| /external/webrtc/webrtc/voice_engine/ |
| channel_manager.cc | 71 channels_.push_back(channel_owner);
79 for (size_t i = 0; i < channels_.size(); ++i) {
80 if (channels_[i].channel()->ChannelId() == channel_id)
81 return channels_[i];
89 *channels = channels_;
99 std::vector<ChannelOwner>::iterator to_delete = channels_.end();
100 for (auto it = channels_.begin(); it != channels_.end(); ++it) {
110 if (to_delete != channels_.end()) {
112 channels_.erase(to_delete)
[all...] |
| channel_manager.h | 92 std::vector<ChannelOwner> channels_; member in class:webrtc::voe::ChannelManager::Iterator
127 std::vector<ChannelOwner> channels_; member in class:webrtc::voe::ChannelManager
|
| /external/webrtc/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...] |
| sync_buffer.h | 79 const AudioVector& Channel(size_t n) const { return *channels_[n]; }
80 AudioVector& Channel(size_t n) { return *channels_[n]; }
|
| audio_decoder_unittest.cc | 107 channels_(1),
150 new int16_t[channels_ * samples_per_10ms]);
157 samples_per_10ms, channels_,
195 // Make sure that frame_size_ * channels_ samples are allocated and free.
196 decoded.resize((processed_samples + frame_size_) * channels_, 0);
200 frame_size_ * channels_ * sizeof(int16_t),
201 &decoded[processed_samples * channels_], &speech_type);
202 EXPECT_EQ(frame_size_ * channels_, dec_len);
213 input, decoded, processed_samples, channels_, tolerance, delay);
214 if (channels_ == 2
280 size_t channels_; member in namespace:webrtc
[all...] |
| /frameworks/native/libs/vr/libpdx_uds/ |
| channel_manager.cpp | 16 auto channel = channels_.find(handle);
17 if (channel == channels_.end()) {
20 channels_.erase(channel);
29 channels_.emplace(handle,
38 auto channel = channels_.find(handle);
39 return channel != channels_.end() ? &channel->second : nullptr;
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| opus_unittest.cc | 67 size_t channels_; member in class:webrtc::OpusTest
75 channels_(static_cast<size_t>(::testing::get<0>(GetParam()))),
120 rtc::CheckedDivExact(input_audio.size(), channels_),
135 PrepareSpeechData(channels_, block_length_ms, 2000);
140 channels_,
142 EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
146 channels_ == 1 ? 32000 : 64000));
149 std::vector<int16_t> silence(samples * channels_, 0);
156 int16_t* output_data_decode = new int16_t[samples * channels_];
233 CheckAudioBounded(output_data_decode, samples, channels_,
[all...] |
| audio_decoder_opus.cc | 18 : channels_(num_channels) {
20 WebRtcOpus_DecoderCreate(&dec_state_, channels_);
38 ret *= static_cast<int>(channels_); // Return total number of samples.
59 ret *= static_cast<int>(channels_); // Return total number of samples.
91 return channels_;
|
| opus_fec_test.cc | 51 size_t channels_; member in class:webrtc::OpusFecTest
70 channels_ = get<0>(GetParam());
72 printf("Coding %" PRIuS " channel signal at %d bps.\n", channels_, bit_rate_);
86 block_length_sample_ * channels_]);
99 block_length_sample_ * channels_ * sizeof(int16_t));
102 max_bytes_ = block_length_sample_ * channels_ * sizeof(int16_t);
104 out_data_.reset(new int16_t[2 * block_length_sample_ * channels_]);
107 // If channels_ == 1, use Opus VOIP mode, otherwise, audio mode.
108 int app = channels_ == 1 ? 0 : 1;
111 EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app))
[all...] |
| opus_speed_test.cc | 44 // If channels_ == 1, use Opus VOIP mode, otherwise, audio mode.
