetKeyFrameRequestMethod, (const int channel, const webrtc::ViEKeyFrameRequestMethod method)) argument 1081 WEBRTC_FUNC(SetSenderBufferingMode, (int channel, int target_delay)) argument 1086 WEBRTC_FUNC(SetReceiverBufferingMode, (int channel, int target_delay)) argument 1093 WEBRTC_FUNC(SetRembStatus, (int channel, bool send, bool receive)) argument 1099 WEBRTC_FUNC(SetTMMBRStatus, (const int channel, const bool enable)) argument 1104 WEBRTC_FUNC(SetSendTimestampOffsetStatus, (int channel, bool enable, int id)) argument 1110 WEBRTC_FUNC(SetReceiveTimestampOffsetStatus, (int channel, bool enable, int id)) argument 1116 WEBRTC_FUNC(SetSendAbsoluteSendTimeStatus, (int channel, bool enable, int id)) argument 1122 WEBRTC_FUNC(SetReceiveAbsoluteSendTimeStatus, (int channel, bool enable, int id)) argument 1129 WEBRTC_FUNC(SetTransmissionSmoothingStatus, (int channel, bool enable)) argument 1134 WEBRTC_FUNC(SetReservedTransmitBitrate, (int channel, unsigned int reserved_transmit_bitrate_bps)) argument 1155 WEBRTC_FUNC_CONST(GetBandwidthUsage, (const int channel, unsigned int& total_bitrate, unsigned int& video_bitrate, unsigned int& fec_bitrate, unsigned int& nack_bitrate)) argument 1173 WEBRTC_FUNC_CONST(GetEstimatedSendBandwidth, (const int channel, unsigned int* send_bandwidth_estimate)) argument 1186 WEBRTC_FUNC_CONST(GetEstimatedReceiveBandwidth, (const int channel, unsigned int* receive_bandwidth_estimate)) argument 1254 WEBRTC_FUNC(RegisterExternalSendCodec, (const int channel, const unsigned char pl_type, webrtc::VideoEncoder*, bool)) argument 1261 WEBRTC_FUNC(DeRegisterExternalSendCodec, (const int channel, const unsigned char pl_type)) argument 1267 WEBRTC_FUNC(RegisterExternalReceiveCodec, (const int channel, const unsigned int pl_type, webrtc::VideoDecoder*, bool, int)) argument 1274 WEBRTC_FUNC(DeRegisterExternalReceiveCodec, (const int channel, const unsigned char pl_type)) argument [all...] |
H A D | fakewebrtcvoiceengine.h | 298 bool GetPlayout(int channel) { argument 299 return channels_[channel]->playout; 301 bool GetSend(int channel) { argument 302 return channels_[channel]->send; 307 bool GetVAD(int channel) { argument 308 return channels_[channel]->vad; 310 bool GetRED(int channel) { argument 311 return channels_[channel]->red; 313 bool GetCodecFEC(int channel) { argument 314 return channels_[channel] 316 GetMaxEncodingBandwidth(int channel) argument 319 GetNACK(int channel) argument 322 GetNACKMaxPackets(int channel) argument 325 GetViENetwork(int channel) argument 331 GetVideoChannel(int channel) argument 335 GetLastRtpPacketTime(int channel) argument 339 GetSendCNPayloadType(int channel, bool wideband) argument 344 GetSendTelephoneEventPayloadType(int channel) argument 347 GetSendREDPayloadType(int channel) argument 350 CheckPacket(int channel, const void* data, size_t len) argument 359 CheckNoPacket(int channel) argument 366 set_playout_fail_channel(int channel) argument 369 set_send_fail_channel(int channel) argument 405 GetSendRtpExtensionId(int channel, const std::string& extension) argument 414 GetReceiveRtpExtensionId(int channel, const std::string& extension) argument 457 WEBRTC_FUNC(DeleteChannel, (int channel)) argument 464 WEBRTC_FUNC(StartPlayout, (int channel)) argument 475 WEBRTC_FUNC(StartSend, (int channel)) argument 487 WEBRTC_FUNC(StopPlayout, (int channel)) argument 492 WEBRTC_FUNC(StopSend, (int channel)) argument 519 WEBRTC_FUNC(SetSendCodec, (int channel, const webrtc::CodecInst& codec)) argument 532 WEBRTC_FUNC(GetSendCodec, (int channel, webrtc::CodecInst& codec)) argument 543 WEBRTC_FUNC(GetRecCodec, (int channel, webrtc::CodecInst& codec)) argument 571 WEBRTC_FUNC(SetRecPayloadType, (int channel, const webrtc::CodecInst& codec)) argument 599 WEBRTC_FUNC(SetSendCNPayloadType, (int channel, int type, webrtc::PayloadFrequencies frequency)) argument 609 WEBRTC_FUNC(GetRecPayloadType, (int channel, webrtc::CodecInst& codec)) argument 624 WEBRTC_FUNC(SetVADStatus, (int channel, bool enable, webrtc::VadModes mode, bool disableDTX)) argument 638 WEBRTC_FUNC(SetFECStatus, (int channel, bool enable)) argument 648 WEBRTC_FUNC(GetFECStatus, (int channel, bool& enable)) argument 654 WEBRTC_FUNC(SetOpusMaxPlaybackRate, (int channel, int frequency_hz)) argument 675 WEBRTC_FUNC(SendTelephoneEvent, (int channel, int event_code, bool out_of_band = true, int length_ms = 160, int attenuation_db = 10)) argument 683 