/libcore/luni/src/main/java/java/nio/channels/ |
Channels.java | 42 * Returns an input stream on the given channel. The resulting stream has 45 * <li>If the stream is closed, then the underlying channel is closed as 48 * <li>It throws an {@link IllegalBlockingModeException} if the channel is 54 * @param channel 55 * the channel to be wrapped by an InputStream. 56 * @return an InputStream that takes bytes from the given byte channel. 58 public static InputStream newInputStream(ReadableByteChannel channel) { 59 return new ChannelInputStream(channel); 63 * Returns an output stream on the given channel. The resulting stream has 66 * <li>If the stream is closed, then the underlying channel is closed a 206 private final ReadableByteChannel channel; field in class:Channels.ChannelInputStream 246 private final WritableByteChannel channel; field in class:Channels.ChannelOutputStream [all...] |
/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
fakewebrtcvoiceengine.h | 90 struct Channel { 91 explicit Channel() 188 for (std::map<int, Channel*>::const_iterator i = channels_.begin(); 200 for (std::map<int, Channel*>::const_iterator iter = channels_.begin(); 208 bool GetPlayout(int channel) { 209 return channels_[channel]->playout; 211 bool GetSend(int channel) { 212 return channels_[channel]->send; 217 bool GetVAD(int channel) { 218 return channels_[channel]->vad [all...] |
/external/chromium_org/third_party/webrtc/test/ |
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/ |
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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mock_voe_connection_observer.h | 22 MOCK_METHOD2(OnPeriodicDeadOrAlive, void(int channel,
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/external/chromium_org/third_party/webrtc/voice_engine/include/ |
voe_volume_control.h | 73 virtual int SetInputMute(int channel, bool enable) = 0; 76 virtual int GetInputMute(int channel, bool& enabled) = 0; 84 virtual int GetSpeechOutputLevel(int channel, unsigned int& level) = 0; 92 int channel, unsigned int& level) = 0; 95 // channel. Valid scale range is [0.0, 10.0]. 96 virtual int SetChannelOutputVolumeScaling(int channel, float scaling) = 0; 98 // Gets the current volume scaling for a specified |channel|. 99 virtual int GetChannelOutputVolumeScaling(int channel, float& scaling) = 0; 103 virtual int SetOutputVolumePan(int channel, float left, float right) = 0; 106 virtual int GetOutputVolumePan(int channel, float& left, float& right) = 0 [all...] |
voe_base.h | 16 // - Multi-channel support (mixing, sending to multiple destinations etc.). 55 virtual void CallbackOnError(int channel, int errCode) = 0; 138 // Creates a new channel and allocates the required resources for it. 139 // One can use |config| to configure the channel. Currently that is used for 144 // Deletes an existing channel and releases the utilized resources. 145 virtual int DeleteChannel(int channel) = 0; 148 // incoming RTP/RTCP packets on the specified |channel|. 149 virtual int StartReceive(int channel) = 0; 151 // Stops receiving incoming RTP/RTCP packets on the specified |channel|. 152 virtual int StopReceive(int channel) = 0 [all...] |
voe_dtmf.h | 59 virtual int SendTelephoneEvent(int channel, int eventCode, 66 virtual int SetSendTelephoneEventPayloadType(int channel, 71 virtual int GetSendTelephoneEventPayloadType(int channel, 76 virtual int SetDtmfPlayoutStatus(int channel, bool enable) = 0; 79 virtual int GetDtmfPlayoutStatus(int channel, bool& enabled) = 0;
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voe_file.h | 64 int channel, 75 int channel, 81 // Stops playback of a file on a specific |channel|. 82 virtual int StopPlayingFileLocally(int channel) = 0; 84 // Returns the current file playing state for a specific |channel|. 85 virtual int IsPlayingFileLocally(int channel) = 0; 90 int channel, 100 int channel, 106 // Stops playing of a file as microphone signal for a specific |channel|. 107 virtual int StopPlayingFileAsMicrophone(int channel) = 0 [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
