| /external/webrtc/webrtc/base/ |
| testutils.h | 304 send_buffer_.insert(send_buffer_.end(), data, data + len);
340 while (sent < send_buffer_.size()) {
341 int result = socket_->Send(&send_buffer_[sent],
342 send_buffer_.size() - sent);
349 size_t new_size = send_buffer_.size() - sent;
350 memmove(&send_buffer_[0], &send_buffer_[sent], new_size);
351 send_buffer_.resize(new_size);
355 if (!send_buffer_.empty())
375 Buffer send_buffer_, recv_buffer_; member in class:testing::SocketTestClient
[all...] |
| virtualsocketserver.cc | 489 size_t capacity = server_->send_buffer_capacity_ - send_buffer_.size();
497 send_buffer_.insert(send_buffer_.end(), cpv, cpv + consumed);
841 size_t data_size = std::min(socket->send_buffer_.size(), max_data_size);
845 AddPacketToNetwork(socket, recipient, cur_time, &socket->send_buffer_[0],
849 size_t new_buffer_size = socket->send_buffer_.size() - data_size;
852 if (data_size < socket->send_buffer_.size()) {
854 memmove(&socket->send_buffer_[0], &socket->send_buffer_[data_size],
857 socket->send_buffer_.resize(new_buffer_size)
[all...] |
| virtualsocketserver.h | 325 SendBuffer send_buffer_; member in class:rtc::VirtualSocket
|
| /external/webrtc/talk/app/webrtc/test/ |
| fakeaudiocapturemodule.cc | 617 Sample* buffer_ptr = reinterpret_cast<Sample*>(send_buffer_);
619 sizeof(send_buffer_) / kNumberBytesPerSample;
733 if (audio_callback_->RecordedDataIsAvailable(send_buffer_, kNumberSamples,
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| fakeaudiocapturemodule.h | 218 // SetBuffer() sets all samples in send_buffer_ to |value|.
272 char send_buffer_[kNumberSamples * kNumberBytesPerSample]; member in class:FakeAudioCaptureModule
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