45 int app = channels_ == 1 ? 0 : 1;
47 EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app));
48 EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
|
| audio_decoder_opus.h | 45 const size_t channels_; member in class:webrtc::final
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/tools/ |
| audio_codec_speed_test.cc | 41 channels_ = get<0>(GetParam());
56 input_length_sample_ * channels_]);
69 input_length_sample_ * channels_ * sizeof(int16_t));
71 max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t);
72 out_data_.reset(new int16_t[output_length_sample_ * channels_]);
104 input_sampling_khz_, channels_, bit_rate_);
115 output_length_sample_ * channels_, out_file_);
117 data_pointer_ = (data_pointer_ + input_length_sample_ * channels_) %
|
| audio_codec_speed_test.h | 78 size_t channels_; member in class:webrtc::AudioCodecSpeedTest
|
| /external/webrtc/talk/media/webrtc/ |
| fakewebrtcvoiceengine.h | 56 if (channels_.find(channel) == channels_.end()) return -1;
218 RTC_CHECK(channels_.empty());
225 int GetNumChannels() const { return static_cast<int>(channels_.size()); }
227 return channels_[channel]->send_ssrc;
230 return channels_[channel]->playout;
233 return channels_[channel]->send;
236 return channels_[channel]->vad;
239 return channels_[channel]->opus_dtx;
242 return channels_[channel]->red
798 std::map<int, Channel*> channels_; member in class:cricket::FakeWebRtcVoiceEngine
[all...] |
| /external/webrtc/webrtc/p2p/base/ |
| transport.cc | 75 for (const auto& kv : channels_) {
81 if (channels_.empty()) {
85 auto iter = channels_.begin();
91 for (const auto& kv : channels_) {
119 for (const auto& kv : channels_) {
150 for (const auto& kv : channels_) {
170 auto iter = channels_.find(component);
171 if (iter == channels_.end()) {
173 channels_.insert(std::pair<int, TransportChannelImpl*>(component, channel));
206 auto iter = channels_.find(component)
[all...] |
| faketransportcontroller.h | 359 const ChannelMap& channels() const { return channels_; }
366 for (const auto& kv : channels_) {
373 for (const auto& kv : channels_) {
392 if (channels_.empty()) {
395 return channels_.begin()->second->GetSslRole(role);
400 for (const auto& kv : channels_) {
414 if (channels_.find(component) != channels_.end()) {
422 channels_[component] = channel;
427 channels_.erase(channel->component())
452 ChannelMap channels_; member in class:cricket::FakeTransport
[all...] |
| /external/webrtc/webrtc/modules/audio_device/include/ |
| audio_device_defines.h | 149 channels_(0),
154 channels_(channels),
159 channels_ = channels;
172 size_t channels() const { return channels_; }
175 size_t GetBytesPerFrame() const { return channels_ * kBitsPerSample / 8; }
182 bool is_valid() const { return ((sample_rate_ > 0) && (channels_ > 0)); }
203 size_t channels_; member in class:webrtc::AudioParameters
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| neteq_external_decoder_test.cc | 25 channels_(decoder_->Channels()) {
29 printf("%" PRIuS "\n", channels_);
58 EXPECT_EQ(channels_, num_channels);
|
| neteq_external_decoder_test.h | 57 size_t channels_; member in class:webrtc::test::NetEqExternalDecoderTest
|
| /external/webrtc/webrtc/modules/audio_coding/test/ |
| PacketLossTest.cc | 114 : channels_(channels),
115 in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz" :
131 int codec_id = acm->Codec("opus", 48000, channels_);
144 sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_,
159 receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_,
|
| PacketLossTest.h | 55 int channels_; member in class:webrtc::PacketLossTest
|
| /external/webrtc/webrtc/common_audio/ |
| channel_buffer.h | 34 // |channels_|:
46 channels_(new T*[num_channels * num_bands]),
54 channels_[j * num_channels_ + i] =
56 bands_[i * num_bands_ + j] = channels_[j * num_channels_ + i];
79 return &channels_[band * num_channels_];
108 slice[i] = &channels_[i][start_frame];
129 rtc::scoped_ptr<T* []> channels_; member in class:webrtc::ChannelBuffer
|
| /frameworks/native/libs/vr/libpdx_uds/private/uds/ |
| channel_manager.h | 33 std::unordered_map<int32_t, ChannelData> channels_; member in class:android::pdx::uds::ChannelManager
|
| /external/webrtc/webrtc/modules/audio_processing/ |
| noise_suppression_impl.h | 45 size_t channels_ GUARDED_BY(crit_) = 0;
|
| /external/webrtc/webrtc/voice_engine/test/auto_test/fixtures/ |
| after_initialization_fixture.h | 66 channels_[ssrc] = channel;
122 if (channels_[ssrc] != 0)
123 channel = channels_[ssrc];
151 std::map<uint32_t, int> channels_ GUARDED_BY(crit_.get());
|