dsp_helper_unittest.cc | 54 for (int channel = 0; channel < kChannels; ++channel) { 55 input[channel][i] = 1000; 72 for (int channel = 0; channel < kChannels; ++channel) { 73 EXPECT_EQ(1000, input[channel][i]); 78 for (int channel = 0; channel < kChannels; ++channel) [all...] |
audio_multi_vector_unittest.cc | 49 // Write 100, 101, 102, ... for first channel. 50 // Write 200, 201, 202, ... for second channel. 88 for (size_t channel = 0; channel < num_channels_; ++channel) { 90 vec[channel][i] = static_cast<int16_t>(i); 92 const AudioVector& audio_vec = vec[channel]; 109 for (size_t channel = 0; channel < vec.Channels(); ++channel) { [all...] |
/external/chromium_org/chrome/browser/sync/glue/ |
chrome_report_unrecoverable_error.cc | 16 chrome::VersionInfo::Channel channel = chrome::VersionInfo::GetChannel(); local 17 if (channel != chrome::VersionInfo::CHANNEL_CANARY && 18 channel != chrome::VersionInfo::CHANNEL_DEV) {
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/external/srec/srec/cfront/ |
chelmel4.c | 60 front_channel *channel); 67 void cepstrum_params(front_channel *channel, front_wave *waveobj, 73 mel_spectrum_correction(freqobj->filterbank, cepobj->mel_loop, channel->num_freq); 76 mel_loglookup_with_offset(cepobj, channel); 79 write_scaled_frames(freqobj->nf, 1, channel->filterbank, D_FIXED, 1 / (float)LOG_SCALE); 83 icostrans(cepobj->cs, channel->filterbank, channel->cep, 84 channel->num_freq, cepobj->mel_dim); 88 write_scaled_frames((cepobj->mel_dim + 1), 1, channel->cep, D_FIXED, (float)1 / (0x01 << (LOG_SCALE_SHIFT + COSINE_TABLE_SHIFT))); 126 front_channel *channel) [all...] |
/hardware/broadcom/wlan/bcmdhd/dhdutil/include/ |
bcmwifi_channels.h | 27 /* A chanspec holds the channel number, band, bandwidth and control sideband */ 30 /* channel defines */ 39 #define CH_MAX_2G_CHANNEL 14 /* Max channel in 2G band */ 40 #define MAXCHANNEL 224 /* max # supported channels. The max channel no is 216, 80 /* channel defines */ 81 #define LOWER_20_SB(channel) (((channel) > CH_10MHZ_APART) ? ((channel) - CH_10MHZ_APART) : 0) 82 #define UPPER_20_SB(channel) (((channel) < (MAXCHANNEL - CH_10MHZ_APART)) ? [all...] |
/external/chromium_org/third_party/webrtc/examples/android/media_demo/src/org/webrtc/webrtcdemo/ |
VoiceEngine.java | 64 public native int deleteChannel(int channel); 65 public native int setLocalReceiver(int channel, int port); 66 public native int setSendDestination(int channel, int port, String ipaddr); 67 public native int startListen(int channel); 68 public native int startPlayout(int channel); 69 public native int startSend(int channel); 70 public native int stopListen(int channel); 71 public native int stopPlayout(int channel); 72 public native int stopSend(int channel); 76 int channel, [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/ |
voe_dtmf_impl.h | 24 int channel, 30 virtual int SetSendTelephoneEventPayloadType(int channel, 33 virtual int GetSendTelephoneEventPayloadType(int channel, 45 virtual int SetDtmfPlayoutStatus(int channel, bool enable); 47 virtual int GetDtmfPlayoutStatus(int channel, bool& enabled);
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voe_file_impl.h | 25 int channel, 34 int channel, 40 virtual int StopPlayingFileLocally(int channel); 42 virtual int IsPlayingFileLocally(int channel); 47 int channel, 55 int channel, 61 virtual int StopPlayingFileAsMicrophone(int channel); 63 virtual int IsPlayingFileAsMicrophone(int channel); 67 virtual int StartRecordingPlayout(int channel, 72 virtual int StartRecordingPlayout(int channel, [all...] |
/external/antlr/antlr-3.4/runtime/Java/src/main/java/org/antlr/runtime/ |
ClassicToken.java | 42 protected int channel=DEFAULT_CHANNEL; field in class:ClassicToken 56 channel = oldToken.getChannel(); 64 public ClassicToken(int type, String text, int channel) { 67 this.channel = channel; 99 return channel; 102 public void setChannel(int channel) { 103 this.channel = channel; 127 if ( channel>0 ) [all...] |
/external/e2fsprogs/lib/ext2fs/ |
