/external/chromium_org/third_party/webrtc/voice_engine/ |
channel_manager.cc | 69 channels_.push_back(channel_owner); 77 for (size_t i = 0; i < channels_.size(); ++i) { 78 if (channels_[i].channel()->ChannelId() == channel_id) 79 return channels_[i]; 87 *channels = channels_; 98 for (std::vector<ChannelOwner>::iterator it = channels_.begin(); 99 it != channels_.end(); 103 channels_.erase(it); 116 references = channels_; 117 channels_.clear() [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
acm_celt.cc | 31 channels_(1) { 70 channels_(1) { // Default send mono. 96 in_audio_ix_read_ += frame_len_smpl_ * channels_; 117 if (codec_params->codec_inst.channels != channels_) { 121 channels_ = codec_params->codec_inst.channels; 122 if (WebRtcCelt_CreateEnc(&enc_inst_ptr_, channels_) < 0) { 128 if (WebRtcCelt_EncoderInit(enc_inst_ptr_, channels_, bitrate_) >= 0) { 145 channels_ = num_channels_; 165 if (WebRtcCelt_EncoderInit(enc_inst_ptr_, channels_, bitrate_) >= 0) {
|
acm_celt.h | 45 uint16_t channels_; member in class:webrtc::acm2::ACMCELT
|
/external/chromium_org/third_party/webrtc/modules/audio_processing/ |
common.h | 43 channels_(new T*[num_channels]), 51 channels_(new T*[num_channels]), 61 channels_(new T*[num_channels]), 73 memcpy(channels_[i], channel_ptr, samples_per_channel_ * sizeof(T)); 82 return channels_[i]; 89 T* const* channels() { return channels_.get(); } 90 const T* const* channels() const { return channels_.get(); } 100 channels_[i] = &data_[i * samples_per_channel_]; 104 scoped_ptr<T*[]> channels_; member in class:webrtc::ChannelBuffer
|
/external/chromium_org/remoting/codec/ |
audio_encoder_opus.cc | 44 channels_(AudioPacket::CHANNELS_STEREO), 59 encoder_ = opus_encoder_create(kOpusSamplingRate, channels_, 73 new char[kFrameSamples * kBytesPerSample * channels_]); 77 channels_, 82 resampler_bus_ = media::AudioBus::Create(channels_, kFrameSamples); 90 new int16[leftover_buffer_size_ * channels_]); 103 if (packet->channels() != channels_ || 107 channels_ = packet->channels(); 110 if (channels_ <= 0 || channels_ > 2 | [all...] |
audio_decoder_opus.cc | 30 channels_(0), 41 decoder_ = opus_decoder_create(kSamplingRate, channels_, &error); 55 if (packet->channels() != channels_ || 59 channels_ = packet->channels(); 62 if (channels_ <= 0 || channels_ > 2 || 65 << channels_ << " channels with " 104 int max_frame_bytes = max_frame_samples * channels_ *
|
audio_decoder_opus.h | 33 int channels_; member in class:remoting::AudioDecoderOpus
|
/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
fakewebrtccommon.h | 65 if (channels_.find(channel) == channels_.end()) return -1; 68 ASSERT(channels_.find(channel) != channels_.end());
|
fakewebrtcvideoengine.h | 393 ASSERT(0 == channels_.size()); 401 for (std::map<int, Channel*>::const_iterator iter = channels_.begin(); 402 iter != channels_.end(); ++iter) { 410 int GetNumChannels() const { return static_cast<int>(channels_.size()); } 412 return (channels_.find(channel) != channels_.end()); 437 return channels_.find(channel)->second->capture_id_; 441 return channels_.find(channel)->second->original_channel_id_; 445 return channels_.find(channel)->second->has_renderer_; 449 return channels_.find(channel)->second->render_started_ 1291 std::map<int, Channel*> channels_; member in class:cricket::FakeWebRtcVideoEngine [all...] |
fakewebrtcvoiceengine.h | 278 for (std::map<int, Channel*>::const_iterator i = channels_.begin(); 279 i != channels_.end(); ++i) { 290 for (std::map<int, Channel*>::const_iterator iter = channels_.begin(); 291 iter != channels_.end(); ++iter) { 297 int GetNumChannels() const { return static_cast<int>(channels_.size()); } 299 return channels_[channel]->playout; 302 return channels_[channel]->send; 308 return channels_[channel]->vad; 311 return channels_[channel]->red; 314 return channels_[channel]->codec_fec 1295 std::map<int, Channel*> channels_; member in class:cricket::FakeWebRtcVoiceEngine [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
audio_multi_vector.cc | 25 channels_.push_back(new AudioVector); 34 channels_.push_back(new AudioVector(initial_size)); 40 std::vector<AudioVector*>::iterator it = channels_.begin(); 41 while (it != channels_.end()) { 49 channels_[i]->Clear(); 55 channels_[i]->Clear(); 56 channels_[i]->Extend(length); 63 channels_[i]->CopyTo(&(*copy_to)[i]); 73 channels_[0]->PushBack(append_this, length); 86 channels_[channel]->PushBack(temp_array, length_per_channel) [all...] |
/external/chromium_org/third_party/webrtc/test/ |
fake_common.h | 43 if (channels_.find(channel) == channels_.end()) return -1; 46 ASSERT(channels_.find(channel) != channels_.end());
|
