/external/webrtc/talk/media/webrtc/ |
D | fakewebrtcvoiceengine.h | 56 if (channels_.find(channel) == channels_.end()) return -1; 218 RTC_CHECK(channels_.empty()); in ~FakeWebRtcVoiceEngine() 225 int GetNumChannels() const { return static_cast<int>(channels_.size()); } in GetNumChannels() 227 return channels_[channel]->send_ssrc; in GetLocalSSRC() 230 return channels_[channel]->playout; in GetPlayout() 233 return channels_[channel]->send; in GetSend() 236 return channels_[channel]->vad; in GetVAD() 239 return channels_[channel]->opus_dtx; in GetOpusDtx() 242 return channels_[channel]->red; in GetRED() 245 return channels_[channel]->codec_fec; in GetCodecFEC() [all …]
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/external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
D | opus_unittest.cc | 67 size_t channels_; member in webrtc::OpusTest 75 channels_(static_cast<size_t>(::testing::get<0>(GetParam()))), in OpusTest() 120 rtc::CheckedDivExact(input_audio.size(), channels_), in EncodeDecode() 135 PrepareSpeechData(channels_, block_length_ms, 2000); in TestDtxEffect() 140 channels_, in TestDtxEffect() 142 EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_)); in TestDtxEffect() 146 channels_ == 1 ? 32000 : 64000)); in TestDtxEffect() 149 std::vector<int16_t> silence(samples * channels_, 0); in TestDtxEffect() 156 int16_t* output_data_decode = new int16_t[samples * channels_]; in TestDtxEffect() 233 CheckAudioBounded(output_data_decode, samples, channels_, in TestDtxEffect() [all …]
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D | opus_fec_test.cc | 51 size_t channels_; member in webrtc::OpusFecTest 70 channels_ = get<0>(GetParam()); in SetUp() 72 printf("Coding %" PRIuS " channel signal at %d bps.\n", channels_, bit_rate_); in SetUp() 86 block_length_sample_ * channels_]); in SetUp() 99 block_length_sample_ * channels_ * sizeof(int16_t)); in SetUp() 102 max_bytes_ = block_length_sample_ * channels_ * sizeof(int16_t); in SetUp() 104 out_data_.reset(new int16_t[2 * block_length_sample_ * channels_]); in SetUp() 108 int app = channels_ == 1 ? 0 : 1; in SetUp() 111 EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app)); in SetUp() 112 EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_)); in SetUp() [all …]
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D | audio_decoder_opus.cc | 18 : channels_(num_channels) { in AudioDecoderOpus() 20 WebRtcOpus_DecoderCreate(&dec_state_, channels_); in AudioDecoderOpus() 38 ret *= static_cast<int>(channels_); // Return total number of samples. in DecodeInternal() 59 ret *= static_cast<int>(channels_); // Return total number of samples. in DecodeRedundantInternal() 91 return channels_; in Channels()
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D | opus_speed_test.cc | 45 int app = channels_ == 1 ? 0 : 1; in SetUp() 47 EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app)); in SetUp() 48 EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_)); in SetUp()
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/external/webrtc/webrtc/modules/audio_coding/neteq/ |
D | audio_multi_vector.cc | 25 channels_.push_back(new AudioVector); in AudioMultiVector() 34 channels_.push_back(new AudioVector(initial_size)); in AudioMultiVector() 40 std::vector<AudioVector*>::iterator it = channels_.begin(); in ~AudioMultiVector() 41 while (it != channels_.end()) { in ~AudioMultiVector() 49 channels_[i]->Clear(); in Clear() 55 channels_[i]->Clear(); in Zeros() 56 channels_[i]->Extend(length); in Zeros() 63 channels_[i]->CopyTo(&(*copy_to)[i]); in CopyTo() 73 channels_[0]->PushBack(append_this, length); in PushBackInterleaved() 86 channels_[channel]->PushBack(temp_array, length_per_channel); in PushBackInterleaved() [all …]
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D | audio_decoder_unittest.cc | 107 channels_(1), in AudioDecoderTest() 150 new int16_t[channels_ * samples_per_10ms]); in EncodeFrame() 157 samples_per_10ms, channels_, in EncodeFrame() 196 decoded.resize((processed_samples + frame_size_) * channels_, 0); in EncodeDecodeTest() 200 frame_size_ * channels_ * sizeof(int16_t), in EncodeDecodeTest() 201 &decoded[processed_samples * channels_], &speech_type); in EncodeDecodeTest() 202 EXPECT_EQ(frame_size_ * channels_, dec_len); in EncodeDecodeTest() 213 input, decoded, processed_samples, channels_, tolerance, delay); in EncodeDecodeTest() 214 if (channels_ == 2) in EncodeDecodeTest() 216 decoded, processed_samples, channels_, channel_diff_tolerance); in EncodeDecodeTest() [all …]
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D | sync_buffer.h | 79 const AudioVector& Channel(size_t n) const { return *channels_[n]; } in Channel() 80 AudioVector& Channel(size_t n) { return *channels_[n]; } in Channel()
