/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | acm_send_test_oldapi.cc | 50 int channels, 55 payload_name, &codec_, sampling_freq_hz, channels)); 59 input_frame_.num_channels_ = channels; 48 RegisterCodec(const char* payload_name, int sampling_freq_hz, int channels, int payload_type, int frame_size_samples) argument
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | PacketLossTest.cc | 30 int channels, 38 Receiver::Setup(acm, rtpStream, ss.str(), channels); 91 int channels, int expected_loss_rate) { 92 Sender::Setup(acm, rtpStream, in_file_name, sample_rate, channels); 112 PacketLossTest::PacketLossTest(int channels, int expected_loss_rate, argument 114 : channels_(channels), 27 Setup(AudioCodingModule *acm, RTPStream *rtpStream, std::string out_file_name, int channels, int loss_rate, int burst_length) argument 89 Setup(AudioCodingModule *acm, RTPStream *rtpStream, std::string in_file_name, int sample_rate, int channels, int expected_loss_rate) argument
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/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/chromium_org/third_party/webrtc/modules/audio_processing/ |
H A D | common.h | 37 // array of the deinterleaved channels. 58 ChannelBuffer(const T* const* channels, int samples_per_channel, argument 66 CopyFrom(channels[i], i); 89 T* const* channels() { return channels_.get(); } function in class:webrtc::ChannelBuffer 90 const T* const* channels() const { return channels_.get(); } function in class:webrtc::ChannelBuffer
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/external/kernel-headers/original/uapi/linux/hsi/ |
H A D | hsi_char.h | 53 uint32_t channels; member in struct:hsc_rx_config 58 uint32_t channels; member in struct:hsc_tx_config
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/external/libopus/celt/ |
H A D | opus_custom_demo.c | 51 opus_int32 frame_size, channels, rate; local 64 fprintf (stderr, "Usage: test_opus_custom <rate> <channels> <frame size> " 71 channels = atoi(argv[2]); 104 enc = opus_custom_encoder_create(mode, channels, &err); 112 dec = opus_custom_decoder_create(mode, channels, &err); 128 in = (opus_int16*)malloc(frame_size*channels*sizeof(opus_int16)); 129 out = (opus_int16*)malloc(frame_size*channels*sizeof(opus_int16)); 134 err = fread(in, sizeof(short), frame_size*channels, fin); 174 for (i=0;i<ret*channels;i++) 178 for (i=0;i<ret*channels; [all...] |
/external/libpng/contrib/gregbook/ |
H A D | readppm.c | 68 int bit_depth, color_type, channels; variable 97 channels = 3; 100 channels = 4; 103 channels = 1; 152 /* GRR WARNING: grayscale needs to be expanded and channels reset! */ 154 *pRowbytes = rowbytes = channels*width; 155 *pChannels = channels;
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/external/libpng/ |
H A D | pngwtran.c | 21 * row_info bit depth should be 8 (one pixel per byte). The channels 30 row_info->channels == 1) 154 row_info->pixel_depth = (png_byte)(bit_depth * row_info->channels); 178 int channels = 0; local 182 shift_start[channels] = row_info->bit_depth - bit_depth->red; 183 shift_dec[channels] = bit_depth->red; 184 channels++; 186 shift_start[channels] = row_info->bit_depth - bit_depth->green; 187 shift_dec[channels] = bit_depth->green; 188 channels [all...] |
/external/qemu/distrib/sdl-1.2.15/src/audio/ |
H A D | SDL_wave.h | 49 Uint16 channels; /* 1 = mono, 2 = stereo */ member in struct:WaveFMT
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/external/chromium_org/chrome/browser/copresence/ |
H A D | chrome_whispernet_client_browsertest.cc | 36 buffer.resize(source->frames() * source->channels() * sizeof(float)); 39 const int channels = source->channels(); local 40 for (int ch = 0; ch < channels; ++ch) { 41 for (int si = 0, di = ch; si < source->frames(); ++si, di += channels)
