/external/chromium_org/content/child/ |
H A D | webmessageportchannel_impl.h | 46 blink::WebMessagePortChannelArray* channels); 61 blink::WebMessagePortChannelArray* channels); 63 blink::WebMessagePortChannelArray& channels); 69 blink::WebMessagePortChannelArray* channels);
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/external/chromium_org/media/base/ |
H A D | audio_block_fifo.h | 22 // of memory can store |channels| of length |frames| data. 23 AudioBlockFifo(int channels, int frames, int blocks); 55 // Number of channels in AudioBus.
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H A D | audio_fifo.h | 20 // Creates a new AudioFifo and allocates |channels| of length |frames|. 21 AudioFifo(int channels, int frames);
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H A D | audio_pull_fifo.cc | 14 AudioPullFifo::AudioPullFifo(int channels, int frames, const ReadCB& read_cb) argument 16 fifo_(AudioBus::Create(channels, frames)), 56 for (int ch = 0; ch < fifo_->channels(); ++ch) {
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H A D | audio_pull_fifo_unittest.cc | 20 // Number of channels in each audio bus. 60 // audio frame that we provide. Note that all channels are given the same 66 EXPECT_EQ(audio_bus->channels(), audio_bus_->channels()); 69 for (int j = 0; j < audio_bus->channels(); ++j) { 70 // Store same value in all channels.
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H A D | channel_layout.cc | 161 ChannelLayout GuessChannelLayout(int channels) { argument 162 switch (channels) { 180 DVLOG(1) << "Unsupported channel count: " << channels;
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H A D | audio_buffer_unittest.cc | 20 for (int ch = 0; ch < bus->channels(); ++ch) { 35 const int channels = ChannelLayoutToChannelCount(channel_layout); local 41 channels, 51 scoped_ptr<AudioBus> bus = AudioBus::Create(channels, frames); 203 EXPECT_EQ(16, buffer->frame_count()); // 2 channels of 8-bit data 212 EXPECT_EQ(2, buffer->frame_count()); // now 4 channels of 32-bit data 217 const int channels = ChannelLayoutToChannelCount(channel_layout); local 222 channels, 228 scoped_ptr<AudioBus> bus = AudioBus::Create(channels, frames); 241 const int channels local 265 const int channels = ChannelLayoutToChannelCount(channel_layout); local 288 const int channels = ChannelLayoutToChannelCount(channel_layout); local 311 const int channels = ChannelLayoutToChannelCount(channel_layout); local 345 const int channels = ChannelLayoutToChannelCount(channel_layout); local 373 const int channels = ChannelLayoutToChannelCount(channel_layout); local 391 const int channels = ChannelLayoutToChannelCount(channel_layout); local [all...] |
/external/chromium_org/media/filters/ |
H A D | ffmpeg_audio_decoder.cc | 33 // Return the number of channels from the data in |frame|. 36 // When use_system_ffmpeg==1, libav's AVFrame doesn't have channels field. 39 return frame->channels; 62 int channels = DetermineChannels(frame); local 63 if (channels <= 0 || channels >= limits::kMaxChannels) { 64 DLOG(ERROR) << "Requested number of channels (" << channels 73 if (s->channels != channels) { 281 const int channels = DetermineChannels(av_frame_.get()); local [all...] |
/external/chromium_org/remoting/codec/ |
H A D | audio_encoder_verbatim.cc | 22 DCHECK_NE(AudioPacket::CHANNELS_INVALID, packet->channels());
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/external/chromium_org/third_party/WebKit/public/web/ |
H A D | WebServiceWorkerContextProxy.h | 52 virtual void dispatchMessageEvent(const WebString& message, const WebMessagePortChannelArray& channels) = 0;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/opus/ |
H A D | opus_interface.c | 34 int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst, int32_t channels) { argument 41 int application = (channels == 1) ? OPUS_APPLICATION_VOIP : 44 state->encoder = opus_encoder_create(48000, channels, application, 147 int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) { argument 160 state->decoder_left = opus_decoder_create(48000, channels, &error_l); 161 state->decoder_right = opus_decoder_create(48000, channels, &error_r); 165 state->channels = channels; 195 return inst->channels; 296 if (inst->channels 495 int frames, channels, payload_length_ms; local [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
