| /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/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
|
| audio_encoder_opus.h | 41 AudioPacket::Channels channels_; member in class:remoting::AudioEncoderOpus
|
| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| fakewebrtccommon.h | 60 if (channels_.find(channel) == channels_.end()) return -1;
63 ASSERT(channels_.find(channel) != channels_.end());
|
| fakewebrtcvideoengine.h | 392 ASSERT(0 == channels_.size());
400 for (std::map<int, Channel*>::const_iterator iter = channels_.begin();
401 iter != channels_.end(); ++iter) {
409 int GetNumChannels() const { return static_cast<int>(channels_.size()); }
411 return (channels_.find(channel) != channels_.end());
436 return channels_.find(channel)->second->capture_id_;
440 return channels_.find(channel)->second->original_channel_id_;
444 return channels_.find(channel)->second->has_renderer_;
448 return channels_.find(channel)->second->render_started_
1275 std::map<int, Channel*> channels_; member in class:cricket::FakeWebRtcVideoEngine
[all...] |
| fakewebrtcvoiceengine.h | 188 for (std::map<int, Channel*>::const_iterator i = channels_.begin();
189 i != channels_.end(); ++i) {
200 for (std::map<int, Channel*>::const_iterator iter = channels_.begin();
201 iter != channels_.end(); ++iter) {
207 int GetNumChannels() const { return static_cast<int>(channels_.size()); }
209 return channels_[channel]->playout;
212 return channels_[channel]->send;
218 return channels_[channel]->vad;
221 return channels_[channel]->red;
224 return channels_[channel]->codec_fec
1164 std::map<int, Channel*> channels_; member in class:cricket::FakeWebRtcVoiceEngine
[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;
47 if (channels_ == -1)
48 channels_ = 1;
50 ChannelLayout channel_layout = GuessChannelLayout(channels_);
53 MEDIA_LOG(log_cb_) << "Unsupported channel count " << channels_;
94 if (channels_ != -1) {
96 << " specified. (" << channels_ << " and " << val
101 channels_ = val;
|
| webm_audio_client.h | 45 int channels_; member in class:media::WebMAudioClient
|
| /external/chromium_org/third_party/webrtc/modules/audio_processing/ |
| common.h | 42 channels_(new T*[num_channels]),
47 channels_[i] = &data_[i * samples_per_channel];
53 memcpy(channels_[i], channel_ptr, samples_per_channel_ * sizeof(T));
59 return channels_[i];
61 T** channels() { return channels_.get(); }
69 scoped_ptr<T*[]> channels_;
|
| /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]->CopyFrom(&(*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/modules/audio_coding/main/acm2/ |
| acm_celt.cc | 31 channels_(1) {
74 channels_(1) { // Default send mono.
100 in_audio_ix_read_ += frame_len_smpl_ * channels_;
121 if (codec_params->codec_inst.channels != channels_) {
125 channels_ = codec_params->codec_inst.channels;
126 if (WebRtcCelt_CreateEnc(&enc_inst_ptr_, channels_) < 0) {
132 if (WebRtcCelt_EncoderInit(enc_inst_ptr_, channels_, bitrate_) >= 0) {
149 channels_ = num_channels_;
176 if (WebRtcCelt_EncoderInit(enc_inst_ptr_, channels_, bitrate_) >= 0) {
|
| /external/chromium_org/media/audio/ |
| audio_parameters.cc | 18 channels_(0),
31 channels_(ChannelLayoutToChannelCount(channel_layout)),
45 channels_(ChannelLayoutToChannelCount(channel_layout)),
59 channels_(channels),
75 channels_ = channels;
85 (channels_ > 0) &&
86 (channels_ <= media::limits::kMaxChannels) &&
108 return channels_ * bits_per_sample_ / 8;
|
| simple_sources.h | 39 int channels_; member in class:media::SineWaveAudioSource
|
| /external/chromium_org/media/filters/ |
| audio_file_reader.h | 47 int channels() const { return channels_; }
75 int channels_; member in class:media::AudioFileReader
|
| audio_renderer_algorithm.cc | 76 : channels_(0),
98 channels_ = params.channels();
138 wsola_output_ = AudioBus::Create(channels_, ola_window_size_ + ola_hop_size_);
142 optimal_block_ = AudioBus::Create(channels_, ola_window_size_);
144 channels_, num_candidate_blocks_ + (ola_window_size_ - 1));
145 target_block_ = AudioBus::Create(channels_, ola_window_size_);
152 DCHECK_EQ(channels_, dest->channels());
256 for (int k = 0; k < channels_; ++k) {
311 for (int k = 0; k < channels_; ++k) {
364 for (int k = 0; k < channels_; ++k)
[all...] |
| /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/neteq/tools/ |
| neteq_quality_test.cc | 35 channels_(channels),
49 max_payload_bytes_ = in_size_samples_ * channels_ * sizeof(int16_t);
50 in_data_.reset(new int16_t[in_size_samples_ * channels_]);
52 out_data_.reset(new int16_t[out_size_samples_ * channels_]);
83 int ret = neteq_->GetAudio(out_size_samples_ * channels_, &out_data_[0],
89 assert(channels == channels_);
101 ASSERT_TRUE(in_file_->Read(in_size_samples_ * channels_, &in_data_[0]));
|
| /external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
| transport.cc | 171 if (channels_.empty())
174 ChannelMap::iterator iter = channels_.begin();
208 if (channels_.find(component) == channels_.end()) {
210 channels_[component] = ChannelMapEntry(impl);
212 impl = channels_[component].get();
217 channels_[component].AddRef();
252 if (channels_.size() == 1) {
263 ChannelMap::iterator iter = channels_.find(component);
264 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/chromium_org/third_party/webrtc/modules/audio_coding/codecs/opus/ |
| opus_fec_test.cc | 59 int channels_; member in class:webrtc::OpusFecTest
78 channels_ = get<0>(GetParam());
80 printf("Coding %d channel signal at %d bps.\n", channels_, bit_rate_);
94 input_length_sample_ * channels_]);
107 input_length_sample_ * channels_ * sizeof(int16_t));
110 max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t);
112 out_data_.reset(new int16_t[2 * output_length_sample_ * channels_]);
116 EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_));
117 EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
172 &out_data_[value_1 * channels_],
[all...] |
| /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
|