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/external/chromium_org/chrome/renderer/media/ |
H A D | cast_rtp_stream.h | 64 // Number of audio channels. 65 int channels; member in struct:CastRtpPayloadParams
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/external/chromium_org/components/copresence/mediums/audio/ |
H A D | audio_recorder.cc | 28 buffer->resize(source->frames() * source->channels() * sizeof(float)); 31 const int channels = source->channels(); local 32 for (int ch = 0; ch < channels; ++ch) { 33 for (int si = 0, di = ch; si < source->frames(); ++si, di += channels) 118 media::AudioBus::Create(dest_params.channels(), total_buffer_frames_);
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/external/chromium_org/content/browser/speech/ |
H A D | speech_recognizer_impl_unittest.cc | 71 const int channels = local 74 const int frames = audio_packet_length_bytes / channels / bytes_per_sample_; 75 audio_bus_ = media::AudioBus::Create(channels, frames);
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/external/chromium_org/media/audio/alsa/ |
H A D | alsa_util.cc | 17 int channels, 30 SND_PCM_ACCESS_RW_INTERLEAVED, channels, 35 << " Channels: " << channels << " Latency: " << latency_us; 97 int channels, 101 return OpenDevice(wrapper, device_name, SND_PCM_STREAM_CAPTURE, channels, 107 int channels, 111 return OpenDevice(wrapper, device_name, SND_PCM_STREAM_PLAYBACK, channels, 14 OpenDevice(media::AlsaWrapper* wrapper, const char* device_name, snd_pcm_stream_t type, int channels, int sample_rate, snd_pcm_format_t pcm_format, int latency_us) argument 95 OpenCaptureDevice(media::AlsaWrapper* wrapper, const char* device_name, int channels, int sample_rate, snd_pcm_format_t pcm_format, int latency_us) argument 105 OpenPlaybackDevice(media::AlsaWrapper* wrapper, const char* device_name, int channels, int sample_rate, snd_pcm_format_t pcm_format, int latency_us) argument
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/external/chromium_org/media/audio/ |
H A D | audio_parameters.h | 64 int channels, int sample_rate, int bits_per_sample, 68 int channels, int sample_rate, int bits_per_sample, 96 int channels() const { return channels_; } function in class:media::AudioParameters 103 ChannelLayout channel_layout_; // Order of surround sound channels. 108 int channels_; // Number of channels. Value set based on 117 if (a.channels() != b.channels()) 118 return a.channels() < b.channels();
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/external/chromium_org/media/base/android/java/src/org/chromium/media/ |
H A D | AudioRecordInput.java | 98 int sampleRate, int channels, int bitsPerSample, int bytesPerBuffer, 100 return new AudioRecordInput(nativeAudioRecordInputStream, sampleRate, channels, 104 private AudioRecordInput(long nativeAudioRecordInputStream, int sampleRate, int channels, argument 108 mChannels = channels; 141 Log.e(TAG, "Unsupported number of channels: " + mChannels); 97 createAudioRecordInput(long nativeAudioRecordInputStream, int sampleRate, int channels, int bitsPerSample, int bytesPerBuffer, boolean usePlatformAEC) argument
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/external/chromium_org/media/base/ |
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...] |
H A D | multi_channel_resampler_unittest.cc | 45 void InitializeAudioData(int channels, int frames) { argument 47 audio_bus_ = AudioBus::Create(channels, frames); 57 EXPECT_EQ(audio_bus->channels(), audio_bus_->channels()); 58 for (int i = 0; i < audio_bus->channels(); ++i) 63 void MultiChannelTest(int channels, int frames, double expected_max_rms_error, argument 65 InitializeAudioData(channels, frames); 67 channels, kScaleFactor, SincResampler::kDefaultRequestSize, base::Bind( 85 void HighLatencyTest(int channels) { argument 86 MultiChannelTest(channels, kHighLatencySiz 90 LowLatencyTest(int channels) argument [all...] |