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/external/webrtc/webrtc/voice_engine/ |
D | channel_manager.cc | 71 channels_.push_back(channel_owner); in CreateChannelInternal() 79 for (size_t i = 0; i < channels_.size(); ++i) { in GetChannel() 80 if (channels_[i].channel()->ChannelId() == channel_id) in GetChannel() 81 return channels_[i]; in GetChannel() 89 *channels = channels_; in GetAllChannels() 99 std::vector<ChannelOwner>::iterator to_delete = channels_.end(); in DestroyChannel() 100 for (auto it = channels_.begin(); it != channels_.end(); ++it) { in DestroyChannel() 110 if (to_delete != channels_.end()) { in DestroyChannel() 112 channels_.erase(to_delete); in DestroyChannel() 123 references = channels_; in DestroyAllChannels() [all …]
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D | channel_manager.h | 92 std::vector<ChannelOwner> channels_; variable 127 std::vector<ChannelOwner> channels_; variable
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/external/webrtc/webrtc/p2p/base/ |
D | transport.cc | 75 for (const auto& kv : channels_) { in SetIceRole() 81 if (channels_.empty()) { in GetRemoteSSLCertificate() 85 auto iter = channels_.begin(); in GetRemoteSSLCertificate() 91 for (const auto& kv : channels_) { in SetIceConfig() 119 for (const auto& kv : channels_) { in SetLocalTransportDescription() 150 for (const auto& kv : channels_) { in SetRemoteTransportDescription() 170 auto iter = channels_.find(component); in CreateChannel() 171 if (iter == channels_.end()) { in CreateChannel() 173 channels_.insert(std::pair<int, TransportChannelImpl*>(component, channel)); in CreateChannel() 206 auto iter = channels_.find(component); in GetChannel() [all …]
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D | faketransportcontroller.h | 359 const ChannelMap& channels() const { return channels_; } in channels() 366 for (const auto& kv : channels_) { in SetDestination() 373 for (const auto& kv : channels_) { in SetWritable() 392 if (channels_.empty()) { in GetSslRole() 395 return channels_.begin()->second->GetSslRole(role); in GetSslRole() 400 for (const auto& kv : channels_) { in SetSslMaxProtocolVersion() 414 if (channels_.find(component) != channels_.end()) { in CreateTransportChannel() 422 channels_[component] = channel; in CreateTransportChannel() 427 channels_.erase(channel->component()); in DestroyTransportChannel() 433 auto it = channels_.find(component); in GetFakeChannel() [all …]
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D | transportcontroller.cc | 148 if (it != channels_.end()) { in CreateTransportChannel_w() 169 channels_.insert(channels_.end(), RefCountedChannel(channel))->AddRef(); in CreateTransportChannel_w() 181 if (it == channels_.end()) { in DestroyTransportChannel_w() 193 channels_.erase(it); in DestroyTransportChannel_w() 267 channels_.begin(), channels_.end(), in FindChannel_w() 556 bool all_connected = !channels_.empty(); in UpdateAggregateStates_w() 557 bool all_completed = !channels_.empty(); in UpdateAggregateStates_w() 559 bool all_done_gathering = !channels_.empty(); in UpdateAggregateStates_w() 560 for (const auto& channel : channels_) { in UpdateAggregateStates_w()
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D | dtlstransportchannel_unittest.cc | 91 channels_.push_back(channel); in SetupChannels() 151 channels_.begin(); it != channels_.end(); ++it) { in Negotiate() 196 if (channels_.empty()) { in all_channels_writable() 199 for (cricket::DtlsTransportChannelWrapper* channel : channels_) { in all_channels_writable() 219 channels_.begin(); it != channels_.end(); ++it) { in CheckSrtp() 235 channels_.begin(); it != channels_.end(); ++it) { in CheckSsl() 250 ASSERT(channel < channels_.size()); in SendPackets() 265 int rv = channels_[channel]->SendPacket( in SendPackets() 274 ASSERT(channel < channels_.size()); in SendInvalidSrtpPacket() 280 return channels_[channel]->SendPacket( in SendInvalidSrtpPacket() [all …]
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D | turnserver.cc | 570 for (ChannelList::iterator it = channels_.begin(); in ~TurnServerAllocation() 571 it != channels_.end(); ++it) { in ~TurnServerAllocation() 751 channels_.push_back(channel1); in HandleChannelBindRequest() 851 for (ChannelList::const_iterator it = channels_.begin(); in FindChannel() 852 it != channels_.end(); ++it) { in FindChannel() 861 for (ChannelList::const_iterator it = channels_.begin(); in FindChannel() 862 it != channels_.end(); ++it) { in FindChannel() 904 std::find(channels_.begin(), channels_.end(), channel); in OnChannelDestroyed() 905 ASSERT(it != channels_.end()); in OnChannelDestroyed() 906 channels_.erase(it); in OnChannelDestroyed()
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/external/webrtc/webrtc/modules/audio_coding/codecs/tools/ |
