| /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));
|
| /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));
|
| mock_voe_connection_observer.h | 22 MOCK_METHOD2(OnPeriodicDeadOrAlive, void(int channel,
|
| /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;
|
| 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) {
|
| /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);
|
| 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...] |