H A D | input_audio_file.h | 38 // |channels| times to create an interleaved multi-channel signal where all 39 // channels are identical. The output |destination| must have the capacity to 40 // hold samples * channels elements. Note that |source| and |destination| can 43 size_t channels, int16_t* destination);
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/external/libvpx/libvpx/third_party/libmkv/ |
H A D | WebMElement.h | 24 double samplingFrequency, unsigned int channels,
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/external/mp4parser/isoparser/src/main/java/com/coremedia/iso/ |
H A D | BoxParser.java | 22 import java.nio.channels.ReadableByteChannel;
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H A D | ChannelHelper.java | 21 import java.nio.channels.FileChannel; 22 import java.nio.channels.ReadableByteChannel; 23 import java.nio.channels.SelectionKey; 24 import java.nio.channels.WritableByteChannel;
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/external/chromium_org/content/test/ |
H A D | blink_test_environment.cc | 40 void EnableBlinkPlatformLogChannels(const std::string& channels) { argument 41 if (channels.empty()) 43 base::StringTokenizer t(channels, ", ");
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/external/chromium_org/media/audio/ |
H A D | simple_sources.cc | 19 SineWaveAudioSource::SineWaveAudioSource(int channels, argument 21 : channels_(channels), 44 for (int i = 1; i < audio_bus->channels(); ++i) {
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/external/chromium_org/sandbox/win/src/ |
H A D | sharedmem_ipc_client.cc | 24 control_->channels[ix].channel_base; 32 ChannelControl* channel = control_->channels; 44 control_->channels[0].channel_base; 58 ChannelControl* channel = control_->channels; 112 // Locking a channel is a simple as looping over all the channels 122 ChannelControl* channel = control_->channels;
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
H A D | rtp_payload_registry.h | 30 const uint8_t channels, 40 const uint8_t channels, 62 const uint8_t channels, 72 const uint8_t channels, 145 const uint8_t channels,
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/external/chromium_org/third_party/opus/src/include/ |
H A D | opus.h | 61 * @li Support for multichannel (up to 255 channels) 86 * enc = opus_encoder_create(Fs, channels, application, &error); 100 * size = opus_encoder_get_size(channels); 102 * error = opus_encoder_init(enc, Fs, channels, application); 167 * @param[in] channels <tt>int</tt>: Number of channels. 171 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels); 200 * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal 203 * @note Regardless of the sampling rate and number channels selecte [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | dtmf_tone_generator_unittest.cc | 31 void TestAllTones(int fs_hz, int channels) { argument 32 AudioMultiVector signal(channels); 52 for (int channel = 0; channel < channels; ++channel) { 62 void TestAmplitudes(int fs_hz, int channels) { argument 63 AudioMultiVector signal(channels); 64 AudioMultiVector ref_signal(channels); 84 for (int channel = 0; channel < channels; ++channel) {
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/external/libopus/include/ |
H A D | opus.h | 61 * @li Support for multichannel (up to 255 channels) 86 * enc = opus_encoder_create(Fs, channels, application, &error); 100 * size = opus_encoder_get_size(channels); 102 * error = opus_encoder_init(enc, Fs, channels, application); 167 * @param[in] channels <tt>int</tt>: Number of channels. 171 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels); 200 * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal 203 * @note Regardless of the sampling rate and number channels selecte [all...] |