D | audio_codec_speed_test.cc | 41 channels_ = get<0>(GetParam()); in SetUp() 56 input_length_sample_ * channels_]); in SetUp() 69 input_length_sample_ * channels_ * sizeof(int16_t)); in SetUp() 71 max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t); in SetUp() 72 out_data_.reset(new int16_t[output_length_sample_ * channels_]); in SetUp() 104 input_sampling_khz_, channels_, bit_rate_); in EncodeDecode() 115 output_length_sample_ * channels_, out_file_); in EncodeDecode() 117 data_pointer_ = (data_pointer_ + input_length_sample_ * channels_) % in EncodeDecode()
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/external/webrtc/webrtc/modules/audio_device/include/ |
D | audio_device_defines.h | 149 channels_(0), in AudioParameters() 154 channels_(channels), in AudioParameters() 159 channels_ = channels; in reset() 172 size_t channels() const { return channels_; } in channels() 175 size_t GetBytesPerFrame() const { return channels_ * kBitsPerSample / 8; } in GetBytesPerFrame() 182 bool is_valid() const { return ((sample_rate_ > 0) && (channels_ > 0)); } in is_valid() 203 size_t channels_; variable
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/external/webrtc/webrtc/common_audio/ |
D | channel_buffer.h | 46 channels_(new T*[num_channels * num_bands]), in data_() 54 channels_[j * num_channels_ + i] = in data_() 56 bands_[i * num_bands_ + j] = channels_[j * num_channels_ + i]; in data_() 79 return &channels_[band * num_channels_]; in channels() 108 slice[i] = &channels_[i][start_frame]; in Slice() 129 rtc::scoped_ptr<T* []> channels_; variable
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/external/webrtc/talk/media/base/ |
D | fakemediaengine.h | 741 channels_.push_back(ch); in CreateChannel() 745 return (channels_.size() > index) ? channels_[index] : NULL; in GetChannel() 748 channels_.erase(std::find(channels_.begin(), channels_.end(), channel)); in UnregisterChannel() 774 std::vector<FakeVoiceMediaChannel*> channels_; 802 channels_.push_back(ch); in CreateChannel() 806 return (channels_.size() > index) ? channels_[index] : NULL; in GetChannel() 809 channels_.erase(std::find(channels_.begin(), channels_.end(), channel)); in UnregisterChannel() 834 std::vector<FakeVideoMediaChannel*> channels_; 920 channels_.push_back(ch); in CreateChannel() 925 return (channels_.size() > index) ? channels_[index] : NULL; in GetChannel() [all …]
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/external/webrtc/webrtc/modules/audio_coding/test/ |
D | PacketLossTest.cc | 114 : channels_(channels), in PacketLossTest() 115 in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz" : in PacketLossTest() 131 int codec_id = acm->Codec("opus", 48000, channels_); in Perform() 144 sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_, in Perform() 159 receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_, in Perform()
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
D | neteq_quality_test.cc | 213 channels_(static_cast<size_t>(FLAGS_channels)), in NetEqQualityTest() 250 max_payload_bytes_ = in_size_samples_ * channels_ * sizeof(int16_t); in NetEqQualityTest() 251 in_data_.reset(new int16_t[in_size_samples_ * channels_]); in NetEqQualityTest() 253 out_data_.reset(new int16_t[out_size_samples_ * channels_]); in NetEqQualityTest() 399 int ret = neteq_->GetAudio(out_size_samples_ * channels_, &out_data_[0], in DecodeBlock() 405 assert(channels == channels_); in DecodeBlock() 418 ASSERT_TRUE(in_file_->Read(in_size_samples_ * channels_, &in_data_[0])); in Simulate()
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D | neteq_external_decoder_test.cc | 25 channels_(decoder_->Channels()) { in NetEqExternalDecoderTest() 29 printf("%" PRIuS "\n", channels_); in NetEqExternalDecoderTest() 58 EXPECT_EQ(channels_, num_channels); in GetOutputAudio()
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/external/webrtc/webrtc/voice_engine/test/auto_test/fixtures/ |
D | after_initialization_fixture.h | 66 channels_[ssrc] = channel; in AddChannel() 122 if (channels_[ssrc] != 0) in SendPackets() 123 channel = channels_[ssrc]; in SendPackets() 151 std::map<uint32_t, int> channels_ GUARDED_BY(crit_.get());
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/external/webrtc/webrtc/modules/audio_coding/neteq/test/ |
D | neteq_opus_quality_test.cc | 139 if (channels_ > 1) { in NetEqOpusQualityTest() 147 WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_); in SetUp() 194 pointer += sub_block_size_samples_ * channels_; in EncodeBlock()
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/external/webrtc/webrtc/modules/audio_processing/ |
D | noise_suppression_impl.cc | 57 channels_ = channels; in Initialize() 115 Initialize(channels_, sample_rate_hz_); in Enable()
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