/external/chromium_org/media/cast/ |
H A D | cast_config.h | 54 int channels; member in struct:media::cast::AudioSenderConfig 141 // Number of channels. For audio, this is normally 2. For video, this must 143 int channels; member in struct:media::cast::FrameReceiverConfig
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/external/chromium_org/media/filters/ |
H A D | audio_file_reader.cc | 108 codec_context_->channel_layout, codec_context_->channels) == 114 channels_ = codec_context_->channels; 131 DCHECK_EQ(audio_bus->channels(), channels()); 132 if (audio_bus->channels() != channels()) 181 int channels = av_get_channel_layout_nb_channels( local 184 int channels = av_frame->channels; local 187 channels ! 213 int channels = audio_bus->channels(); local [all...] |
H A D | audio_file_reader_unittest.cc | 35 AudioBus::Create(reader_->channels(), reader_->GetNumberOfFrames()); 84 int channels, 91 EXPECT_EQ(channels, reader_->channels()); 110 AudioBus::Create(reader_->channels(), reader_->GetNumberOfFrames()); 82 RunTest(const char* fn, const char* hash, int channels, int sample_rate, base::TimeDelta duration, int frames, int trimmed_frames) argument
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/external/chromium_org/third_party/WebKit/Source/core/dom/ |
H A D | MessagePort.cpp | 72 OwnPtr<MessagePortChannelArray> channels; local 82 channels = MessagePort::disentanglePorts(ports, exceptionState); 88 OwnPtr<WebMessagePortChannelArray> webChannels = toWebMessagePortChannelArray(channels.release()); 93 PassOwnPtr<WebMessagePortChannelArray> MessagePort::toWebMessagePortChannelArray(PassOwnPtr<MessagePortChannelArray> channels) argument 96 if (channels && channels->size()) { 97 webChannels = adoptPtr(new WebMessagePortChannelArray(channels->size())); 98 for (size_t i = 0; i < channels->size(); ++i) 99 (*webChannels)[i] = (*channels)[i].leakPtr(); 109 OwnPtr<MessagePortChannelArray> channels local 168 tryGetMessageFrom(WebMessagePortChannel& webChannel, RefPtr<SerializedScriptValue>& message, OwnPtr<MessagePortChannelArray>& channels) argument 192 OwnPtr<MessagePortChannelArray> channels; local 244 entanglePorts(ExecutionContext& context, PassOwnPtr<MessagePortChannelArray> channels) argument [all...] |
/external/chromium_org/third_party/WebKit/Source/core/events/ |
H A D | MessageEvent.cpp | 87 MessageEvent::MessageEvent(PassRefPtr<SerializedScriptValue> data, const String& origin, const String& lastEventId, PassRefPtrWillBeRawPtr<EventTarget> source, PassOwnPtr<MessagePortChannelArray> channels) argument 94 , m_channels(channels)
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H A D | MessageEvent.h | 65 static PassRefPtrWillBeRawPtr<MessageEvent> create(PassOwnPtr<MessagePortChannelArray> channels, PassRefPtr<SerializedScriptValue> data, const String& origin = String(), const String& lastEventId = String(), PassRefPtrWillBeRawPtr<EventTarget> source = nullptr) argument 67 return adoptRefWillBeNoop(new MessageEvent(data, origin, lastEventId, source, channels)); 91 MessagePortChannelArray* channels() const { return m_channels ? m_channels.get() : 0; } function in class:blink::FINAL
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/external/chromium_org/third_party/WebKit/Source/core/workers/ |
H A D | DedicatedWorkerGlobalScope.cpp | 68 OwnPtr<MessagePortChannelArray> channels = MessagePort::disentanglePorts(ports, exceptionState); local 71 thread()->workerObjectProxy().postMessageToWorkerObject(message, channels.release());
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