inode_io.c | 49 static errcode_t inode_open(const char *name, int flags, io_channel *channel); 50 static errcode_t inode_close(io_channel channel); 51 static errcode_t inode_set_blksize(io_channel channel, int blksize); 52 static errcode_t inode_read_blk(io_channel channel, unsigned long block, 54 static errcode_t inode_write_blk(io_channel channel, unsigned long block, 56 static errcode_t inode_flush(io_channel channel); 57 static errcode_t inode_write_byte(io_channel channel, unsigned long offset, 59 static errcode_t inode_read_blk64(io_channel channel, 61 static errcode_t inode_write_blk64(io_channel channel, 115 static errcode_t inode_open(const char *name, int flags, io_channel *channel) [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/ |
voe_cpu_test.cc | 54 int channel(-1); 65 channel = base->CreateChannel(); 68 new VoiceChannelTransport(voe_network, channel)); 73 CHECK(codec->SetRecPayloadType(channel, isac)); 74 CHECK(codec->SetSendCodec(channel, isac)); 76 CHECK(base->StartReceive(channel)); 77 CHECK(base->StartPlayout(channel)); 78 CHECK(base->StartSend(channel)); 79 CHECK(file->StartPlayingFileAsMicrophone(channel, _mgr.AudioFilename(), 82 CHECK(codec->SetVADStatus(channel, true)) [all...] |
/external/mp4parser/isoparser/src/main/java/com/coremedia/iso/ |
ChannelHelper.java | 30 public static ByteBuffer readFully(final ReadableByteChannel channel, long size) throws IOException { 32 if (channel instanceof FileChannel && size > 1024 * 1024) { 33 ByteBuffer bb = ((FileChannel) channel).map(FileChannel.MapMode.READ_ONLY, ((FileChannel) channel).position(), size); 34 ((FileChannel) channel).position(((FileChannel) channel).position() + size); method 38 readFully(channel, buf, buf.limit()); 48 public static void readFully(final ReadableByteChannel channel, final ByteBuffer buf) 50 readFully(channel, buf, buf.remaining()); 53 public static int readFully(final ReadableByteChannel channel, final ByteBuffer buf, final int length [all...] |
/external/webrtc/src/modules/audio_processing/ |
audio_buffer.cc | 97 int16_t* AudioBuffer::data(int channel) const { 98 assert(channel >= 0 && channel < num_channels_); 103 return channels_[channel].data; 106 int16_t* AudioBuffer::low_pass_split_data(int channel) const { 107 assert(channel >= 0 && channel < num_channels_); 109 return data(channel); 112 return split_channels_[channel].low_pass_data; 115 int16_t* AudioBuffer::high_pass_split_data(int channel) const [all...] |
audio_buffer.h | 32 int16_t* data(int channel) const; 33 int16_t* low_pass_split_data(int channel) const; 34 int16_t* high_pass_split_data(int channel) const; 35 int16_t* mixed_data(int channel) const; 36 int16_t* mixed_low_pass_data(int channel) const; 37 int16_t* low_pass_reference(int channel) const; 39 int32_t* analysis_filter_state1(int channel) const; 40 int32_t* analysis_filter_state2(int channel) const; 41 int32_t* synthesis_filter_state1(int channel) const; 42 int32_t* synthesis_filter_state2(int channel) const [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_processing/ |
audio_buffer.h | 59 int16_t* data(int channel); 60 const int16_t* data(int channel) const; 61 int16_t* low_pass_split_data(int channel); 62 const int16_t* low_pass_split_data(int channel) const; 63 int16_t* high_pass_split_data(int channel); 64 const int16_t* high_pass_split_data(int channel) const; 65 const int16_t* mixed_data(int channel) const; 66 const int16_t* mixed_low_pass_data(int channel) const; 67 const int16_t* low_pass_reference(int channel) const; 71 float* data_f(int channel); [all...] |
audio_buffer.cc | 137 int16_t* low_channel(int i) { return low_.ibuf()->channel(i); } 138 int16_t* high_channel(int i) { return high_.ibuf()->channel(i); } 139 float* low_channel_f(int i) { return low_.fbuf()->channel(i); } 140 float* high_channel_f(int i) { return high_.fbuf()->channel(i); } 227 input_buffer_->channel(0), 237 process_buffer_->channel(i), 246 channels_->ibuf()->channel(i)); 263 ScaleToFloat(channels_->ibuf()->channel(i), 287 const int16_t* AudioBuffer::data(int channel) const { 288 assert(channel >= 0 && channel < num_proc_channels_) [all...] |