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_payload_registry.cc | 42 const uint8_t channels, 89 channels, rate)) { 100 payload_name, payload_name_length, frequency, channels, rate); 121 payload_name, payload_type, frequency, channels, rate); 143 // There can't be several codecs with the same rate, frequency and channels 150 const uint8_t channels, 164 channels, rate)) { 182 const uint8_t channels, 203 // [default] audio, check freq and channels. 205 payload->typeSpecific.Audio.channels 38 RegisterReceivePayload( const char payload_name[RTP_PAYLOAD_NAME_SIZE], const int8_t payload_type, const uint32_t frequency, const uint8_t channels, const uint32_t rate, bool* created_new_payload) argument 146 DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType( const char payload_name[RTP_PAYLOAD_NAME_SIZE], const size_t payload_name_length, const uint32_t frequency, const uint8_t channels, const uint32_t rate) argument 179 ReceivePayloadType( const char payload_name[RTP_PAYLOAD_NAME_SIZE], const uint32_t frequency, const uint8_t channels, const uint32_t rate, int8_t* payload_type) const argument [all...] |
/external/chromium_org/third_party/opus/src/src/ |
H A D | opus_encoder.c | 66 int channels; member in struct:OpusEncoder 148 int opus_encoder_get_size(int channels) argument 152 if (channels<1 || channels > 2) 158 celtEncSizeBytes = celt_encoder_get_size(channels); 162 int opus_encoder_init(OpusEncoder* st, opus_int32 Fs, int channels, int application) argument 169 if((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000)||(channels!=1&&channels!=2)|| 174 OPUS_CLEAR((char*)st, opus_encoder_get_size(channels)); 185 st->stream_channels = st->channels 248 gen_toc(int mode, int framerate, int bandwidth, int channels) argument 320 hp_cutoff(const opus_val16 *in, opus_int32 cutoff_Hz, opus_val16 *out, opus_val32 *hp_mem, int len, int channels, opus_int32 Fs) argument 356 dc_reject(const opus_val16 *in, opus_int32 cutoff_Hz, opus_val16 *out, opus_val32 *hp_mem, int len, int channels, opus_int32 Fs) argument 381 dc_reject(const opus_val16 *in, opus_int32 cutoff_Hz, opus_val16 *out, opus_val32 *hp_mem, int len, int channels, opus_int32 Fs) argument 405 stereo_fade(const opus_val16 *in, opus_val16 *out, opus_val16 g1, opus_val16 g2, int overlap48, int frame_size, int channels, const opus_val16 *window, opus_int32 Fs) argument 437 gain_fade(const opus_val16 *in, opus_val16 *out, opus_val16 g1, opus_val16 g2, int overlap48, int frame_size, int channels, const opus_val16 *window, opus_int32 Fs) argument 476 opus_encoder_create(opus_int32 Fs, int channels, int application, int *error) argument [all...] |
/external/libopus/src/ |
H A D | opus_encoder.c | 66 int channels; member in struct:OpusEncoder 148 int opus_encoder_get_size(int channels) argument 152 if (channels<1 || channels > 2) 158 celtEncSizeBytes = celt_encoder_get_size(channels); 162 int opus_encoder_init(OpusEncoder* st, opus_int32 Fs, int channels, int application) argument 169 if((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000)||(channels!=1&&channels!=2)|| 174 OPUS_CLEAR((char*)st, opus_encoder_get_size(channels)); 185 st->stream_channels = st->channels 248 gen_toc(int mode, int framerate, int bandwidth, int channels) argument 320 hp_cutoff(const opus_val16 *in, opus_int32 cutoff_Hz, opus_val16 *out, opus_val32 *hp_mem, int len, int channels, opus_int32 Fs) argument 356 dc_reject(const opus_val16 *in, opus_int32 cutoff_Hz, opus_val16 *out, opus_val32 *hp_mem, int len, int channels, opus_int32 Fs) argument 381 dc_reject(const opus_val16 *in, opus_int32 cutoff_Hz, opus_val16 *out, opus_val32 *hp_mem, int len, int channels, opus_int32 Fs) argument 405 stereo_fade(const opus_val16 *in, opus_val16 *out, opus_val16 g1, opus_val16 g2, int overlap48, int frame_size, int channels, const opus_val16 *window, opus_int32 Fs) argument 437 gain_fade(const opus_val16 *in, opus_val16 *out, opus_val16 g1, opus_val16 g2, int overlap48, int frame_size, int channels, const opus_val16 *window, opus_int32 Fs) argument 476 opus_encoder_create(opus_int32 Fs, int channels, int application, int *error) argument [all...] |