/external/chromium_org/media/formats/webm/ |
webm_audio_client.cc | 22 channels_ = -1; 48 if (channels_ == -1) 49 channels_ = 1; 51 ChannelLayout channel_layout = GuessChannelLayout(channels_); 54 MEDIA_LOG(log_cb_) << "Unsupported channel count " << channels_; 102 if (channels_ != -1) { 104 << " specified. (" << channels_ << " and " << val 109 channels_ = val;
|
/external/chromium_org/media/audio/ |
audio_parameters.cc | 18 channels_(0), 30 channels_(ChannelLayoutToChannelCount(channel_layout)), 42 channels_(ChannelLayoutToChannelCount(channel_layout)), 55 channels_(channels), 69 channels_ = channels; 78 (channels_ > 0) && 79 (channels_ <= media::limits::kMaxChannels) && 99 return channels_ * bits_per_sample_ / 8; 112 channels_ == other.channels() &&
|
simple_sources.h | 39 int channels_; member in class:media::SineWaveAudioSource
|
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/tools/ |
audio_codec_speed_test.cc | 38 channels_ = get<0>(GetParam()); 53 input_length_sample_ * channels_]); 66 input_length_sample_ * channels_ * sizeof(int16_t)); 68 max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t); 69 out_data_.reset(new int16_t[output_length_sample_ * channels_]); 101 input_sampling_khz_, channels_, bit_rate_); 112 output_length_sample_ * channels_, out_file_); 114 data_pointer_ = (data_pointer_ + input_length_sample_ * channels_) %
|
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/opus/ |
opus_fec_test.cc | 50 int channels_; member in class:webrtc::OpusFecTest 69 channels_ = get<0>(GetParam()); 71 printf("Coding %d channel signal at %d bps.\n", channels_, bit_rate_); 85 block_length_sample_ * channels_]); 98 block_length_sample_ * channels_ * sizeof(int16_t)); 101 max_bytes_ = block_length_sample_ * channels_ * sizeof(int16_t); 103 out_data_.reset(new int16_t[2 * block_length_sample_ * channels_]); 107 EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_)); 108 EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_)); 161 &out_data_[value_1 * channels_], [all...] |
/external/chromium_org/media/filters/ |
audio_file_reader.h | 48 int channels() const { return channels_; } 88 int channels_; member in class:media::AudioFileReader
|
audio_renderer_algorithm.cc | 76 : channels_(0), 95 channels_ = params.channels(); 134 wsola_output_ = AudioBus::Create(channels_, ola_window_size_ + ola_hop_size_); 138 optimal_block_ = AudioBus::Create(channels_, ola_window_size_); 140 channels_, num_candidate_blocks_ + (ola_window_size_ - 1)); 141 target_block_ = AudioBus::Create(channels_, ola_window_size_); 150 DCHECK_EQ(channels_, dest->channels()); 244 for (int k = 0; k < channels_; ++k) { 300 for (int k = 0; k < channels_; ++k) { 353 for (int k = 0; k < channels_; ++k) [all...] |
/external/chromium_org/remoting/protocol/ |
fake_datagram_socket.cc | 111 for (ChannelsMap::iterator it = channels_.begin(); it != channels_.end(); 125 return channels_[name].get(); 131 EXPECT_TRUE(channels_[name] == NULL); 134 channels_[name] = channel->GetWeakPtr(); 160 if (channels_.find(name) != channels_.end()) 166 channels_.erase(name);
|
/external/chromium_org/media/base/ |
audio_block_fifo.h | 56 const int channels_; member in class:media::AudioBlockFifo
|
/external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
transport.cc | 189 if (channels_.empty()) 192 ChannelMap::iterator iter = channels_.begin(); 226 if (channels_.find(component) == channels_.end()) { 228 channels_[component] = ChannelMapEntry(impl); 230 impl = channels_[component].get(); 235 channels_[component].AddRef(); 270 if (channels_.size() == 1) { 281 ChannelMap::iterator iter = channels_.find(component); 282 return (iter != channels_.end()) ? iter->second.get() : NULL [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
PacketLossTest.cc | 114 : channels_(channels), 115 in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz" : 131 int codec_id = acm->Codec("opus", 48000, channels_); 143 sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_, 159 receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_,
|
/external/webrtc/src/modules/audio_processing/ |
audio_buffer.cc | 77 channels_(NULL), 83 channels_.reset(new AudioChannel[max_num_channels_]); 103 return channels_[channel].data; 213 int16_t* deinterleaved = channels_[i].data; 234 channels_[0].data, 246 int16_t* deinterleaved = channels_[i].data; 263 StereoToMono(channels_[0].data, 264 channels_[1].data, 265 channels_[0].data, 277 StereoToMono(channels_[0].data [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/interface/ |
audio_decoder.h | 68 channels_(1), 138 size_t channels() const { return channels_; } 144 size_t channels_; member in class:webrtc::AudioDecoder
|