WEBRTC_FUNC(SetSendTelephoneEventPayloadType, (int channel, unsigned char type)) argument 707 WEBRTC_FUNC(StartPlayingFileLocally, (int channel, const char* fileNameUTF8, bool loop, webrtc::FileFormats format, float volumeScaling, int startPointMs, int stopPointMs)) argument 715 WEBRTC_FUNC(StartPlayingFileLocally, (int channel, webrtc::InStream* stream, webrtc::FileFormats format, float volumeScaling, int startPointMs, int stopPointMs)) argument 723 WEBRTC_FUNC(StopPlayingFileLocally, (int channel)) argument 728 WEBRTC_FUNC(IsPlayingFileLocally, (int channel)) argument 844 WEBRTC_FUNC_CONST(GetDecodingCallStatistics, (int channel, webrtc::AudioDecodingCallStats*)) argument 851 WEBRTC_FUNC(RegisterExternalTransport, (int channel, webrtc::Transport& transport)) argument 857 WEBRTC_FUNC(DeRegisterExternalTransport, (int channel)) argument 862 WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data, unsigned int length)) argument 870 WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data, unsigned int length, const webrtc::PacketTime& packet_time)) argument 891 WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) argument 896 WEBRTC_FUNC(GetLocalSSRC, (int channel, unsigned int& ssrc)) argument 902 WEBRTC_FUNC(SetSendAudioLevelIndicationStatus, (int channel, bool enable, unsigned char id)) argument 909 WEBRTC_FUNC(SetReceiveAudioLevelIndicationStatus, (int channel, bool enable, unsigned char id)) argument 916 WEBRTC_FUNC(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable, unsigned char id)) argument 923 WEBRTC_FUNC(SetReceiveAbsoluteSenderTimeStatus, (int channel, bool enable, unsigned char id)) argument 945 WEBRTC_FUNC(GetRemoteRTCPReportBlocks, (int channel, std::vector<webrtc::ReportBlock>* receive_blocks)) argument 970 WEBRTC_FUNC(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats)) argument 984 WEBRTC_FUNC(SetREDStatus, (int channel, bool enable, int redPayloadtype)) argument 990 WEBRTC_FUNC(SetFECStatus, (int channel, bool enable, int redPayloadtype)) argument 997 WEBRTC_FUNC(GetREDStatus, (int channel, bool& enable, int& redPayloadtype)) argument 1003 WEBRTC_FUNC(GetFECStatus, (int channel, bool& enable, int& redPayloadtype)) argument 1009 WEBRTC_FUNC(SetNACKStatus, (int channel, bool enable, int maxNoPackets)) argument 1024 WEBRTC_FUNC(SetVideoEngineBWETarget, (int channel, webrtc::ViENetwork* vie_network, int video_channel)) argument 1065 WEBRTC_FUNC(SetChannelOutputVolumeScaling, (int channel, float scale)) argument 1070 WEBRTC_FUNC(GetChannelOutputVolumeScaling, (int channel, float& scale)) argument 1075 WEBRTC_FUNC(SetOutputVolumePan, (int channel, float left, float right)) argument 1081 WEBRTC_FUNC(GetOutputVolumePan, (int channel, float& left, float& right)) argument 1146 WEBRTC_FUNC(SetRxAgcStatus, (int channel, bool enable, webrtc::AgcModes mode)) argument 1152 WEBRTC_FUNC(GetRxAgcStatus, (int channel, bool& enabled, webrtc::AgcModes& mode)) argument 1159 WEBRTC_FUNC(SetRxAgcConfig, (int channel, webrtc::AgcConfig config)) argument 1163 WEBRTC_FUNC(GetRxAgcConfig, (int channel, webrtc::AgcConfig& config)) argument 1214 WasSendTelephoneEventCalled(int channel, int event_code, int length_ms) argument 1224 WEBRTC_FUNC(RegisterExternalMediaProcessing, (int channel, webrtc::ProcessingTypes type, webrtc::VoEMediaProcess& processObject)) argument 1235 WEBRTC_FUNC(DeRegisterExternalMediaProcessing, (int channel, webrtc::ProcessingTypes type)) argument [all...] |
/external/chromium_org/third_party/webrtc/test/ |
H A D | mock_transport.h | 22 int(int channel, const void* data, int len)); 24 int(int channel, const void* data, int len));
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/external/chromium_org/third_party/webrtc/voice_engine/include/mock/ |
H A D | mock_voe_volume_control.h | 30 MOCK_METHOD2(SetInputMute, int(int channel, bool enable)); 31 MOCK_METHOD2(GetInputMute, int(int channel, bool& enabled)); 35 MOCK_METHOD2(GetSpeechOutputLevel, int(int channel, unsigned int& level)); 38 int(int channel, unsigned int& level)); 39 MOCK_METHOD2(SetChannelOutputVolumeScaling, int(int channel, float scaling)); 40 MOCK_METHOD2(GetChannelOutputVolumeScaling, int(int channel, float& scaling)); 41 MOCK_METHOD3(SetOutputVolumePan, int(int channel, float left, float right)); 42 MOCK_METHOD3(GetOutputVolumePan, int(int channel, float& left, float& right));
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